From ibc at aliax.net  Fri Apr  1 04:19:56 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Fri, 1 Apr 2011 02:19:56 +0200
Subject: [Freeswitch-users] Why FS rewrites From header?
In-Reply-To: <AANLkTin2X8FvBhPf65rDXqDXMv8O36FwUS_Par9iPR3s@mail.gmail.com>
References: <538261301575539@web100.yandex.ru>
	<AANLkTin2X8FvBhPf65rDXqDXMv8O36FwUS_Par9iPR3s@mail.gmail.com>
Message-ID: <BANLkTimm5nhyWEw33QeyRmpHgmhB-nmKig@mail.gmail.com>

2011/3/31 Steven Ayre <steveayre at gmail.com>:
> The aleg and bleg are 2 different separate calls, and FS joins the
> signalling media on the 2.
>
> The From etc headers have to have the address of FS because that's what's
> making the call.

A B2BUA could handle different domains (local domains). It's common in
a multidomain IP environment. Doesn't FS allow it?
A SIP user is identified by a complete AoR (user and domain, like in
mail world), does FS assume that just the username part is the
identifier so alice at domainA.org is the same as alice at domainB.org for
FS?

-- 
I?aki Baz Castillo
<ibc at aliax.net>


From ibc at aliax.net  Fri Apr  1 04:24:03 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Fri, 1 Apr 2011 02:24:03 +0200
Subject: [Freeswitch-users] Why FS rewrites From header?
In-Reply-To: <BANLkTimm5nhyWEw33QeyRmpHgmhB-nmKig@mail.gmail.com>
References: <538261301575539@web100.yandex.ru>
	<AANLkTin2X8FvBhPf65rDXqDXMv8O36FwUS_Par9iPR3s@mail.gmail.com>
	<BANLkTimm5nhyWEw33QeyRmpHgmhB-nmKig@mail.gmail.com>
Message-ID: <BANLkTin8gut1aqo172QB=XznOGX-snx8Ww@mail.gmail.com>

2011/4/1 I?aki Baz Castillo <ibc at aliax.net>:
> A B2BUA could handle different domains (local domains). It's common in
> a multidomain IP environment. Doesn't FS allow it?
> A SIP user is identified by a complete AoR (user and domain, like in
> mail world), does FS assume that just the username part is the
> identifier so alice at domainA.org is the same as alice at domainB.org for
> FS?

In my case I've a SIP proxy that manages different local domains, and
I plan to put some FS boxes behind it to offer PBX services. But for
that I need that FS understands that alice at domainA.org is a different
user than alice at domainB.org, and when it routes back a call to the SIP
proxy/registrar it must keep the original From URI (also the domain).

This is:

  alice at domainA.org  ----> Proxy/Registrar -----> FS  ----> same
Proxy/Registrar ----> alice at domainB.org

When alice at domainB.org receives the call, she must see
alice at domainA.org in the From header.

Does FS allow it?


-- 
I?aki Baz Castillo
<ibc at aliax.net>


From victor.chukalovskiy at utoronto.ca  Fri Apr  1 04:33:09 2011
From: victor.chukalovskiy at utoronto.ca (Victor Chukalovskiy)
Date: Thu, 31 Mar 2011 20:33:09 -0400
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <4D94C282.1090903@KennedySoftware.ie>
References: <4D94C282.1090903@KennedySoftware.ie>
Message-ID: <4D951D45.5010005@utoronto.ca>

Hi Mike,

A bit off-topic but here are my 50 cents:

-Did you consider building a wireless bridge with a $40 WiFi router 
running DD-WRT/Tomato/OpenWRT etc?
This way you can plug wired phones into LAN ports of the "bridge" and 
the router will bridge them to your main access point.
Asus WL-520GU will work and is really cheap.

-If you go with WiFi you should only use WPA or WPA2.
Less secure options (WEP :-) ) make all conversations accessible to public.

Regards,
Victor




On 03/31/2011 02:05 PM, Michael Kennedy wrote:
> Hello,
>
> Newbie here, and newbie on FS - but been lurking for a few YEARS!...
>
> I'm hoping to roll out FS where some areas in a building are wired, and
> other areas are on WiFi, and to deploy some SIP phones in both areas.
>
> I expected that many phone suppliers would have handsets with EITHER
> RJ45 or WiFi connectivity to the LAN, or even both! I've found only a
> single device, a Cisco SPA525G2! Furthermore, searching the FS site, and
> various VoIP sites, and running general searches, I've found no other
> SIP WiFi phones that look like standard desktop handsets.
>
> I'd appreciate any pointers to WiFi devices that are recommended with
> FS. Preferably "standard-looking" desktop units, and better still, if
> they had wired "sisters" - in appearance and functionality!
>
> Or... maybe best to invest in a few rolls of Cat-6 cable!!
>
> Thank you - and many thanks again to some of the regulars here for
> off-list guidance.
>     - 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 infos at madovsky.org  Fri Apr  1 05:16:18 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 31 Mar 2011 21:16:18 -0400
Subject: [Freeswitch-users] invite in conference
Message-ID: <9EDC24631B3343918890B2D2D60FF755@e1705>

I have more info after dozen different tests.
If I invite in conf the same number several time, each time
the invited leg answers, like 1 second of latency is added (exponential)
so after 3 invites hangups I got 8 seconds of latency for the conference moderator voice
in the invited phone. concerning the invited voice into the conference the latency stays exactly the same after 3 invites.
Sorry I didn't triy to do it with 3 different numbers as my cell is cut (no credits) and my landline phone also (bill due).. ;)

Thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, March 30, 2011 6:27 PM
  Subject: Re: [Freeswitch-users] invite in conference


  /usr/local/freeswitch/bin/fs_cli -x "conference confText dial\{inconf=true,originate_timeout=20,ignore_early_media=true,instant_ringback=true}user/11111 22222 hiConf"

  and

   /usr/local/freeswitch/bin/fs_cli -x "conference confText dial\{inconf=true,originate_timeout=20,ignore_early_media=true,instant_ringback=true}loopback/11111 22222 hiConf"

  is this dial event can be in other place that conference::maintenance in ESL ?

  thanks

    ----- Original Message ----- 
    From: Michael Collins 
    To: FreeSWITCH Users Help 
    Sent: Wednesday, March 30, 2011 1:20 PM
    Subject: Re: [Freeswitch-users] invite in conference


    what syntax are you using for the invitation? I would like to try it on my system and see if i can reproduce. 
    -MC


    On Wed, Mar 30, 2011 at 9:27 AM, Madovsky <infos at madovsky.org> wrote:

      When I invite in conference, I can't see any conference esl event of the new member invited and accepted in conference
      is it normal ?

      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






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


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From fieldpeak at gmail.com  Fri Apr  1 06:03:03 2011
From: fieldpeak at gmail.com (Charles)
Date: Fri, 1 Apr 2011 10:03:03 +0800
Subject: [Freeswitch-users] How to realize -GIT pull latest version to a
	local copy and work with prevoius changes
Message-ID: <4d953258.c4b3ec0a.5e9b.5c5a@mx.google.com>


I'm a newbie with GIT, a few months ago, i use git to pull the Head to a local copy and then on this copy i change some code, e.g. 'my test code' insides a lots of files.

now i found previous copy has some bugs and latest head fixed it, so i want to pull out the latest Head, however, meanwhile i want to keep my 
previous changes, is it doable? 

if not, i have to pull the latest Git to another local copy and on this copy  i have to manually rewrite my changes in all of relative fiels, it will be a heavy workload (my god ), would somebody can help how to minimize the manual work load to realize it, thanks in advance!

P.S. i'm using 'Git Extensions'  as git tool on windows platform...

Regards.
Charles

2011-04-01 



Charles 
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From jeff at jefflenk.com  Fri Apr  1 06:18:46 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 31 Mar 2011 19:18:46 -0700 (PDT)
Subject: [Freeswitch-users] How to realize -GIT pull latest version to a
 local copy and work with prevoius changes
In-Reply-To: <4d953258.c4b3ec0a.5e9b.5c5a@mx.google.com>
References: <4d953258.c4b3ec0a.5e9b.5c5a@mx.google.com>
Message-ID: <1301624326796-6229497.post@n2.nabble.com>

git stash save
git pull
git stash pop

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-realize-GIT-pull-latest-version-to-a-local-copy-and-work-with-prevoius-changes-tp6229486p6229497.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fieldpeak at gmail.com  Fri Apr  1 06:23:57 2011
From: fieldpeak at gmail.com (Charles)
Date: Fri, 1 Apr 2011 10:23:57 +0800
Subject: [Freeswitch-users] How to realize -GIT pull latest version to
	alocal copy and work with prevoius changes
References: <4d953258.c4b3ec0a.5e9b.5c5a@mx.google.com>,
	<1301624326796-6229497.post@n2.nabble.com>
Message-ID: <4d953740.1836640a.3853.6a3b@mx.google.com>

Hi Jeff,

Your reply of each time are all so valuable... admire you very much! 
Thank you very much!

Regards,
Charles


???? Jeff Lenk 
????? 2011-04-01  10:19:27 
???? freeswitch-users 
??? 
??? Re: [Freeswitch-users] How to realize -GIT pull latest version to alocal copy and work with prevoius changes 
 
git stash save
git pull
git stash pop
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-realize-GIT-pull-latest-version-to-a-local-copy-and-work-with-prevoius-changes-tp6229486p6229497.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
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From fieldpeak at gmail.com  Fri Apr  1 06:46:18 2011
From: fieldpeak at gmail.com (Charles)
Date: Fri, 1 Apr 2011 10:46:18 +0800
Subject: [Freeswitch-users] How to realize -GIT pull latest version to
	alocal copy and work with prevoius changes
References: <4d953258.c4b3ec0a.5e9b.5c5a@mx.google.com>,
	<1301624326796-6229497.post@n2.nabble.com>
Message-ID: <4d953c7d.0a3fec0a.4e36.5a1e@mx.google.com>

Hi Jeff,

During the procedure, will my changed code be uploaded to remote server or just in local?
thanks.


2011-04-01 



Charles 



???? Jeff Lenk 
????? 2011-04-01  10:19:27 
???? freeswitch-users 
??? 
??? Re: [Freeswitch-users] How to realize -GIT pull latest version to alocal copy and work with prevoius changes 
 
git stash save
git pull
git stash pop
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-realize-GIT-pull-latest-version-to-a-local-copy-and-work-with-prevoius-changes-tp6229486p6229497.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
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From jeff at jefflenk.com  Fri Apr  1 07:08:35 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Thu, 31 Mar 2011 20:08:35 -0700 (PDT)
Subject: [Freeswitch-users] How to realize -GIT pull latest version to
 alocal copy and work with prevoius changes
In-Reply-To: <4d953c7d.0a3fec0a.4e36.5a1e@mx.google.com>
References: <4d953258.c4b3ec0a.5e9b.5c5a@mx.google.com>
	<1301624326796-6229497.post@n2.nabble.com>
	<4d953c7d.0a3fec0a.4e36.5a1e@mx.google.com>
Message-ID: <1301627315889-6229566.post@n2.nabble.com>

no problem!

The stash save will move your local changes aside(to a holding place so to
speak) so you can pull the changes from the remote repo and then the stash
pop merges your changes back into the local directory. all this is done on
your local directory and repository.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-realize-GIT-pull-latest-version-to-a-local-copy-and-work-with-prevoius-changes-tp6229486p6229566.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fieldpeak at gmail.com  Fri Apr  1 07:10:58 2011
From: fieldpeak at gmail.com (Charles)
Date: Fri, 1 Apr 2011 11:10:58 +0800
Subject: [Freeswitch-users] How to realize -GIT pull latest version
	toalocal copy and work with prevoius changes
References: <4d953258.c4b3ec0a.5e9b.5c5a@mx.google.com>,
	<1301624326796-6229497.post@n2.nabble.com>,
	<4d953c7d.0a3fec0a.4e36.5a1e@mx.google.com>,
	<1301627315889-6229566.post@n2.nabble.com>
Message-ID: <4d954245.49c2ec0a.5daa.5975@mx.google.com>

thanks! understood.

2011-04-01 



Charles 



???? Jeff Lenk 
????? 2011-04-01  11:09:36 
???? freeswitch-users 
??? 
??? Re: [Freeswitch-users] How to realize -GIT pull latest version toalocal copy and work with prevoius changes 
 
no problem!
The stash save will move your local changes aside(to a holding place so to
speak) so you can pull the changes from the remote repo and then the stash
pop merges your changes back into the local directory. all this is done on
your local directory and repository.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-realize-GIT-pull-latest-version-to-a-local-copy-and-work-with-prevoius-changes-tp6229486p6229566.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
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From msc at freeswitch.org  Fri Apr  1 08:28:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 31 Mar 2011 21:28:13 -0700
Subject: [Freeswitch-users] phone callers muted when joining to
	conference
In-Reply-To: <BANLkTikQDd+X_AV-2LddEhXV_EG6rTC6xg@mail.gmail.com>
References: <BANLkTikQDd+X_AV-2LddEhXV_EG6rTC6xg@mail.gmail.com>
Message-ID: <AANLkTikoemFAehfTuCO409gmui3xUAFnGEtG4ajBc-CS@mail.gmail.com>

When you send a call into a conference it is by default not muted. There are
member flags that can be set to have callers be deaf, mute, etc.:

http://wiki.freeswitch.org/wiki/Mod_conference#Conference_Parameters

See "member-flags".

You might want to pastebin the script and the console debug from a call that
goes in auto-muted. There might be clues to what's going on.
-MC

On Wed, Mar 30, 2011 at 8:06 PM, deniro <ayhkor at gmail.com> wrote:

>
> Hi,
>    When join to the conference  through perl program
>    $session->execute("conference",conf-name at conf-profile);
> somehow phones were getting muted
> is there any parameters to pass while joining conference that will prevent
> phone callers be muted automatically
>
> thx
> deniro--
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Apr  1 08:30:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 31 Mar 2011 21:30:52 -0700
Subject: [Freeswitch-users] Variable interpolation of bridge b leg
In-Reply-To: <AANLkTi=ofqj3yY80U7WGVvRhD2ZJiYTB_3r=Gr9zQvoa@mail.gmail.com>
References: <AANLkTi=ofqj3yY80U7WGVvRhD2ZJiYTB_3r=Gr9zQvoa@mail.gmail.com>
Message-ID: <AANLkTikpV8kghVwtPrt4Ss=LTrZJYVvcJYH10DUwgkOE@mail.gmail.com>

Try this instead:
http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_bleg_app
and
http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_bleg_data

-MC

On Thu, Mar 31, 2011 at 1:32 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> I am setting channel variable execute_on_answer in my call to application
> bridge. Like this:
>
> <action application='bridge' data='{execute_on_answer=record_session
> $${base_dir}/storage/recorded_sessions/${uuid}_${strftime(%Y%m%d-%H%M%S)}.wav}sofia/external/
> DestinationNumber at 192.168.2.2' />
>
> The above works, and the application record_session is executed on the leg
> b. However, the uuid it gets is from leg a, and the timestamp is from the
> time bridge was executed, which as I understand, is happening because the
> variable interpolation is performed at the moment the application bridge is
> executed.
> So, is there a way to delay variable interpolation to the instant the b leg
> app is executed?
>
> 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
>
>
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From msc at freeswitch.org  Fri Apr  1 08:33:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 31 Mar 2011 21:33:32 -0700
Subject: [Freeswitch-users] user and public dialplan
In-Reply-To: <E1899A6727B348329FE4734599F4232A@e1705>
References: <E1899A6727B348329FE4734599F4232A@e1705>
Message-ID: <AANLkTin4db7XJU5Gpm0BkakruyZxkUAP7eHgfFE42m4O@mail.gmail.com>

Why can't you just route from the public context?
-MC

On Thu, Mar 31, 2011 at 9:48 AM, Madovsky <infos at madovsky.org> wrote:

>  forgot to say
> it's an external call but from inside a cluster
> and the dialstring is like /sofia/external/9999 at domain.ltd
>
> thanks
>
> ----- Original Message -----
> *From:* Madovsky <infos at madovsky.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Thursday, March 31, 2011 12:33 PM
> *Subject:* user and public dialplan
>
> example:
> - 9999 extension exisits in conf/directory
> - no public dialplan that matches 9999
>
> external call is coming to public dialplan.
> is FS will consider that 9999 exists or not ?
>
> 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
>
>
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From msc at freeswitch.org  Fri Apr  1 08:35:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 31 Mar 2011 21:35:09 -0700
Subject: [Freeswitch-users] Full username in caller_profile->username
In-Reply-To: <367151301558892@web113.yandex.ru>
References: <367151301558892@web113.yandex.ru>
Message-ID: <AANLkTinYKSEo3jJwEtzduPJpj6zwKXT669baX_6PLxfd@mail.gmail.com>

Could you please expand on this? A code snippet would be helpful, as would a
little context.

-MC

On Thu, Mar 31, 2011 at 1:08 AM, Serge S. Yuriev <me at nevian.org> wrote:

> Hello,
>
> caller_profile->channel_name shows sofia/internal/user at domain
> but caller_profile->username shows only user w/o domain part.
> How i can set username to include domain name in caller_profile->username?
>
> --
> wbr,
> Serge
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Apr  1 08:41:04 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 31 Mar 2011 21:41:04 -0700
Subject: [Freeswitch-users] Gateway with dynamic IP address
In-Reply-To: <C9C8E1CC-4621-44FF-8D19-D8DB35D41530@gmail.com>
References: <C9C8E1CC-4621-44FF-8D19-D8DB35D41530@gmail.com>
Message-ID: <AANLkTinQNJpc0UP4hSjnSaa5zqcqgVoLhV0ScrPDvwNN@mail.gmail.com>

On Thu, Mar 31, 2011 at 6:25 AM, Juan Wajnerman <juan.wajnerman at gmail.com>wrote:

> I asked this question yesterday in the IRC but I couldn't get a solution.
> I'd like to have a gateway configured in FreeSwitch without specifying the
> static IP address.
> I have this configuration:
>
> <user id="gw">
>   <gateways>
>     <gateway name="gw">
>       <param name="username" value="gw"/>
>       <param name="password" value="password"/>
>       <param name="register" value="false"/>
>       <param name="extension" value="2"/>
>     </gateway>
>   </gateways>
>   <params>
>     <param name="password" value="password"/>
>   </params>
>   <variables>
>     <variable name="user_context" value="default" />
>   </variables>
>  </user>
>
> and the SIP device is registering properly, but I cannot dial with
> addresses like: "sofia/gateway/gw/123456789".
> Note that this works if the gateway name is the IP address or host name, or
> if I add a "proxy" setting with the IP address.
>

You haven't set the realm parameter. Look at the example.com.xml file in
conf/sip_profiles/external/ and you'll see in the comments that if you don't
set the realm param then it goes to the name of the gateway. Set the realm
to the target IP or host name and try again.

-MC


>
> I have a similar configuration in asterisk, where the sip.conf contains:
>
> [gw]
> type=friend
> secret=password
> context=default
> host=dynamic
>
> And once the gateway is registered in asterisk, I can dial with
> "SIP/gw/123456789".
> Is there any way to make a similar configuration in FreeSwitch?
>
> Thanks!
> - Juan
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Apr  1 08:44:46 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 31 Mar 2011 21:44:46 -0700
Subject: [Freeswitch-users] incoming fax calls
In-Reply-To: <2AEA11608B0642348D4C867C5058F0AC@e1705>
References: <2AEA11608B0642348D4C867C5058F0AC@e1705>
Message-ID: <AANLkTikFZHE_QD7na=uABeaGaUeX85cibpGjfUNcwCsr@mail.gmail.com>

Is this an incoming call? If so then why are you doing "execute_on_media"?
Wouldn't you want to pre_answer the call, do the tone_detect and sleep for
5000ms or so, and then proceed on to the bridge?

-MC

On Thu, Mar 31, 2011 at 10:10 AM, Madovsky <infos at madovsky.org> wrote:

>  I'm trying to find a way to dectect a fax or call from the same extension
>
>         <extension name="myphone">
>                 <condition field="${sip_to_uri}"
> expression="^(9999)@$${domain}$">
>                         <action inline="true" application="export"
> data="dialed_extension=$1"/>
>                         <action application="set"
> data="session_in_hangup_hook=true"/>
>                         <action application="set"
> data="hangup_after_bridge=true"/>
>                         <action application="set"
> data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,CALL_REJECTED,NORMAL_CLEARING,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
>                         <action application="set"
> data="group_confirm_cancel_timeout=1"/>
>                         <action application="ring_ready"/>
>                         <action application="set"
> data="execute_on_media=tone_detect fax 1100 r +5000 transfer 'receivefax XML
> features' 1"/>
>                         <action application="export"
> data="originate_timeout=25"/>
>                         <action application="bridge"
> data="{sip_invite_domain=${sip_from_host},nibble_account=,nibble_rate=,origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number,}}user/${dialed_extension}"/>
>                 </condition>
>          </extension>
> is there a way to detect a fax before answer (2 rings for example) and
> avoid
> phone rings until no answer ?
>
> 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
>
>
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From juan.wajnerman at gmail.com  Fri Apr  1 08:50:28 2011
From: juan.wajnerman at gmail.com (Juan Wajnerman)
Date: Fri, 1 Apr 2011 01:50:28 -0300
Subject: [Freeswitch-users] Gateway with dynamic IP address
In-Reply-To: <AANLkTinQNJpc0UP4hSjnSaa5zqcqgVoLhV0ScrPDvwNN@mail.gmail.com>
References: <C9C8E1CC-4621-44FF-8D19-D8DB35D41530@gmail.com>
	<AANLkTinQNJpc0UP4hSjnSaa5zqcqgVoLhV0ScrPDvwNN@mail.gmail.com>
Message-ID: <828493E7-A5E7-4896-844F-271AB72AD38B@gmail.com>

That's exactly what I don't want to set: a static IP address for the gateway. In other words I'd like to use a "user" as if it were a gateway. Is that even possible in FreeSwitch?

On Apr 1, 2011, at 1:41 AM, Michael Collins <msc at freeswitch.org> wrote:

> 
> 
> On Thu, Mar 31, 2011 at 6:25 AM, Juan Wajnerman <juan.wajnerman at gmail.com> wrote:
> I asked this question yesterday in the IRC but I couldn't get a solution.
> I'd like to have a gateway configured in FreeSwitch without specifying the static IP address.
> I have this configuration:
> 
> <user id="gw">
>   <gateways>
>     <gateway name="gw">
>       <param name="username" value="gw"/>
>       <param name="password" value="password"/>
>       <param name="register" value="false"/>
>       <param name="extension" value="2"/>
>     </gateway>
>   </gateways>
>   <params>
>     <param name="password" value="password"/>
>   </params>
>   <variables>
>     <variable name="user_context" value="default" />
>   </variables>
>  </user>
> 
> and the SIP device is registering properly, but I cannot dial with addresses like: "sofia/gateway/gw/123456789".
> Note that this works if the gateway name is the IP address or host name, or if I add a "proxy" setting with the IP address.
> 
> You haven't set the realm parameter. Look at the example.com.xml file in conf/sip_profiles/external/ and you'll see in the comments that if you don't set the realm param then it goes to the name of the gateway. Set the realm to the target IP or host name and try again.
> 
> -MC
>  
> 
> I have a similar configuration in asterisk, where the sip.conf contains:
> 
> [gw]
> type=friend
> secret=password
> context=default
> host=dynamic
> 
> And once the gateway is registered in asterisk, I can dial with "SIP/gw/123456789".
> Is there any way to make a similar configuration in FreeSwitch?
> 
> Thanks!
> - Juan
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From infos at madovsky.org  Fri Apr  1 09:03:32 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 1 Apr 2011 01:03:32 -0400
Subject: [Freeswitch-users] user and public dialplan
References: <E1899A6727B348329FE4734599F4232A@e1705>
	<AANLkTin4db7XJU5Gpm0BkakruyZxkUAP7eHgfFE42m4O@mail.gmail.com>
Message-ID: <2127A79310B84102B18D2ADA2DF76183@e1705>

I thought it sofia/external was the way to rout to the public context.
how can I do it ?
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Friday, April 01, 2011 12:33 AM
  Subject: Re: [Freeswitch-users] user and public dialplan


  Why can't you just route from the public context?
  -MC


  On Thu, Mar 31, 2011 at 9:48 AM, Madovsky <infos at madovsky.org> wrote:

    forgot to say
    it's an external call but from inside a cluster
    and the dialstring is like /sofia/external/9999 at domain.ltd

    thanks
      ----- Original Message ----- 
      From: Madovsky 
      To: freeswitch-users at lists.freeswitch.org 
      Sent: Thursday, March 31, 2011 12:33 PM
      Subject: user and public dialplan


      example:
      - 9999 extension exisits in conf/directory
      - no public dialplan that matches 9999

      external call is coming to public dialplan.
      is FS will consider that 9999 exists or not ?

      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






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


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From fieldpeak at gmail.com  Fri Apr  1 09:37:11 2011
From: fieldpeak at gmail.com (Charles)
Date: Fri, 1 Apr 2011 13:37:11 +0800
Subject: [Freeswitch-users] How to realize -GIT pull latest version
	toalocal copy and work with prevoius changes
References: <4d953258.c4b3ec0a.5e9b.5c5a@mx.google.com>,
	<1301624326796-6229497.post@n2.nabble.com>,
	<4d953c7d.0a3fec0a.4e36.5a1e@mx.google.com>
Message-ID: <4d95648a.204b640a.6598.7377@mx.google.com>


Hi Jeff,

As my lastest understanding (learning from google just now ), i can also create a branch at local, on this branch i change my codes, and after a few days i can merge the lastest master(git from remote) to my branches... so, i can periodically (few days or months) do like this to keep my changes code and update latest master to my codes (when i need)... is my understanding correct?

Thanks!

Regards,
Charles

2011-04-01 



Charles 



???? Jeff Lenk 
????? 2011-04-01  11:09:36 
???? freeswitch-users 
??? 
??? Re: [Freeswitch-users] How to realize -GIT pull latest version toalocal copy and work with prevoius changes 
 
no problem!
The stash save will move your local changes aside(to a holding place so to
speak) so you can pull the changes from the remote repo and then the stash
pop merges your changes back into the local directory. all this is done on
your local directory and repository.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-realize-GIT-pull-latest-version-to-a-local-copy-and-work-with-prevoius-changes-tp6229486p6229566.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
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From infos at madovsky.org  Fri Apr  1 09:45:53 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 1 Apr 2011 01:45:53 -0400
Subject: [Freeswitch-users] user and public dialplan
Message-ID: <27ED4710EB7249FDAFB59B94444D4915@e1705>

oops, ok with sofia/default and sofia/public I think
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Friday, April 01, 2011 1:03 AM
  Subject: Re: [Freeswitch-users] user and public dialplan


  I thought it sofia/external was the way to rout to the public context.
  how can I do it ?
    ----- Original Message ----- 
    From: Michael Collins 
    To: FreeSWITCH Users Help 
    Sent: Friday, April 01, 2011 12:33 AM
    Subject: Re: [Freeswitch-users] user and public dialplan


    Why can't you just route from the public context? 
    -MC


    On Thu, Mar 31, 2011 at 9:48 AM, Madovsky <infos at madovsky.org> wrote:

      forgot to say
      it's an external call but from inside a cluster
      and the dialstring is like /sofia/external/9999 at domain.ltd

      thanks
        ----- Original Message ----- 
        From: Madovsky 
        To: freeswitch-users at lists.freeswitch.org 
        Sent: Thursday, March 31, 2011 12:33 PM
        Subject: user and public dialplan


        example:
        - 9999 extension exisits in conf/directory
        - no public dialplan that matches 9999

        external call is coming to public dialplan.
        is FS will consider that 9999 exists or not ?

        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
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From pablosaro at gmail.com  Fri Apr  1 10:07:06 2011
From: pablosaro at gmail.com (Pablo Hernan Saro)
Date: Fri, 1 Apr 2011 03:07:06 -0300
Subject: [Freeswitch-users] Application redirect
Message-ID: <AANLkTinLBB91HSJhVPP8W++-PNsYy9oLkW7UGPm-S_U2@mail.gmail.com>

Hi there,

As far as I understood from the documentation, when action application
redirect is invoked, the REFER method is implemented. Recently I signed up
for a VoIP service and realized that the provider only supports redirection
by implementing the UPDATE method. I want to redirect incoming calls
requesting specific DIDs to other FS box in a different network (not behind
the FS receiving the original INVITE). Perform the redirection by answering
the call and bridging it to the other FS box will result in resource wasting
and probably call quality degradation (and actually it will be transfer not
redirect). So, is there any way to use UPDATE instead of REFER to accomplish
redirection for requests received through specific gateway?
Thanks in advance.

Pablo
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From kbdfck at gmail.com  Fri Apr  1 10:41:40 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Fri, 1 Apr 2011 10:41:40 +0400
Subject: [Freeswitch-users] Gateway with dynamic IP address
In-Reply-To: <828493E7-A5E7-4896-844F-271AB72AD38B@gmail.com>
References: <C9C8E1CC-4621-44FF-8D19-D8DB35D41530@gmail.com>
	<AANLkTinQNJpc0UP4hSjnSaa5zqcqgVoLhV0ScrPDvwNN@mail.gmail.com>
	<828493E7-A5E7-4896-844F-271AB72AD38B@gmail.com>
Message-ID: <AANLkTinZWWijJBBWr0ArbtSGjHuv1cpHDDxKiZqj4Agg@mail.gmail.com>

You can create usual user in directory, it will register with FS, and then
you can dial it with arbitrary number, getting its host/port using
sofia_contact and constructing request URI you need.

2011/4/1 Juan Wajnerman <juan.wajnerman at gmail.com>

> That's exactly what I don't want to set: a static IP address for the
> gateway. In other words I'd like to use a "user" as if it were a gateway. Is
> that even possible in FreeSwitch?
>
>
> On Apr 1, 2011, at 1:41 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
>
> On Thu, Mar 31, 2011 at 6:25 AM, Juan Wajnerman <juan.wajnerman at gmail.com>wrote:
>
>> I asked this question yesterday in the IRC but I couldn't get a solution.
>> I'd like to have a gateway configured in FreeSwitch without specifying the
>> static IP address.
>> I have this configuration:
>>
>> <user id="gw">
>>   <gateways>
>>     <gateway name="gw">
>>       <param name="username" value="gw"/>
>>       <param name="password" value="password"/>
>>       <param name="register" value="false"/>
>>       <param name="extension" value="2"/>
>>     </gateway>
>>   </gateways>
>>   <params>
>>     <param name="password" value="password"/>
>>   </params>
>>   <variables>
>>     <variable name="user_context" value="default" />
>>   </variables>
>>  </user>
>>
>> and the SIP device is registering properly, but I cannot dial with
>> addresses like: "sofia/gateway/gw/123456789".
>> Note that this works if the gateway name is the IP address or host name,
>> or if I add a "proxy" setting with the IP address.
>>
>
> You haven't set the realm parameter. Look at the example.com.xml file in
> conf/sip_profiles/external/ and you'll see in the comments that if you don't
> set the realm param then it goes to the name of the gateway. Set the realm
> to the target IP or host name and try again.
>
> -MC
>
>
>>
>> I have a similar configuration in asterisk, where the sip.conf contains:
>>
>> [gw]
>> type=friend
>> secret=password
>> context=default
>> host=dynamic
>>
>> And once the gateway is registered in asterisk, I can dial with
>> "SIP/gw/123456789".
>> Is there any way to make a similar configuration in FreeSwitch?
>>
>> Thanks!
>> - Juan
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From ibc at aliax.net  Fri Apr  1 12:02:10 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Fri, 1 Apr 2011 10:02:10 +0200
Subject: [Freeswitch-users] Why FS rewrites From header?
In-Reply-To: <4D955D71.90108@opensipstack.org>
References: <538261301575539@web100.yandex.ru>
	<AANLkTin2X8FvBhPf65rDXqDXMv8O36FwUS_Par9iPR3s@mail.gmail.com>
	<BANLkTimm5nhyWEw33QeyRmpHgmhB-nmKig@mail.gmail.com>
	<BANLkTin8gut1aqo172QB=XznOGX-snx8Ww@mail.gmail.com>
	<4D955D71.90108@opensipstack.org>
Message-ID: <BANLkTimPAxvssD3kVdDa131X3eqFuXPd0Q@mail.gmail.com>

2011/4/1 Joegen E. Baclor <joegen at opensipstack.org>:
>> This is:
>>
>> ? alice at domainA.org ?----> ?Proxy/Registrar -----> ?FS ?----> ?same
>> Proxy/Registrar ----> ?alice at domainB.org
>>
>> When alice at domainB.org receives the call, she must see
>> alice at domainA.org in the From header.
>>
>> Does FS allow it?
>
>
> This might be what you need -
> http://wiki.freeswitch.org/wiki/Channel_Variables#sip_invite_domain

That's good, thanks for pointing out it.

However I would prefer some option like "trust_from_domain" or
something static in the profile configuration rather than the
dialplan.

Thanks.

-- 
I?aki Baz Castillo
<ibc at aliax.net>


From covici at ccs.covici.com  Fri Apr  1 12:40:42 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 01 Apr 2011 04:40:42 -0400
Subject: [Freeswitch-users] user and public dialplan
In-Reply-To: <2127A79310B84102B18D2ADA2DF76183@e1705>
References: <E1899A6727B348329FE4734599F4232A@e1705>
	<AANLkTin4db7XJU5Gpm0BkakruyZxkUAP7eHgfFE42m4O@mail.gmail.com>
	<2127A79310B84102B18D2ADA2DF76183@e1705>
Message-ID: <19605.1301647242@ccs.covici.com>

The public dialplan is where you do this.

Madovsky <infos at madovsky.org> wrote:

> I thought it sofia/external was the way to rout to the public context.
> how can I do it ?
>   ----- Original Message ----- 
>   From: Michael Collins 
>   To: FreeSWITCH Users Help 
>   Sent: Friday, April 01, 2011 12:33 AM
>   Subject: Re: [Freeswitch-users] user and public dialplan
> 
> 
>   Why can't you just route from the public context?
>   -MC
> 
> 
>   On Thu, Mar 31, 2011 at 9:48 AM, Madovsky <infos at madovsky.org> wrote:
> 
>     forgot to say
>     it's an external call but from inside a cluster
>     and the dialstring is like /sofia/external/9999 at domain.ltd
> 
>     thanks
>       ----- Original Message ----- 
>       From: Madovsky 
>       To: freeswitch-users at lists.freeswitch.org 
>       Sent: Thursday, March 31, 2011 12:33 PM
>       Subject: user and public dialplan
> 
> 
>       example:
>       - 9999 extension exisits in conf/directory
>       - no public dialplan that matches 9999
> 
>       external call is coming to public dialplan.
>       is FS will consider that 9999 exists or not ?
> 
>       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
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From richocet2 at hotmail.com  Fri Apr  1 09:03:38 2011
From: richocet2 at hotmail.com (Dave Bracken)
Date: Fri, 1 Apr 2011 05:03:38 +0000
Subject: [Freeswitch-users] freeswitch-users@lists.freeswitch.org
Message-ID: <BLU0-SMTP16442C77D7EE91C4951E3AF0BE0@phx.gbl>

freeswitch-users at lists.freeswitch.org nothing has ever been handed to me all the stress was starting to take a toll on me this is the most unique solution I came across http://j.mp/hfRo4B now im the one that makes the calls please keep this between us


From covici at ccs.covici.com  Fri Apr  1 13:41:34 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 01 Apr 2011 05:41:34 -0400
Subject: [Freeswitch-users] cannot compile 64 bit freeswitch
Message-ID: <23739.1301650894@ccs.covici.com>

Here is some of what I get when i try to compile fs using 64-bit C
compiler and libraries -- using gentoo.

/usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
i386 architecture of input file `./.libs/libsqlite3.a(complete.o)' is
incompatib\le with i386:x86-64 output
/usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
i386 architecture of input file `./.libs/libsqlite3.a(main.o)' is
incompatible w\ith i386:x86-64 output
/usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
i386 architecture of input file `./.libs/libsqlite3.a(os_unix.o)' is
incompatibl\e with i386:x86-64 output
/usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
i386 architecture of input file `./.libs/libsqlite3.a(prepare.o)' is
incompatibl\e with i386:x86-64 output
/usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
i386 architecture of input file `./.libs/libsqlite3.a(printf.o)' is
incompatible\ with i386:x86-64 output


There is more like that and then it dies with a number of undefined
references.

How can I fix this library or what else should I do?  I restarted from
./bootstrap.sh  ./config and make.

Any assistance would be appreciated.

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From ibc at aliax.net  Fri Apr  1 13:43:15 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Fri, 1 Apr 2011 11:43:15 +0200
Subject: [Freeswitch-users] Application redirect
In-Reply-To: <AANLkTinLBB91HSJhVPP8W++-PNsYy9oLkW7UGPm-S_U2@mail.gmail.com>
References: <AANLkTinLBB91HSJhVPP8W++-PNsYy9oLkW7UGPm-S_U2@mail.gmail.com>
Message-ID: <BANLkTimTw4+m-nw+ipOCsD0VEp8OerOpVg@mail.gmail.com>

2011/4/1 Pablo Hernan Saro <pablosaro at gmail.com>:
> As far as I understood from the documentation, when action application
> redirect is invoked, the REFER method is implemented. Recently I signed up
> for a VoIP service and realized that the provider only supports redirection
> by implementing the UPDATE method.?I want to redirect incoming calls
> requesting specific DIDs to other FS box in a different network (not behind
> the FS receiving the original INVITE). Perform the redirection by answering
> the call and bridging it to the other FS box will result in resource wasting
> and probably call quality degradation (and actually it will be transfer not
> redirect). So, is there any way to use UPDATE instead of REFER to accomplish
> redirection for requests received through specific gateway?

UPDATE for a redirection? that's not possible, UPDATE is just use to
allow modifyng an SDP during an early-dialog (but also works for
establixhed dialogs similar to a re-INVITE).

Usually SIP providers don't allow neither receiving a 302 from a
client (to make a redirection) or a REFER, they are SIP trunk
providers, not a PBX. Is your responsability to handle redirections in
your platfform so the provider gets not involved at all.

A good solution could be setting a SIP proxy (i.e. Kamailio) between
the provider and your FS boxes so it can route incoming calls to the
appropriate FS (or based on load-balancing/failover fashion) and could
also receive a 302 from the first selected FS and convert it into a
new INVITE for the second FS.

Don't disturb a SIP trunk provider with your internal logic stuff, never.

-- 
I?aki Baz Castillo
<ibc at aliax.net>


From peder at networkoblivion.com  Fri Apr  1 17:08:33 2011
From: peder at networkoblivion.com (Peder)
Date: Fri, 1 Apr 2011 08:08:33 -0500
Subject: [Freeswitch-users] "Free" Conference Calling
In-Reply-To: <AANLkTini7aKRL2mcV2LYsCPxEcUr+G+eg_93m8rPSyzP@mail.gmail.com>
References: <AANLkTikK8zUBnPtBX4n+4S3dYn+_VPiNJieSLzO7Vuhi@mail.gmail.com>
	<AANLkTini7aKRL2mcV2LYsCPxEcUr+G+eg_93m8rPSyzP@mail.gmail.com>
Message-ID: <011e01cbf06d$e7c32ae0$b74980a0$@com>

Beware calling these numbers on pay as you go.  I've seen rates of $0.20/min
and higher (I seem to recall even seeing one at $0.40).  It gets expensive
for a "free" call.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Tuesday, March 29, 2011 1:07 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] "Free" Conference Calling

 

grnvoip doesn't block them either, but does charge a premium for termination
to those numbers..

-Avi

On Tue, Mar 29, 2011 at 7:49 PM, Max Clark <max.clark at gmail.com> wrote:

Hello,

I've noticed a consistent pattern for SIP Termination providers not
completing calls to the "Free" Conference lines due to costs. What's
the best way to deal with this? Are there published lists of these
providers numbers that I can use to influence my LCR? Are there SIP
Termination providers that explicitly deal with these lines?

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

 

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From richocet2 at hotmail.com  Fri Apr  1 13:35:41 2011
From: richocet2 at hotmail.com (Dave Bracken)
Date: Fri, 1 Apr 2011 09:35:41 +0000
Subject: [Freeswitch-users] freeswitch-users@lists.freeswitch.org
Message-ID: <BLU0-SMTP151A8BEAD5965C220479F39F0BE0@phx.gbl>

freeswitch-users at lists.freeswitch.org ive overcome my fair share of hardships despite the circumstances I stayed optimistic this was my last resort http://j.mp/fMBYbI now I am complete you can pull this off too


From mayamatakeshi at gmail.com  Fri Apr  1 17:38:07 2011
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Fri, 1 Apr 2011 22:38:07 +0900
Subject: [Freeswitch-users] Variable interpolation of bridge b leg
In-Reply-To: <AANLkTikpV8kghVwtPrt4Ss=LTrZJYVvcJYH10DUwgkOE@mail.gmail.com>
References: <AANLkTi=ofqj3yY80U7WGVvRhD2ZJiYTB_3r=Gr9zQvoa@mail.gmail.com>
	<AANLkTikpV8kghVwtPrt4Ss=LTrZJYVvcJYH10DUwgkOE@mail.gmail.com>
Message-ID: <AANLkTimhC_Qrv4oguQvjQSgbfcUwZo+MXpJuiV31gJF2@mail.gmail.com>

Thanks,
I tried that but still it doesn't work.
The following dialplan (starting record_session 2 times) generates filenames
with the same UUID/TimeStamp:

<action application='set'
data='bridge_pre_execute_bleg_app=record_session'/>
<action application='set'
data='bridge_pre_execute_bleg_data=/recorded_sessions/file1_${uuid}_${strftime(%Y%m%d-%H%M%S)}.wav'/>
<action application='bridge' data='{execute_on_answer=record_session
/recorded_sessions/file2_${uuid}_${strftime(%Y%m%d-%H%M%S)}.wav}sofia/internal/
term1 at 127.0.0.1'/>

file1_b3b338d8-8e52-4077-8b41-0ddc587716da_20110401-222314.wav
file2_b3b338d8-8e52-4077-8b41-0ddc587716da_20110401-222314.wav

But I think there is a simple solution: I just have to use ESL to watch for
CHANNEL_ANSWER or set execute_on_answer=lua somescript.lua" and set the
filename inside the script.

br,
takeshi

On Fri, Apr 1, 2011 at 1:30 PM, Michael Collins <msc at freeswitch.org> wrote:

> Try this instead:
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_bleg_app
> and
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#bridge_pre_execute_bleg_data
>
> -MC
>
> On Thu, Mar 31, 2011 at 1:32 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:
>
>> I am setting channel variable execute_on_answer in my call to application
>> bridge. Like this:
>>
>> <action application='bridge' data='{execute_on_answer=record_session
>> $${base_dir}/storage/recorded_sessions/${uuid}_${strftime(%Y%m%d-%H%M%S)}.wav}sofia/external/
>> DestinationNumber at 192.168.2.2' />
>>
>> The above works, and the application record_session is executed on the leg
>> b. However, the uuid it gets is from leg a, and the timestamp is from the
>> time bridge was executed, which as I understand, is happening because the
>> variable interpolation is performed at the moment the application bridge is
>> executed.
>> So, is there a way to delay variable interpolation to the instant the b
>> leg app is executed?
>>
>> 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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jeff at jefflenk.com  Fri Apr  1 17:56:28 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 1 Apr 2011 06:56:28 -0700 (PDT)
Subject: [Freeswitch-users] How to realize -GIT pull latest version
 toalocal copy and work with prevoius changes
In-Reply-To: <4d95648a.204b640a.6598.7377@mx.google.com>
References: <4d953258.c4b3ec0a.5e9b.5c5a@mx.google.com>
	<1301624326796-6229497.post@n2.nabble.com>
	<4d953c7d.0a3fec0a.4e36.5a1e@mx.google.com>
	<4d95648a.204b640a.6598.7377@mx.google.com>
Message-ID: <1301666188189-6230871.post@n2.nabble.com>

Hi Charles,

You sure can do that. Thats what git it for. Just be aware if you have local
changes and have problems it may be more difficult for "you" to determine
whether your modifications or the base code possibly is at fault. Potential
problems in the fs base code have to be reproduced with git head with no
other modifications or being able to demonstrate clearly what the problem is
and then submitting diffs against git head.

Jeff

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-realize-GIT-pull-latest-version-to-a-local-copy-and-work-with-prevoius-changes-tp6229486p6230871.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From me at nevian.org  Fri Apr  1 17:57:24 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Fri, 01 Apr 2011 17:57:24 +0400
Subject: [Freeswitch-users] Full username in caller_profile->username
In-Reply-To: <AANLkTinYKSEo3jJwEtzduPJpj6zwKXT669baX_6PLxfd@mail.gmail.com>
References: <367151301558892@web113.yandex.ru>
	<AANLkTinYKSEo3jJwEtzduPJpj6zwKXT669baX_6PLxfd@mail.gmail.com>
Message-ID: <1067231301666244@web109.yandex.ru>

Hello,

Mea culpa..

I'm talking lets say about mod_radius_cdr

Call From test at domain.tld to me at domain2.tld

Using this snippet
			if (profile->username) {
				if (rc_avpair_add(rad_config, &send, PW_USER_NAME, (void *) profile->username, -1, 0) == NULL) {
					switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "failed adding User-Name: %s\n", profile->username);
					rc_destroy(rad_config);
					goto end;
				}
			}

Gives me (w/o domain.tld part)
01.04.2011 17:51:10 VERBOSE     [0x2b0ff723e910] [ParseBody]    Attribute 
'User-Name', value: "test"

That should I do to get full URI?

01.04.2011, 08:35, "Michael Collins" <msc at freeswitch.org>:
> Could you please expand on this? A code snippet would be helpful, as would a little context.
>
> -MC
>
> On Thu, Mar 31, 2011 at 1:08 AM, Serge S. Yuriev <me at nevian.org> wrote:
>> Hello,
>>
>> caller_profile->channel_name shows sofia/internal/user at domain
>> but caller_profile->username shows only user w/o domain part.
>> How i can set username to include domain name in caller_profile->username?
>>
>> --
>> wbr,
>> Serge
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
wbr,
Serge


From me at nevian.org  Fri Apr  1 18:12:04 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Fri, 01 Apr 2011 18:12:04 +0400
Subject: [Freeswitch-users] Why FS rewrites From header?
In-Reply-To: <AANLkTin2X8FvBhPf65rDXqDXMv8O36FwUS_Par9iPR3s@mail.gmail.com>
References: <538261301575539@web100.yandex.ru>
	<AANLkTin2X8FvBhPf65rDXqDXMv8O36FwUS_Par9iPR3s@mail.gmail.com>
Message-ID: <1070491301667124@web98.yandex.ru>

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From curriegrad2004 at gmail.com  Fri Apr  1 18:32:45 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 1 Apr 2011 07:32:45 -0700
Subject: [Freeswitch-users] cannot compile 64 bit freeswitch
In-Reply-To: <23739.1301650894@ccs.covici.com>
References: <23739.1301650894@ccs.covici.com>
Message-ID: <AANLkTikzjWkFpz9mO=nKsGRO0yJ152pasgaNp2RS2=+b@mail.gmail.com>

Seems like your compiler is trying to link x86 code with x86_64 code.
Have you tried adding this before the ./configure command?

eg. CFLAGS=-m64 CXXFLAGS=-m64 LDFLAGS=-m64 ./configure

On Fri, Apr 1, 2011 at 2:41 AM,  <covici at ccs.covici.com> wrote:
> Here is some of what I get when i try to compile fs using 64-bit C
> compiler and libraries -- using gentoo.
>
> /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> i386 architecture of input file `./.libs/libsqlite3.a(complete.o)' is
> incompatib\le with i386:x86-64 output
> /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> i386 architecture of input file `./.libs/libsqlite3.a(main.o)' is
> incompatible w\ith i386:x86-64 output
> /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> i386 architecture of input file `./.libs/libsqlite3.a(os_unix.o)' is
> incompatibl\e with i386:x86-64 output
> /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> i386 architecture of input file `./.libs/libsqlite3.a(prepare.o)' is
> incompatibl\e with i386:x86-64 output
> /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> i386 architecture of input file `./.libs/libsqlite3.a(printf.o)' is
> incompatible\ with i386:x86-64 output
>
>
> There is more like that and then it dies with a number of undefined
> references.
>
> How can I fix this library or what else should I do? ?I restarted from
> ./bootstrap.sh ?./config and make.
>
> Any assistance would be appreciated.
>
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.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 pablosaro at gmail.com  Fri Apr  1 18:36:38 2011
From: pablosaro at gmail.com (Pablo Hernan Saro)
Date: Fri, 1 Apr 2011 11:36:38 -0300
Subject: [Freeswitch-users] Application redirect
In-Reply-To: <BANLkTimTw4+m-nw+ipOCsD0VEp8OerOpVg@mail.gmail.com>
References: <AANLkTinLBB91HSJhVPP8W++-PNsYy9oLkW7UGPm-S_U2@mail.gmail.com>
	<BANLkTimTw4+m-nw+ipOCsD0VEp8OerOpVg@mail.gmail.com>
Message-ID: <AANLkTi=u905KfpzwPZqjE4_n7mE_5tb+RVDT3yHv8hqz@mail.gmail.com>

Thanks for your answer I?aki. You're right, I was messing it up... Handling
SIP trunks with a proxy between providers and my FS boxes is the best way to
accomplish that.

On Fri, Apr 1, 2011 at 6:43 AM, I?aki Baz Castillo <ibc at aliax.net> wrote:

> 2011/4/1 Pablo Hernan Saro <pablosaro at gmail.com>:
> > As far as I understood from the documentation, when action application
> > redirect is invoked, the REFER method is implemented. Recently I signed
> up
> > for a VoIP service and realized that the provider only supports
> redirection
> > by implementing the UPDATE method. I want to redirect incoming calls
> > requesting specific DIDs to other FS box in a different network (not
> behind
> > the FS receiving the original INVITE). Perform the redirection by
> answering
> > the call and bridging it to the other FS box will result in resource
> wasting
> > and probably call quality degradation (and actually it will be transfer
> not
> > redirect). So, is there any way to use UPDATE instead of REFER to
> accomplish
> > redirection for requests received through specific gateway?
>
> UPDATE for a redirection? that's not possible, UPDATE is just use to
> allow modifyng an SDP during an early-dialog (but also works for
> establixhed dialogs similar to a re-INVITE).
>
> Usually SIP providers don't allow neither receiving a 302 from a
> client (to make a redirection) or a REFER, they are SIP trunk
> providers, not a PBX. Is your responsability to handle redirections in
> your platfform so the provider gets not involved at all.
>
> A good solution could be setting a SIP proxy (i.e. Kamailio) between
> the provider and your FS boxes so it can route incoming calls to the
> appropriate FS (or based on load-balancing/failover fashion) and could
> also receive a 302 from the first selected FS and convert it into a
> new INVITE for the second FS.
>
> Don't disturb a SIP trunk provider with your internal logic stuff, never.
>
> --
> I?aki Baz Castillo
> <ibc at aliax.net>
>
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From michal.bielicki at seventhsignal.de  Fri Apr  1 18:47:37 2011
From: michal.bielicki at seventhsignal.de (Michal Bielicki)
Date: Fri, 1 Apr 2011 16:47:37 +0200
Subject: [Freeswitch-users] Build on solaris fails in libs/esl
In-Reply-To: <AANLkTi=ksjc+aeGHfnvbnzGhPUvkyxWCXb-Gn5wvDD0y@mail.gmail.com>
References: <AANLkTimGtXWdRPAv8OJYWeu8VYtCF8ZfQt0osGVgU7G=@mail.gmail.com>
	<AANLkTinab+WgVXUYtWaSccbwBV7G7LO-qjjcE4bYAKXj@mail.gmail.com>
	<AANLkTinAtxnnmbVm-tBvEuOxUQ7=z2JebB9TK+0WdBuF@mail.gmail.com>
	<AANLkTim77j3aBU0YrR8YGZEp=bDf9ND3-T0NXSznW4ov@mail.gmail.com>
	<AANLkTikJ4bYiWcsU5uLjtV4pfcL_LV1bQbqUNsfAAZu0@mail.gmail.com>
	<6ADAA4B6-802B-49F6-8C52-6BBCDE176F16@seventhsignal.de>
	<AANLkTi=ksjc+aeGHfnvbnzGhPUvkyxWCXb-Gn5wvDD0y@mail.gmail.com>
Message-ID: <1882A54F-08F0-4BD3-B928-DF88240719DA@seventhsignal.de>

Will take another day or two, I am a bit overloaded currently and automaking stuff is always a PITA :)

Am 30.03.2011 um 09:09 schrieb A E [Gmail]:

> On Tue, Mar 29, 2011 at 11:33 AM, Michal Bielicki <michal.bielicki at seventhsignal.de> wrote:
> Its a problem with the build scripts for esl. looking at that right now ..
> 
> 
> Hi Michal,
> 
> did you find anything. I did see your later post which said that all settings in automake are missing? Is this just a case with my installation or is it missing in the source in git? I'm assuming it's only the settings that affect Solaris? I'm still confused how you get it to build all the time although like I'd said, I noticed you don't build any of the problem modules like hash, esl, silk and a few others I get problems with. 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Gesch?ftsf?hrer / CEO

Seventh Signal Ltd. & Co. KG
Weigandufer 45, B?ro 115, D-12059 Berlin
Voice: +49 30 60988730

Amtsgericht Charlottenburg HRA 44413 B
Ust.-ID: DE266981999
Gesch?ftsf?hrer: Michal Bielicki
Pers?nlich Haftende Gesellschafterin:
Seventh Signal Ltd, 69 Great Hampton St. Birmingham, 
B18 6EW, GB, Company Nr.: 06889439
WWW.: http://www.seventhsignal.de

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From Info at KennedySoftware.ie  Fri Apr  1 18:59:05 2011
From: Info at KennedySoftware.ie (Michael Kennedy)
Date: Fri, 01 Apr 2011 15:59:05 +0100
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <4D951D45.5010005@utoronto.ca>
References: <4D94C282.1090903@KennedySoftware.ie>
	<4D951D45.5010005@utoronto.ca>
Message-ID: <4D95E839.2070403@KennedySoftware.ie>

Victor,

> A bit off-topic but here are my 50 cents:

Oopppssss, my apologies - I thought it might be a common query for folks 
thinking about FS - but maybe in another "list"?

> -Did you consider building a wireless bridge with a $40 WiFi router
> running DD-WRT/Tomato/OpenWRT etc?

I did NOT - and I've deployed a lot of them to support "PC"s! THANK YOU!

> This way you can plug wired phones into LAN ports of the "bridge" and
> the router will bridge them to your main access point.
> Asus WL-520GU will work and is really cheap.

EXCELLENT suggestion!

(Maybe I'm drifting even more O-T, but... I'm also glad you did not 
mention WiFi devices from Linksys - in my experience, some of these 
boxes performed very poorly, but I seem to be the only one on the planet 
with these experiences!).

> -If you go with WiFi you should only use WPA or WPA2.
> Less secure options (WEP :-) ) make all conversations accessible to public.

Yes, I think all APs are currently running on WPA2.

Thank you VERY much, Victor!
   - Mike


From me at nevian.org  Fri Apr  1 19:01:34 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Fri, 01 Apr 2011 19:01:34 +0400
Subject: [Freeswitch-users] Why FS rewrites From header?
In-Reply-To: <BANLkTimm5nhyWEw33QeyRmpHgmhB-nmKig@mail.gmail.com>
References: <538261301575539@web100.yandex.ru>
	<AANLkTin2X8FvBhPf65rDXqDXMv8O36FwUS_Par9iPR3s@mail.gmail.com>
	<BANLkTimm5nhyWEw33QeyRmpHgmhB-nmKig@mail.gmail.com>
Message-ID: <1066421301670094@web9.yandex.ru>


Hello

Yes, this is the point!
It's a must for multi-tenancy

01.04.2011, 04:19, "I?aki Baz Castillo" <ibc at aliax.net>;:

> ?2011/3/31 Steven Ayre <steveayre at gmail.com>;;:
>> ??The aleg and bleg are 2 different separate calls, and FS joins the
>> ??signalling media on the 2.
>>
>> ??The From etc headers have to have the address of FS because that's what's
>> ??making the call.
> ?A B2BUA could handle different domains (local domains). It's common in
> ?a multidomain IP environment. Doesn't FS allow it?
> ?A SIP user is identified by a complete AoR (user and domain, like in
> ?mail world), does FS assume that just the username part is the
> ?identifier so alice at domainA.org is the same as alice at domainB.org for
> ?FS?
>
> ?--
> ?I?aki Baz Castillo
> ?<ibc at aliax.net>;;
>
> ?_______________________________________________
> ?FreeSWITCH-users mailing list
> ?FreeSWITCH-users at lists.freeswitch.org
> ?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> ?UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> ?http://www.freeswitch.org
-- 
wbr,
Serge


From me at nevian.org  Fri Apr  1 19:01:58 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Fri, 01 Apr 2011 19:01:58 +0400
Subject: [Freeswitch-users] Why FS rewrites From header?
In-Reply-To: <BANLkTimPAxvssD3kVdDa131X3eqFuXPd0Q@mail.gmail.com>
References: <538261301575539@web100.yandex.ru>
	<AANLkTin2X8FvBhPf65rDXqDXMv8O36FwUS_Par9iPR3s@mail.gmail.com>
	<BANLkTimm5nhyWEw33QeyRmpHgmhB-nmKig@mail.gmail.com>
	<BANLkTin8gut1aqo172QB=XznOGX-snx8Ww@mail.gmail.com>
	<4D955D71.90108@opensipstack.org>
	<BANLkTimPAxvssD3kVdDa131X3eqFuXPd0Q@mail.gmail.com>
Message-ID: <1066491301670118@web9.yandex.ru>


Hello

01.04.2011, 12:02, "I?aki Baz Castillo" <ibc at aliax.net>;:

> ?2011/4/1 Joegen E. Baclor <joegen at opensipstack.org>;;:
>>> ??This is:
>>>
>>> ??? alice at domainA.org ?----> ?Proxy/Registrar -----> ?FS ?----> ?same
>>> ??Proxy/Registrar ----> ?alice at domainB.org
>>>
>>> ??When alice at domainB.org receives the call, she must see
>>> ??alice at domainA.org in the From header.
>>>
>>> ??Does FS allow it?
>> ??This might be what you need -
>> ??http://wiki.freeswitch.org/wiki/Channel_Variables#sip_invite_domain
> ?That's good, thanks for pointing out it.

Yes that's good.. as a workaround I used for my and only domain but using this for every domain in config is not pretty ;)

> ?However I would prefer some option like "trust_from_domain" or
> ?something static in the profile configuration rather than the
> ?dialplan.

I'm with you and even more I wanna this to be default and have some tools in dialplan for hiding..


-- 
wbr,
Serge


From all.eforums at gmail.com  Fri Apr  1 19:03:35 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Fri, 1 Apr 2011 11:03:35 -0400
Subject: [Freeswitch-users] Build on solaris fails in libs/esl
In-Reply-To: <1882A54F-08F0-4BD3-B928-DF88240719DA@seventhsignal.de>
References: <AANLkTimGtXWdRPAv8OJYWeu8VYtCF8ZfQt0osGVgU7G=@mail.gmail.com>
	<AANLkTinab+WgVXUYtWaSccbwBV7G7LO-qjjcE4bYAKXj@mail.gmail.com>
	<AANLkTinAtxnnmbVm-tBvEuOxUQ7=z2JebB9TK+0WdBuF@mail.gmail.com>
	<AANLkTim77j3aBU0YrR8YGZEp=bDf9ND3-T0NXSznW4ov@mail.gmail.com>
	<AANLkTikJ4bYiWcsU5uLjtV4pfcL_LV1bQbqUNsfAAZu0@mail.gmail.com>
	<6ADAA4B6-802B-49F6-8C52-6BBCDE176F16@seventhsignal.de>
	<AANLkTi=ksjc+aeGHfnvbnzGhPUvkyxWCXb-Gn5wvDD0y@mail.gmail.com>
	<1882A54F-08F0-4BD3-B928-DF88240719DA@seventhsignal.de>
Message-ID: <AANLkTikSEKwqX6rQR_mjn3HU_hQmhpc7TY5uKkcYLLhZ@mail.gmail.com>

On Fri, Apr 1, 2011 at 10:47 AM, Michal Bielicki <
michal.bielicki at seventhsignal.de> wrote:


> Will take another day or two, I am a bit overloaded currently and
> automaking stuff is always a PITA :)
>
>
No worries :) I am confident that a little time spent between you and
Anthony will get it working, so I'm going by the assumption that eventually
I'll be able to build on Solaris. So I'm moving on to learn other stuff in
FS and design the rest of the platform etc. So another few days is no
biggie.

Thanks
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From joegen at opensipstack.org  Fri Apr  1 09:06:57 2011
From: joegen at opensipstack.org (Joegen E. Baclor)
Date: Fri, 01 Apr 2011 13:06:57 +0800
Subject: [Freeswitch-users] Why FS rewrites From header?
In-Reply-To: <BANLkTin8gut1aqo172QB=XznOGX-snx8Ww@mail.gmail.com>
References: <538261301575539@web100.yandex.ru>	<AANLkTin2X8FvBhPf65rDXqDXMv8O36FwUS_Par9iPR3s@mail.gmail.com>	<BANLkTimm5nhyWEw33QeyRmpHgmhB-nmKig@mail.gmail.com>
	<BANLkTin8gut1aqo172QB=XznOGX-snx8Ww@mail.gmail.com>
Message-ID: <4D955D71.90108@opensipstack.org>

On 04/01/2011 08:24 AM, I?aki Baz Castillo wrote:
> 2011/4/1 I?aki Baz Castillo<ibc at aliax.net>:
>> A B2BUA could handle different domains (local domains). It's common in
>> a multidomain IP environment. Doesn't FS allow it?
>> A SIP user is identified by a complete AoR (user and domain, like in
>> mail world), does FS assume that just the username part is the
>> identifier so alice at domainA.org is the same as alice at domainB.org for
>> FS?
> In my case I've a SIP proxy that manages different local domains, and
> I plan to put some FS boxes behind it to offer PBX services. But for
> that I need that FS understands that alice at domainA.org is a different
> user than alice at domainB.org, and when it routes back a call to the SIP
> proxy/registrar it must keep the original From URI (also the domain).
>
> This is:
>
>    alice at domainA.org  ---->  Proxy/Registrar ----->  FS  ---->  same
> Proxy/Registrar ---->  alice at domainB.org
>
> When alice at domainB.org receives the call, she must see
> alice at domainA.org in the From header.
>
> Does FS allow it?


This might be what you need - 
http://wiki.freeswitch.org/wiki/Channel_Variables#sip_invite_domain




From dunchan at freemail.hu  Fri Apr  1 12:23:02 2011
From: dunchan at freemail.hu (dunchan)
Date: Fri, 01 Apr 2011 10:23:02 +0200
Subject: [Freeswitch-users] Hangup problem, missing SIP BYE
In-Reply-To: <AANLkTinkKO97_466mRnN+AFPO+Ra7dagbasOLxQ04nNH@mail.gmail.com>
References: <4D8F53CD.7000607@freemail.hu>
	<AANLkTinkKO97_466mRnN+AFPO+Ra7dagbasOLxQ04nNH@mail.gmail.com>
Message-ID: <4D958B66.5060304@freemail.hu>

Hi!

In first of all: thanks for your answer.
But i'm confused in this situation...

The machine hostname is: debian, and my UA gets their IP address from 
the debian hosts file,
this address used as Contact header field in INVITE request
/etc/hosts:
  x.x.x.x debian

  The server real IP address is 1.2.3.4

  There are 3 cases (called party hang's up the call in all cases):
	/etc/hosts: 1.2.3.4 debian
  	UA is in the FS macine and UA's ip addr is 1.2.3.4 contact field=1.2.3.4
  	-> it doesn't get BYE messegage
  	
  	/etc/hosts: 1.2.3.4 debian
  	UA is in other machine and behind NAT ip addr=x.x.x.x contact 
field=x.x.x.x
  	-> it gets the BYE messegage
  	
  	/etc/hosts: 9.9.9.9 debian (anyone which different then 1.2.3.4)
  	UA is in the FS macine and UA's ip addr is 9.9.9.9 contact field=9.9.9.9
  	-> it gets the BYE messegage
  	
Where can i set the config to handle the internal UA same as external one?
Or where i went wrong?

thanks,
Viktor

> Do you have any NAT in place? What's the Contact header of the INVITE
> message?
>
> The BYE is a separate request to the INVITE, and so it is sent to the
> address in the Contact header from the INVITE. If you're behind NAT then
> FS might not be able to reach you at that address (e.g. if it's an
> internal IP).
>
> -Steve
>
>
> On 27 March 2011 16:12, dunchan <dunchan at freemail.hu
> <mailto:dunchan at freemail.hu>> wrote:
>
>     Hi!
>
>     I have a server machine with Freeswitch, and SIP gateway to make
>     outgoung call. The gateway has only IP address authentication.
>     I have own created UA, and if i run it from the server and the called
>     party hangs up, it don't get BYE message.
>
>     If my UA runs in different machine (other IP), it works well.
>
>     I have no idea about this. :(
>
>     I've checked the logs, and compared the two methods, and it seems there
>     are no differnces.
>     I use the newest FS.
>
>     Any suggestions?
>
>     thanks,
>     Viktor
>
>     _______________________________________________
>     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
>
>



From msc at freeswitch.org  Fri Apr  1 19:16:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 1 Apr 2011 08:16:29 -0700
Subject: [Freeswitch-users] user and public dialplan
In-Reply-To: <27ED4710EB7249FDAFB59B94444D4915@e1705>
References: <27ED4710EB7249FDAFB59B94444D4915@e1705>
Message-ID: <AANLkTinqRM9VMc9gHbga_DpLFL-06VQPkU6_8iyoSUVF@mail.gmail.com>

On Thu, Mar 31, 2011 at 10:45 PM, Madovsky <infos at madovsky.org> wrote:

>  oops, ok with sofia/default and sofia/public I think
>

I mean literally add an extension to public.xml to handle your 9999 dest
number.

-MC
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From covici at ccs.covici.com  Fri Apr  1 19:19:00 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 01 Apr 2011 11:19:00 -0400
Subject: [Freeswitch-users] cannot compile 64 bit freeswitch
In-Reply-To: <AANLkTikzjWkFpz9mO=nKsGRO0yJ152pasgaNp2RS2=+b@mail.gmail.com>
References: <23739.1301650894@ccs.covici.com>
	<AANLkTikzjWkFpz9mO=nKsGRO0yJ152pasgaNp2RS2=+b@mail.gmail.com>
Message-ID: <14122.1301671140@ccs.covici.com>

It was using the correct compiler, but this file is something which is
there and dated in September of 2010!

curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> Seems like your compiler is trying to link x86 code with x86_64 code.
> Have you tried adding this before the ./configure command?
> 
> eg. CFLAGS=-m64 CXXFLAGS=-m64 LDFLAGS=-m64 ./configure
> 
> On Fri, Apr 1, 2011 at 2:41 AM,  <covici at ccs.covici.com> wrote:
> > Here is some of what I get when i try to compile fs using 64-bit C
> > compiler and libraries -- using gentoo.
> >
> > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > i386 architecture of input file `./.libs/libsqlite3.a(complete.o)' is
> > incompatib\le with i386:x86-64 output
> > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > i386 architecture of input file `./.libs/libsqlite3.a(main.o)' is
> > incompatible w\ith i386:x86-64 output
> > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > i386 architecture of input file `./.libs/libsqlite3.a(os_unix.o)' is
> > incompatibl\e with i386:x86-64 output
> > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > i386 architecture of input file `./.libs/libsqlite3.a(prepare.o)' is
> > incompatibl\e with i386:x86-64 output
> > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > i386 architecture of input file `./.libs/libsqlite3.a(printf.o)' is
> > incompatible\ with i386:x86-64 output
> >
> >
> > There is more like that and then it dies with a number of undefined
> > references.
> >
> > How can I fix this library or what else should I do? ?I restarted from
> > ./bootstrap.sh ?./config and make.
> >
> > Any assistance would be appreciated.
> >
> > --
> > Your life is like a penny. ?You're going to lose it. ?The question is:
> > How do
> > you spend it?
> >
> > ? ? ? ? John Covici
> > ? ? ? ? covici at ccs.covici.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

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From infos at madovsky.org  Fri Apr  1 19:28:16 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 1 Apr 2011 11:28:16 -0400
Subject: [Freeswitch-users] user and public dialplan
References: <27ED4710EB7249FDAFB59B94444D4915@e1705>
	<AANLkTinqRM9VMc9gHbga_DpLFL-06VQPkU6_8iyoSUVF@mail.gmail.com>
Message-ID: <54F84FB12DA0483CA14FE8B8A4AF820F@e1705>

Mike, sometimes I mix profile and context. ;)
thanks
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Friday, April 01, 2011 11:16 AM
  Subject: Re: [Freeswitch-users] user and public dialplan





  On Thu, Mar 31, 2011 at 10:45 PM, Madovsky <infos at madovsky.org> wrote:

    oops, ok with sofia/default and sofia/public I think


  I mean literally add an extension to public.xml to handle your 9999 dest number.


  -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
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From peter.olsson at visionutveckling.se  Fri Apr  1 19:28:39 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 1 Apr 2011 17:28:39 +0200
Subject: [Freeswitch-users] cannot compile 64 bit freeswitch
In-Reply-To: <14122.1301671140@ccs.covici.com>
References: <23739.1301650894@ccs.covici.com>
	<AANLkTikzjWkFpz9mO=nKsGRO0yJ152pasgaNp2RS2=+b@mail.gmail.com>,
	<14122.1301671140@ccs.covici.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFC@cooper>

Did you clean before building (if it was not newly pulled from git)?

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r covici at ccs.covici.com [covici at ccs.covici.com]
Skickat: den 1 april 2011 17:19
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] cannot compile 64 bit freeswitch

It was using the correct compiler, but this file is something which is
there and dated in September of 2010!

curriegrad2004 <curriegrad2004 at gmail.com> wrote:

> Seems like your compiler is trying to link x86 code with x86_64 code.
> Have you tried adding this before the ./configure command?
>
> eg. CFLAGS=-m64 CXXFLAGS=-m64 LDFLAGS=-m64 ./configure
>
> On Fri, Apr 1, 2011 at 2:41 AM,  <covici at ccs.covici.com> wrote:
> > Here is some of what I get when i try to compile fs using 64-bit C
> > compiler and libraries -- using gentoo.
> >
> > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > i386 architecture of input file `./.libs/libsqlite3.a(complete.o)' is
> > incompatib\le with i386:x86-64 output
> > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > i386 architecture of input file `./.libs/libsqlite3.a(main.o)' is
> > incompatible w\ith i386:x86-64 output
> > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > i386 architecture of input file `./.libs/libsqlite3.a(os_unix.o)' is
> > incompatibl\e with i386:x86-64 output
> > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > i386 architecture of input file `./.libs/libsqlite3.a(prepare.o)' is
> > incompatibl\e with i386:x86-64 output
> > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > i386 architecture of input file `./.libs/libsqlite3.a(printf.o)' is
> > incompatible\ with i386:x86-64 output
> >
> >
> > There is more like that and then it dies with a number of undefined
> > references.
> >
> > How can I fix this library or what else should I do?  I restarted from
> > ./bootstrap.sh  ./config and make.
> >
> > Any assistance would be appreciated.
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >         John Covici
> >         covici at ccs.covici.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

--
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.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

!DSPAM:4d95ed4832767123130774!



From ibc at aliax.net  Fri Apr  1 19:37:49 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Fri, 1 Apr 2011 17:37:49 +0200
Subject: [Freeswitch-users] Why FS rewrites From header?
In-Reply-To: <1066491301670118@web9.yandex.ru>
References: <538261301575539@web100.yandex.ru>
	<AANLkTin2X8FvBhPf65rDXqDXMv8O36FwUS_Par9iPR3s@mail.gmail.com>
	<BANLkTimm5nhyWEw33QeyRmpHgmhB-nmKig@mail.gmail.com>
	<BANLkTin8gut1aqo172QB=XznOGX-snx8Ww@mail.gmail.com>
	<4D955D71.90108@opensipstack.org>
	<BANLkTimPAxvssD3kVdDa131X3eqFuXPd0Q@mail.gmail.com>
	<1066491301670118@web9.yandex.ru>
Message-ID: <BANLkTinMHRsqcKa=zoM6rZ+bjPF3nu69RQ@mail.gmail.com>

2011/4/1 Serge S. Yuriev <me at nevian.org>:
>>> ??http://wiki.freeswitch.org/wiki/Channel_Variables#sip_invite_domain
>> ?That's good, thanks for pointing out it.
>
> Yes that's good.. as a workaround I used for my and only domain but using this for every domain in config is not pretty ;)
>
>> ?However I would prefer some option like "trust_from_domain" or
>> ?something static in the profile configuration rather than the
>> ?dialplan.
>
> I'm with you and even more I wanna this to be default and have some tools in dialplan for hiding..

True. Doing that in the dialplan is a bit "hack". IMHO it should be
possible FS to entirely bypass the From URI (including port and URI
params if present).

-- 
I?aki Baz Castillo
<ibc at aliax.net>


From covici at ccs.covici.com  Fri Apr  1 19:40:30 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 01 Apr 2011 11:40:30 -0400
Subject: [Freeswitch-users] cannot compile 64 bit freeswitch
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFC@cooper>
References: <23739.1301650894@ccs.covici.com>
	<AANLkTikzjWkFpz9mO=nKsGRO0yJ152pasgaNp2RS2=+b@mail.gmail.com>,
	<14122.1301671140@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFC@cooper>
Message-ID: <15056.1301672430@ccs.covici.com>

You bet -- I even did a bootstrap.sh.  I wonder what this file is?
Should I delete it or substitute the one from the system library?

Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> Did you clean before building (if it was not newly pulled from git)?
> 
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r covici at ccs.covici.com [covici at ccs.covici.com]
> Skickat: den 1 april 2011 17:19
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] cannot compile 64 bit freeswitch
> 
> It was using the correct compiler, but this file is something which is
> there and dated in September of 2010!
> 
> curriegrad2004 <curriegrad2004 at gmail.com> wrote:
> 
> > Seems like your compiler is trying to link x86 code with x86_64 code.
> > Have you tried adding this before the ./configure command?
> >
> > eg. CFLAGS=-m64 CXXFLAGS=-m64 LDFLAGS=-m64 ./configure
> >
> > On Fri, Apr 1, 2011 at 2:41 AM,  <covici at ccs.covici.com> wrote:
> > > Here is some of what I get when i try to compile fs using 64-bit C
> > > compiler and libraries -- using gentoo.
> > >
> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > i386 architecture of input file `./.libs/libsqlite3.a(complete.o)' is
> > > incompatib\le with i386:x86-64 output
> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > i386 architecture of input file `./.libs/libsqlite3.a(main.o)' is
> > > incompatible w\ith i386:x86-64 output
> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > i386 architecture of input file `./.libs/libsqlite3.a(os_unix.o)' is
> > > incompatibl\e with i386:x86-64 output
> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > i386 architecture of input file `./.libs/libsqlite3.a(prepare.o)' is
> > > incompatibl\e with i386:x86-64 output
> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > i386 architecture of input file `./.libs/libsqlite3.a(printf.o)' is
> > > incompatible\ with i386:x86-64 output
> > >
> > >
> > > There is more like that and then it dies with a number of undefined
> > > references.
> > >
> > > How can I fix this library or what else should I do?  I restarted from
> > > ./bootstrap.sh  ./config and make.
> > >
> > > Any assistance would be appreciated.
> > >
> > > --
> > > Your life is like a penny.  You're going to lose it.  The question is:
> > > How do
> > > you spend it?
> > >
> > >         John Covici
> > >         covici at ccs.covici.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
> 
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.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
> 
> !DSPAM:4d95ed4832767123130774!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From mayamatakeshi at gmail.com  Fri Apr  1 19:46:40 2011
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Sat, 2 Apr 2011 00:46:40 +0900
Subject: [Freeswitch-users] Variable interpolation of bridge b leg
In-Reply-To: <AANLkTi=9sr3pRXbvbNtNc43JttWVvyPR=8w11esmNOiQ@mail.gmail.com>
References: <AANLkTi=ofqj3yY80U7WGVvRhD2ZJiYTB_3r=Gr9zQvoa@mail.gmail.com>
	<AANLkTikpV8kghVwtPrt4Ss=LTrZJYVvcJYH10DUwgkOE@mail.gmail.com>
	<AANLkTimhC_Qrv4oguQvjQSgbfcUwZo+MXpJuiV31gJF2@mail.gmail.com>
	<AANLkTi=9sr3pRXbvbNtNc43JttWVvyPR=8w11esmNOiQ@mail.gmail.com>
Message-ID: <AANLkTimB3XaT+Dsvw2vWOaSAqRw2VoT657kEv-0tSeYq@mail.gmail.com>

On Sat, Apr 2, 2011 at 12:14 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>> But I think there is a simple solution: I just have to use ESL to watch
>> for CHANNEL_ANSWER or set execute_on_answer=lua somescript.lua" and set the
>> filename inside the script.
>>
>>
> That works. You could also use execute_extension and set the filename
> there, since the "set" app wouldn't run until the extension is executed.
>
>
Ah yes, thanks for point that out. So it is possible to avoid starting a
script just for this.
-------------- next part --------------
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From peter.olsson at visionutveckling.se  Fri Apr  1 19:51:06 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 1 Apr 2011 17:51:06 +0200
Subject: [Freeswitch-users] cannot compile 64 bit freeswitch
In-Reply-To: <15056.1301672430@ccs.covici.com>
References: <23739.1301650894@ccs.covici.com>
	<AANLkTikzjWkFpz9mO=nKsGRO0yJ152pasgaNp2RS2=+b@mail.gmail.com>,
	<14122.1301671140@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFC@cooper>,
	<15056.1301672430@ccs.covici.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFD@cooper>

Well, bootstrap doesn't clean though.. :) Seems like it's a leftover from an old build, just remove it. I usually do "make current", which cleans, updates from git, builds and installs - all in one step.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r covici at ccs.covici.com [covici at ccs.covici.com]
Skickat: den 1 april 2011 17:40
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] cannot compile 64 bit freeswitch

You bet -- I even did a bootstrap.sh.  I wonder what this file is?
Should I delete it or substitute the one from the system library?

Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> Did you clean before building (if it was not newly pulled from git)?
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r covici at ccs.covici.com [covici at ccs.covici.com]
> Skickat: den 1 april 2011 17:19
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] cannot compile 64 bit freeswitch
>
> It was using the correct compiler, but this file is something which is
> there and dated in September of 2010!
>
> curriegrad2004 <curriegrad2004 at gmail.com> wrote:
>
> > Seems like your compiler is trying to link x86 code with x86_64 code.
> > Have you tried adding this before the ./configure command?
> >
> > eg. CFLAGS=-m64 CXXFLAGS=-m64 LDFLAGS=-m64 ./configure
> >
> > On Fri, Apr 1, 2011 at 2:41 AM,  <covici at ccs.covici.com> wrote:
> > > Here is some of what I get when i try to compile fs using 64-bit C
> > > compiler and libraries -- using gentoo.
> > >
> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > i386 architecture of input file `./.libs/libsqlite3.a(complete.o)' is
> > > incompatib\le with i386:x86-64 output
> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > i386 architecture of input file `./.libs/libsqlite3.a(main.o)' is
> > > incompatible w\ith i386:x86-64 output
> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > i386 architecture of input file `./.libs/libsqlite3.a(os_unix.o)' is
> > > incompatibl\e with i386:x86-64 output
> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > i386 architecture of input file `./.libs/libsqlite3.a(prepare.o)' is
> > > incompatibl\e with i386:x86-64 output
> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > i386 architecture of input file `./.libs/libsqlite3.a(printf.o)' is
> > > incompatible\ with i386:x86-64 output
> > >
> > >
> > > There is more like that and then it dies with a number of undefined
> > > references.
> > >
> > > How can I fix this library or what else should I do?  I restarted from
> > > ./bootstrap.sh  ./config and make.
> > >
> > > Any assistance would be appreciated.
> > >
> > > --
> > > Your life is like a penny.  You're going to lose it.  The question is:
> > > How do
> > > you spend it?
> > >
> > >         John Covici
> > >         covici at ccs.covici.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
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>          John Covici
>          covici at ccs.covici.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

--
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.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

!DSPAM:4d95f28e32763212311503!



From anthony.minessale at gmail.com  Fri Apr  1 20:12:13 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 1 Apr 2011 11:12:13 -0500
Subject: [Freeswitch-users] cannot compile 64 bit freeswitch
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFD@cooper>
References: <23739.1301650894@ccs.covici.com>
	<AANLkTikzjWkFpz9mO=nKsGRO0yJ152pasgaNp2RS2=+b@mail.gmail.com>
	<14122.1301671140@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFC@cooper>
	<15056.1301672430@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFD@cooper>
Message-ID: <AANLkTi=qgHUP9=gLPKMN0LfXiP-qvTK5d-vC+aLT4VK-@mail.gmail.com>

looks like this needs a deeper clean that is better off as a fresh checkout.
The errs show sqlite has 32 bit binaries in it.


On Fri, Apr 1, 2011 at 10:51 AM, Peter Olsson
<peter.olsson at visionutveckling.se> wrote:
> Well, bootstrap doesn't clean though.. :) Seems like it's a leftover from an old build, just remove it. I usually do "make current", which cleans, updates from git, builds and installs - all in one step.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r covici at ccs.covici.com [covici at ccs.covici.com]
> Skickat: den 1 april 2011 17:40
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] cannot compile 64 bit freeswitch
>
> You bet -- I even did a bootstrap.sh. ?I wonder what this file is?
> Should I delete it or substitute the one from the system library?
>
> Peter Olsson <peter.olsson at visionutveckling.se> wrote:
>
>> Did you clean before building (if it was not newly pulled from git)?
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r covici at ccs.covici.com [covici at ccs.covici.com]
>> Skickat: den 1 april 2011 17:19
>> Till: FreeSWITCH Users Help
>> ?mne: Re: [Freeswitch-users] cannot compile 64 bit freeswitch
>>
>> It was using the correct compiler, but this file is something which is
>> there and dated in September of 2010!
>>
>> curriegrad2004 <curriegrad2004 at gmail.com> wrote:
>>
>> > Seems like your compiler is trying to link x86 code with x86_64 code.
>> > Have you tried adding this before the ./configure command?
>> >
>> > eg. CFLAGS=-m64 CXXFLAGS=-m64 LDFLAGS=-m64 ./configure
>> >
>> > On Fri, Apr 1, 2011 at 2:41 AM, ?<covici at ccs.covici.com> wrote:
>> > > Here is some of what I get when i try to compile fs using 64-bit C
>> > > compiler and libraries -- using gentoo.
>> > >
>> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
>> > > i386 architecture of input file `./.libs/libsqlite3.a(complete.o)' is
>> > > incompatib\le with i386:x86-64 output
>> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
>> > > i386 architecture of input file `./.libs/libsqlite3.a(main.o)' is
>> > > incompatible w\ith i386:x86-64 output
>> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
>> > > i386 architecture of input file `./.libs/libsqlite3.a(os_unix.o)' is
>> > > incompatibl\e with i386:x86-64 output
>> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
>> > > i386 architecture of input file `./.libs/libsqlite3.a(prepare.o)' is
>> > > incompatibl\e with i386:x86-64 output
>> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
>> > > i386 architecture of input file `./.libs/libsqlite3.a(printf.o)' is
>> > > incompatible\ with i386:x86-64 output
>> > >
>> > >
>> > > There is more like that and then it dies with a number of undefined
>> > > references.
>> > >
>> > > How can I fix this library or what else should I do? ?I restarted from
>> > > ./bootstrap.sh ?./config and make.
>> > >
>> > > Any assistance would be appreciated.
>> > >
>> > > --
>> > > Your life is like a penny. ?You're going to lose it. ?The question is:
>> > > How do
>> > > you spend it?
>> > >
>> > > ? ? ? ? John Covici
>> > > ? ? ? ? covici at ccs.covici.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
>>
>> --
>> Your life is like a penny. ?You're going to lose it. ?The question is:
>> How do
>> you spend it?
>>
>> ? ? ? ? ?John Covici
>> ? ? ? ? ?covici at ccs.covici.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
>
> --
> Your life is like a penny. ?You're going to lose it. ?The question is:
> How do
> you spend it?
>
> ? ? ? ? John Covici
> ? ? ? ? covici at ccs.covici.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
>
> !DSPAM:4d95f28e32763212311503!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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From covici at ccs.covici.com  Fri Apr  1 20:23:36 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 01 Apr 2011 12:23:36 -0400
Subject: [Freeswitch-users] cannot compile 64 bit freeswitch
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFD@cooper>
References: <23739.1301650894@ccs.covici.com>
	<AANLkTikzjWkFpz9mO=nKsGRO0yJ152pasgaNp2RS2=+b@mail.gmail.com>,
	<14122.1301671140@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFC@cooper>,
	<15056.1301672430@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFD@cooper>
Message-ID: <24650.1301675016@ccs.covici.com>

Removing the files just causes the link to complain  that the files were
not there.

Even a ./configure does no good, its still looking for those files --
libsqlite3.a and .lai as well.

Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> Well, bootstrap doesn't clean though.. :) Seems like it's a leftover from an old build, just remove it. I usually do "make current", which cleans, updates from git, builds and installs - all in one step.
> 
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r covici at ccs.covici.com [covici at ccs.covici.com]
> Skickat: den 1 april 2011 17:40
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] cannot compile 64 bit freeswitch
> 
> You bet -- I even did a bootstrap.sh.  I wonder what this file is?
> Should I delete it or substitute the one from the system library?
> 
> Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> 
> > Did you clean before building (if it was not newly pulled from git)?
> >
> > /Peter
> > ________________________________________
> > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r covici at ccs.covici.com [covici at ccs.covici.com]
> > Skickat: den 1 april 2011 17:19
> > Till: FreeSWITCH Users Help
> > ?mne: Re: [Freeswitch-users] cannot compile 64 bit freeswitch
> >
> > It was using the correct compiler, but this file is something which is
> > there and dated in September of 2010!
> >
> > curriegrad2004 <curriegrad2004 at gmail.com> wrote:
> >
> > > Seems like your compiler is trying to link x86 code with x86_64 code.
> > > Have you tried adding this before the ./configure command?
> > >
> > > eg. CFLAGS=-m64 CXXFLAGS=-m64 LDFLAGS=-m64 ./configure
> > >
> > > On Fri, Apr 1, 2011 at 2:41 AM,  <covici at ccs.covici.com> wrote:
> > > > Here is some of what I get when i try to compile fs using 64-bit C
> > > > compiler and libraries -- using gentoo.
> > > >
> > > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > > i386 architecture of input file `./.libs/libsqlite3.a(complete.o)' is
> > > > incompatib\le with i386:x86-64 output
> > > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > > i386 architecture of input file `./.libs/libsqlite3.a(main.o)' is
> > > > incompatible w\ith i386:x86-64 output
> > > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > > i386 architecture of input file `./.libs/libsqlite3.a(os_unix.o)' is
> > > > incompatibl\e with i386:x86-64 output
> > > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > > i386 architecture of input file `./.libs/libsqlite3.a(prepare.o)' is
> > > > incompatibl\e with i386:x86-64 output
> > > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > > i386 architecture of input file `./.libs/libsqlite3.a(printf.o)' is
> > > > incompatible\ with i386:x86-64 output
> > > >
> > > >
> > > > There is more like that and then it dies with a number of undefined
> > > > references.
> > > >
> > > > How can I fix this library or what else should I do?  I restarted from
> > > > ./bootstrap.sh  ./config and make.
> > > >
> > > > Any assistance would be appreciated.
> > > >
> > > > --
> > > > Your life is like a penny.  You're going to lose it.  The question is:
> > > > How do
> > > > you spend it?
> > > >
> > > >         John Covici
> > > >         covici at ccs.covici.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
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >          John Covici
> >          covici at ccs.covici.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
> 
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
> 
>          John Covici
>          covici at ccs.covici.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
> 
> !DSPAM:4d95f28e32763212311503!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From covici at ccs.covici.com  Fri Apr  1 20:28:59 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 01 Apr 2011 12:28:59 -0400
Subject: [Freeswitch-users] cannot compile 64 bit freeswitch
In-Reply-To: <AANLkTi=qgHUP9=gLPKMN0LfXiP-qvTK5d-vC+aLT4VK-@mail.gmail.com>
References: <23739.1301650894@ccs.covici.com>
	<AANLkTikzjWkFpz9mO=nKsGRO0yJ152pasgaNp2RS2=+b@mail.gmail.com>
	<14122.1301671140@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFC@cooper>
	<15056.1301672430@ccs.covici.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFD@cooper>
	<AANLkTi=qgHUP9=gLPKMN0LfXiP-qvTK5d-vC+aLT4VK-@mail.gmail.com>
Message-ID: <24735.1301675339@ccs.covici.com>

But how ome make current never fixed it?  I will try checking out a new
tree and see what I get.

Anthony Minessale <anthony.minessale at gmail.com> wrote:

> looks like this needs a deeper clean that is better off as a fresh checkout.
> The errs show sqlite has 32 bit binaries in it.
> 
> 
> On Fri, Apr 1, 2011 at 10:51 AM, Peter Olsson
> <peter.olsson at visionutveckling.se> wrote:
> > Well, bootstrap doesn't clean though.. :) Seems like it's a leftover from an old build, just remove it. I usually do "make current", which cleans, updates from git, builds and installs - all in one step.
> >
> > /Peter
> > ________________________________________
> > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r covici at ccs.covici.com [covici at ccs.covici.com]
> > Skickat: den 1 april 2011 17:40
> > Till: FreeSWITCH Users Help
> > ?mne: Re: [Freeswitch-users] cannot compile 64 bit freeswitch
> >
> > You bet -- I even did a bootstrap.sh. ?I wonder what this file is?
> > Should I delete it or substitute the one from the system library?
> >
> > Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> >
> >> Did you clean before building (if it was not newly pulled from git)?
> >>
> >> /Peter
> >> ________________________________________
> >> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r covici at ccs.covici.com [covici at ccs.covici.com]
> >> Skickat: den 1 april 2011 17:19
> >> Till: FreeSWITCH Users Help
> >> ?mne: Re: [Freeswitch-users] cannot compile 64 bit freeswitch
> >>
> >> It was using the correct compiler, but this file is something which is
> >> there and dated in September of 2010!
> >>
> >> curriegrad2004 <curriegrad2004 at gmail.com> wrote:
> >>
> >> > Seems like your compiler is trying to link x86 code with x86_64 code.
> >> > Have you tried adding this before the ./configure command?
> >> >
> >> > eg. CFLAGS=-m64 CXXFLAGS=-m64 LDFLAGS=-m64 ./configure
> >> >
> >> > On Fri, Apr 1, 2011 at 2:41 AM, ?<covici at ccs.covici.com> wrote:
> >> > > Here is some of what I get when i try to compile fs using 64-bit C
> >> > > compiler and libraries -- using gentoo.
> >> > >
> >> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> >> > > i386 architecture of input file `./.libs/libsqlite3.a(complete.o)' is
> >> > > incompatib\le with i386:x86-64 output
> >> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> >> > > i386 architecture of input file `./.libs/libsqlite3.a(main.o)' is
> >> > > incompatible w\ith i386:x86-64 output
> >> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> >> > > i386 architecture of input file `./.libs/libsqlite3.a(os_unix.o)' is
> >> > > incompatibl\e with i386:x86-64 output
> >> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> >> > > i386 architecture of input file `./.libs/libsqlite3.a(prepare.o)' is
> >> > > incompatibl\e with i386:x86-64 output
> >> > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> >> > > i386 architecture of input file `./.libs/libsqlite3.a(printf.o)' is
> >> > > incompatible\ with i386:x86-64 output
> >> > >
> >> > >
> >> > > There is more like that and then it dies with a number of undefined
> >> > > references.
> >> > >
> >> > > How can I fix this library or what else should I do? ?I restarted from
> >> > > ./bootstrap.sh ?./config and make.
> >> > >
> >> > > Any assistance would be appreciated.
> >> > >
> >> > > --
> >> > > Your life is like a penny. ?You're going to lose it. ?The question is:
> >> > > How do
> >> > > you spend it?
> >> > >
> >> > > ? ? ? ? John Covici
> >> > > ? ? ? ? covici at ccs.covici.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
> >>
> >> --
> >> Your life is like a penny. ?You're going to lose it. ?The question is:
> >> How do
> >> you spend it?
> >>
> >> ? ? ? ? ?John Covici
> >> ? ? ? ? ?covici at ccs.covici.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
> >
> > --
> > Your life is like a penny. ?You're going to lose it. ?The question is:
> > How do
> > you spend it?
> >
> > ? ? ? ? John Covici
> > ? ? ? ? covici at ccs.covici.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
> >
> > !DSPAM:4d95f28e32763212311503!
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> 
> 
> -- 
> Anthony Minessale II
> 
> 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
> 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

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com


From infos at madovsky.org  Fri Apr  1 20:32:36 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 1 Apr 2011 12:32:36 -0400
Subject: [Freeswitch-users] cannot compile 64 bit freeswitch
References: <23739.1301650894@ccs.covici.com><AANLkTikzjWkFpz9mO=nKsGRO0yJ152pasgaNp2RS2=+b@mail.gmail.com>,
	<14122.1301671140@ccs.covici.com><549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFC@cooper>,
	<15056.1301672430@ccs.covici.com><549CFEF87AEDE841A38E9D15EAB4C04C58BE186BFD@cooper>
	<24650.1301675016@ccs.covici.com>
Message-ID: <568493FE356240749BA62F025B0C1C5C@e1705>

in some linux distrib sometimes there are lib path into /lib64 or /lib that
cause trouble

----- Original Message ----- 
From: <covici at ccs.covici.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, April 01, 2011 12:23 PM
Subject: Re: [Freeswitch-users] cannot compile 64 bit freeswitch


Removing the files just causes the link to complain  that the files were
not there.

Even a ./configure does no good, its still looking for those files --
libsqlite3.a and .lai as well.

Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> Well, bootstrap doesn't clean though.. :) Seems like it's a leftover from 
> an old build, just remove it. I usually do "make current", which cleans, 
> updates from git, builds and installs - all in one step.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r 
> covici at ccs.covici.com [covici at ccs.covici.com]
> Skickat: den 1 april 2011 17:40
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] cannot compile 64 bit freeswitch
>
> You bet -- I even did a bootstrap.sh.  I wonder what this file is?
> Should I delete it or substitute the one from the system library?
>
> Peter Olsson <peter.olsson at visionutveckling.se> wrote:
>
> > Did you clean before building (if it was not newly pulled from git)?
> >
> > /Peter
> > ________________________________________
> > Fr?n: freeswitch-users-bounces at lists.freeswitch.org 
> > [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r 
> > covici at ccs.covici.com [covici at ccs.covici.com]
> > Skickat: den 1 april 2011 17:19
> > Till: FreeSWITCH Users Help
> > ?mne: Re: [Freeswitch-users] cannot compile 64 bit freeswitch
> >
> > It was using the correct compiler, but this file is something which is
> > there and dated in September of 2010!
> >
> > curriegrad2004 <curriegrad2004 at gmail.com> wrote:
> >
> > > Seems like your compiler is trying to link x86 code with x86_64 code.
> > > Have you tried adding this before the ./configure command?
> > >
> > > eg. CFLAGS=-m64 CXXFLAGS=-m64 LDFLAGS=-m64 ./configure
> > >
> > > On Fri, Apr 1, 2011 at 2:41 AM,  <covici at ccs.covici.com> wrote:
> > > > Here is some of what I get when i try to compile fs using 64-bit C
> > > > compiler and libraries -- using gentoo.
> > > >
> > > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > > i386 architecture of input file `./.libs/libsqlite3.a(complete.o)' 
> > > > is
> > > > incompatib\le with i386:x86-64 output
> > > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > > i386 architecture of input file `./.libs/libsqlite3.a(main.o)' is
> > > > incompatible w\ith i386:x86-64 output
> > > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > > i386 architecture of input file `./.libs/libsqlite3.a(os_unix.o)' is
> > > > incompatibl\e with i386:x86-64 output
> > > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > > i386 architecture of input file `./.libs/libsqlite3.a(prepare.o)' is
> > > > incompatibl\e with i386:x86-64 output
> > > > /usr/lib/gcc/x86_64-pc-linux-gnu/4.4.5/../../../../x86_64-pc-linux-gnu/bin/ld:
> > > > i386 architecture of input file `./.libs/libsqlite3.a(printf.o)' is
> > > > incompatible\ with i386:x86-64 output
> > > >
> > > >
> > > > There is more like that and then it dies with a number of undefined
> > > > references.
> > > >
> > > > How can I fix this library or what else should I do?  I restarted 
> > > > from
> > > > ./bootstrap.sh  ./config and make.
> > > >
> > > > Any assistance would be appreciated.
> > > >
> > > > --
> > > > Your life is like a penny.  You're going to lose it.  The question 
> > > > is:
> > > > How do
> > > > you spend it?
> > > >
> > > >         John Covici
> > > >         covici at ccs.covici.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
> >
> > --
> > Your life is like a penny.  You're going to lose it.  The question is:
> > How do
> > you spend it?
> >
> >          John Covici
> >          covici at ccs.covici.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
>
> --
> Your life is like a penny.  You're going to lose it.  The question is:
> How do
> you spend it?
>
>          John Covici
>          covici at ccs.covici.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
> !DSPAM:4d95f28e32763212311503!
>
>
> _______________________________________________
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
Your life is like a penny.  You're going to lose it.  The question is:
How do
you spend it?

         John Covici
         covici at ccs.covici.com

_______________________________________________
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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  Fri Apr  1 22:00:57 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 1 Apr 2011 14:00:57 -0400
Subject: [Freeswitch-users] invite in conference
Message-ID: <07BED634C983425880DEB73E865D0BCE@e1705>

also moh-sound param
helps to increase dramatically the latency, especially if mod_shout is used
so if I don't use any moh-sound param the latency is much better, but the 
result of tests below are the same, after X invites/hangup the audio conference into the invited phone
gets more and more latency

Thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Thursday, March 31, 2011 9:16 PM
  Subject: Re: [Freeswitch-users] invite in conference


  I have more info after dozen different tests.
  If I invite in conf the same number several time, each time
  the invited leg answers, like 1 second of latency is added (exponential)
  so after 3 invites hangups I got 8 seconds of latency for the conference moderator voice
  in the invited phone. concerning the invited voice into the conference the latency stays exactly the same after 3 invites.
  Sorry I didn't triy to do it with 3 different numbers as my cell is cut (no credits) and my landline phone also (bill due).. ;)

  Thanks
    ----- Original Message ----- 
    From: Madovsky 
    To: FreeSWITCH Users Help 
    Sent: Wednesday, March 30, 2011 6:27 PM
    Subject: Re: [Freeswitch-users] invite in conference


    /usr/local/freeswitch/bin/fs_cli -x "conference confText dial\{inconf=true,originate_timeout=20,ignore_early_media=true,instant_ringback=true}user/11111 22222 hiConf"

    and

     /usr/local/freeswitch/bin/fs_cli -x "conference confText dial\{inconf=true,originate_timeout=20,ignore_early_media=true,instant_ringback=true}loopback/11111 22222 hiConf"

    is this dial event can be in other place that conference::maintenance in ESL ?

    thanks

      ----- Original Message ----- 
      From: Michael Collins 
      To: FreeSWITCH Users Help 
      Sent: Wednesday, March 30, 2011 1:20 PM
      Subject: Re: [Freeswitch-users] invite in conference


      what syntax are you using for the invitation? I would like to try it on my system and see if i can reproduce. 
      -MC


      On Wed, Mar 30, 2011 at 9:27 AM, Madovsky <infos at madovsky.org> wrote:

        When I invite in conference, I can't see any conference esl event of the new member invited and accepted in conference
        is it normal ?

        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






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


      _______________________________________________
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From anthony.minessale at gmail.com  Fri Apr  1 23:45:33 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 1 Apr 2011 14:45:33 -0500
Subject: [Freeswitch-users] Why FS rewrites From header?
In-Reply-To: <BANLkTinMHRsqcKa=zoM6rZ+bjPF3nu69RQ@mail.gmail.com>
References: <538261301575539@web100.yandex.ru>
	<AANLkTin2X8FvBhPf65rDXqDXMv8O36FwUS_Par9iPR3s@mail.gmail.com>
	<BANLkTimm5nhyWEw33QeyRmpHgmhB-nmKig@mail.gmail.com>
	<BANLkTin8gut1aqo172QB=XznOGX-snx8Ww@mail.gmail.com>
	<4D955D71.90108@opensipstack.org>
	<BANLkTimPAxvssD3kVdDa131X3eqFuXPd0Q@mail.gmail.com>
	<1066491301670118@web9.yandex.ru>
	<BANLkTinMHRsqcKa=zoM6rZ+bjPF3nu69RQ@mail.gmail.com>
Message-ID: <AANLkTinyqyXadnhF77KJRS1iMcJR51Pkwi-W_nNomRKy@mail.gmail.com>

Everyone wants the way they want it to work in their specific single
use case to be the default.

It's not a hack, it's the way you want it solved by a documented
config option and its not any more ugly than a cisco dial-plan is it?

FreeSWITCH can be mostly anything you want it to be, besides a proxy.
It's your job to configure it how you would like.

For your connivence, latest git has a new option you can specify in
the from-domain param on a gateway xml to "auto-aleg" indicating you
want this behavior that in now way should be the default.......



On Fri, Apr 1, 2011 at 10:37 AM, I?aki Baz Castillo <ibc at aliax.net> wrote:
> 2011/4/1 Serge S. Yuriev <me at nevian.org>:
>>>> ??http://wiki.freeswitch.org/wiki/Channel_Variables#sip_invite_domain
>>> ?That's good, thanks for pointing out it.
>>
>> Yes that's good.. as a workaround I used for my and only domain but using this for every domain in config is not pretty ;)
>>
>>> ?However I would prefer some option like "trust_from_domain" or
>>> ?something static in the profile configuration rather than the
>>> ?dialplan.
>>
>> I'm with you and even more I wanna this to be default and have some tools in dialplan for hiding..
>
> True. Doing that in the dialplan is a bit "hack". IMHO it should be
> possible FS to entirely bypass the From URI (including port and URI
> params if present).
>
> --
> I?aki Baz Castillo
> <ibc at aliax.net>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From infos at madovsky.org  Sat Apr  2 00:17:25 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 1 Apr 2011 16:17:25 -0400
Subject: [Freeswitch-users] originate from cli
Message-ID: <F38382DFFBE041AF8562F02269FE45EC@e1705>

I make some test with originate from cli.

/usr/local/freeswitch/bin/fs_cli -x "originate user/9999 7777 XML public"
-ERR RECOVERY_ON_TIMER_EXPIRE

9999 and 7777 are ready to receive calls and I don't have any NAT.

is anyone can explain what it means ?

Thanks
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From anthony.minessale at gmail.com  Sat Apr  2 00:53:10 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 1 Apr 2011 15:53:10 -0500
Subject: [Freeswitch-users] originate from cli
In-Reply-To: <F38382DFFBE041AF8562F02269FE45EC@e1705>
References: <F38382DFFBE041AF8562F02269FE45EC@e1705>
Message-ID: <AANLkTim=_=O_1Vyw9OmduJuY_Fenv3W=nKCzgc=L9-ef@mail.gmail.com>

turn on the sip trace and look at the logs


On Fri, Apr 1, 2011 at 3:17 PM, Madovsky <infos at madovsky.org> wrote:
> I make some test with originate from cli.
>
> /usr/local/freeswitch/bin/fs_cli -x "originate user/9999 7777 XML public"
> -ERR RECOVERY_ON_TIMER_EXPIRE
>
> 9999 and 7777 are ready to receive calls and I don't have any NAT.
>
> is anyone can explain what it means ?
>
> 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
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From steveayre at gmail.com  Sat Apr  2 01:29:50 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 1 Apr 2011 22:29:50 +0100
Subject: [Freeswitch-users] originate from cli
In-Reply-To: <F38382DFFBE041AF8562F02269FE45EC@e1705>
References: <F38382DFFBE041AF8562F02269FE45EC@e1705>
Message-ID: <AANLkTiks_xJX8vyX-QWZ+ar+Bwhn0Mm6TPvcp4=_39DC@mail.gmail.com>

It means FS sent a message and didn't get a reply (timed out).

As anthm says, look at the siptrace - that'll show you what's being sent /
received.

-Steve


On 1 April 2011 21:17, Madovsky <infos at madovsky.org> wrote:

>  I make some test with originate from cli.
>
> /usr/local/freeswitch/bin/fs_cli -x "originate user/9999 7777 XML public"
> -ERR RECOVERY_ON_TIMER_EXPIRE
>
> 9999 and 7777 are ready to receive calls and I don't have any NAT.
>
> is anyone can explain what it means ?
>
> 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
>
>
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From steveayre at gmail.com  Sat Apr  2 01:30:21 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 1 Apr 2011 22:30:21 +0100
Subject: [Freeswitch-users] originate from cli
In-Reply-To: <AANLkTiks_xJX8vyX-QWZ+ar+Bwhn0Mm6TPvcp4=_39DC@mail.gmail.com>
References: <F38382DFFBE041AF8562F02269FE45EC@e1705>
	<AANLkTiks_xJX8vyX-QWZ+ar+Bwhn0Mm6TPvcp4=_39DC@mail.gmail.com>
Message-ID: <AANLkTik3or8JqMUmMwaYxgSumjD+Hm+HBtCh_WLusk6=@mail.gmail.com>

It can also be that the other side sent FS a reply saying that *it* had
timed out. siptrace will also show if that's the case.


On 1 April 2011 22:29, Steven Ayre <steveayre at gmail.com> wrote:

> It means FS sent a message and didn't get a reply (timed out).
>
> As anthm says, look at the siptrace - that'll show you what's being sent /
> received.
>
> -Steve
>
>
> On 1 April 2011 21:17, Madovsky <infos at madovsky.org> wrote:
>
>>  I make some test with originate from cli.
>>
>> /usr/local/freeswitch/bin/fs_cli -x "originate user/9999 7777 XML public"
>> -ERR RECOVERY_ON_TIMER_EXPIRE
>>
>> 9999 and 7777 are ready to receive calls and I don't have any NAT.
>>
>> is anyone can explain what it means ?
>>
>> 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
>>
>>
>
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From infos at madovsky.org  Sat Apr  2 03:58:25 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 1 Apr 2011 19:58:25 -0400
Subject: [Freeswitch-users] originate from cli
References: <F38382DFFBE041AF8562F02269FE45EC@e1705><AANLkTiks_xJX8vyX-QWZ+ar+Bwhn0Mm6TPvcp4=_39DC@mail.gmail.com>
	<AANLkTik3or8JqMUmMwaYxgSumjD+Hm+HBtCh_WLusk6=@mail.gmail.com>
Message-ID: <A702E4D3569A4006877B147C15F5C828@e1705>

I tried the command in a cluster environment
so 9999 can be on node1 and 7777 on node2

here is the log

2011-04-01 19:54:45.192621 [DEBUG] switch_ivr_originate.c:1973 variable string 0 = [presence_id=9999 at boophone.com]
2011-04-01 19:54:45.193638 [NOTICE] switch_channel.c:812 New Channel sofia/internal/sip:9999 at 192.168.0.18:58251 [43cf3947-e9b3-4828-befc-e284dc4a9e3a]
2011-04-01 19:54:45.193638 [DEBUG] mod_sofia.c:4286 (sofia/internal/sip:9999 at 192.168.0.18:58251) State Change CS_NEW -> CS_INIT
2011-04-01 19:54:45.193638 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:9999 at 192.168.0.18:58251 [BREAK]
2011-04-01 19:54:45.193638 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:9999 at 192.168.0.18:58251) Running State Change CS_INIT
2011-04-01 19:54:45.193638 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/sip:9999 at 192.168.0.18:58251) State INIT
2011-04-01 19:54:45.193638 [DEBUG] mod_sofia.c:84 sofia/internal/sip:9999 at 192.168.0.18:58251 SOFIA INIT
2011-04-01 19:54:45.194651 [DEBUG] mod_sofia.c:124 (sofia/internal/sip:9999 at 192.168.0.18:58251) State Change CS_INIT -> CS_ROUTING
2011-04-01 19:54:45.194651 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:9999 at 192.168.0.18:58251 [BREAK]
2011-04-01 19:54:45.194651 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/sip:9999 at 192.168.0.18:58251) State INIT going to sleep
2011-04-01 19:54:45.194651 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:9999 at 192.168.0.18:58251) Running State Change CS_ROUTING
2011-04-01 19:54:45.194651 [DEBUG] switch_channel.c:1668 (sofia/internal/sip:9999 at 192.168.0.18:58251) Callstate Change DOWN -> RINGING
2011-04-01 19:54:45.194651 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/sip:9999 at 192.168.0.18:58251) State ROUTING
2011-04-01 19:54:45.194651 [DEBUG] mod_sofia.c:147 sofia/internal/sip:9999 at 192.168.0.18:58251 SOFIA ROUTING
2011-04-01 19:54:45.194651 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:9999 at 192.168.0.18:58251) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-04-01 19:54:45.194651 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:9999 at 192.168.0.18:58251 [BREAK]
2011-04-01 19:54:45.194651 [DEBUG] switch_core_state_machine.c:359 (sofia/internal/sip:9999 at 192.168.0.18:58251) State ROUTING going to sleep
2011-04-01 19:54:45.194651 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:9999 at 192.168.0.18:58251) Running State Change CS_CONSUME_MEDIA
2011-04-01 19:54:45.194651 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/sip:9999 at 192.168.0.18:58251) State CONSUME_MEDIA
2011-04-01 19:54:45.194651 [DEBUG] switch_core_state_machine.c:378 (sofia/internal/sip:9999 at 192.168.0.18:58251) State CONSUME_MEDIA going to sleep
2011-04-01 19:54:45.195670 [DEBUG] sofia.c:4754 Channel sofia/internal/sip:9999 at 192.168.0.18:58251 entering state [calling][0]
2011-04-01 19:54:48.856682 [DEBUG] mod_nibblebill.c:572 Received request via SESSION_HEARTBEAT!
2011-04-01 19:54:48.856682 [DEBUG] mod_nibblebill.c:433 Attempting to bill at $0.03 per minute to account 9999
2011-04-01 19:54:48.856682 [DEBUG] mod_nibblebill.c:491 60 seconds passed since last bill time of 2011-04-01 19:53:48
2011-04-01 19:54:48.856682 [DEBUG] mod_nibblebill.c:498 Billing $0.030010 to 9999 (Call: 704fa1fc-7b97-40b3-ad16-b5fc9da6d667 / 0.134583 so far)
2011-04-01 19:54:48.856682 [DEBUG] mod_nibblebill.c:321 Doing update query
[UPDATE accounts SET cash=cash-0.030010 WHERE id='9999']
2011-04-01 19:54:48.944981 [DEBUG] mod_nibblebill.c:366 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='9999']
2011-04-01 19:54:48.951074 [DEBUG] mod_nibblebill.c:376 Retrieved current balance for account 9999 (balance = 10.105113)
2011-04-01 19:55:01.857456 [DEBUG] mod_nibblebill.c:572 Received request via SESSION_HEARTBEAT!
2011-04-01 19:55:01.857456 [DEBUG] mod_nibblebill.c:433 Attempting to bill at $0.03 per minute to account 9999
2011-04-01 19:55:01.857456 [DEBUG] mod_nibblebill.c:491 60 seconds passed since last bill time of 2011-04-01 19:54:01
2011-04-01 19:55:01.857456 [DEBUG] mod_nibblebill.c:498 Billing $0.030010 to 9999 (Call: 2879366a-0478-4480-9977-8b22847be252 / 0.150150 so far)
2011-04-01 19:55:01.857456 [DEBUG] mod_nibblebill.c:321 Doing update query
[UPDATE accounts SET cash=cash-0.030010 WHERE id='9999']
2011-04-01 19:55:01.943873 [DEBUG] mod_nibblebill.c:366 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='9999']
2011-04-01 19:55:01.949965 [DEBUG] mod_nibblebill.c:376 Retrieved current balance for account 9999 (balance = 10.075103)
2011-04-01 19:55:17.196147 [DEBUG] sofia.c:4754 Channel sofia/internal/sip:9999 at 192.168.0.18:58251 entering state [terminated][408]
2011-04-01 19:55:17.196147 [DEBUG] switch_channel.c:2563 (sofia/internal/sip:9999 at 192.168.0.18:58251) Callstate Change RINGING -> HANGUP
2011-04-01 19:55:17.196147 [NOTICE] sofia.c:5394 Hangup sofia/internal/sip:9999 at 192.168.0.18:58251 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]
2011-04-01 19:55:17.196147 [DEBUG] switch_channel.c:2579 Send signal sofia/internal/sip:9999 at 192.168.0.18:58251 [KILL]
2011-04-01 19:55:17.196147 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:9999 at 192.168.0.18:58251 [BREAK]
2011-04-01 19:55:17.196147 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:9999 at 192.168.0.18:58251) Running State Change CS_HANGUP
2011-04-01 19:55:17.196147 [DEBUG] switch_core_state_machine.c:560 (sofia/internal/sip:9999 at 192.168.0.18:58251) State HANGUP
2011-04-01 19:55:17.196147 [DEBUG] mod_sofia.c:451 sofia/internal/sip:9999 at 192.168.0.18:58251 Overriding SIP cause 504 with 408 from the other leg
2011-04-01 19:55:17.196147 [DEBUG] mod_sofia.c:457 Channel sofia/internal/sip:9999 at 192.168.0.18:58251 hanging up, cause: RECOVERY_ON_TIMER_EXPIRE
2011-04-01 19:55:17.204366 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:9999 at 192.168.0.18:58251 Standard HANGUP, cause: RECOVERY_ON_TIMER_EXPIRE
2011-04-01 19:55:17.204366 [DEBUG] switch_core_state_machine.c:560 (sofia/internal/sip:9999 at 192.168.0.18:58251) State HANGUP going to sleep
2011-04-01 19:55:17.204366 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/sip:9999 at 192.168.0.18:58251) State Change CS_HANGUP -> CS_REPORTING
2011-04-01 19:55:17.204366 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:9999 at 192.168.0.18:58251 [BREAK]
2011-04-01 19:55:17.204366 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:9999 at 192.168.0.18:58251) Running State Change CS_REPORTING
2011-04-01 19:55:17.204366 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/sip:9999 at 192.168.0.18:58251) State REPORTING
2011-04-01 19:55:17.204366 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:9999 at 192.168.0.18:58251 Standard REPORTING, cause: RECOVERY_ON_TIMER_EXPIRE
2011-04-01 19:55:17.204366 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/sip:9999 at 192.168.0.18:58251) State REPORTING going to sleep
2011-04-01 19:55:17.204366 [DEBUG] switch_core_state_machine.c:345 (sofia/internal/sip:9999 at 192.168.0.18:58251) State Change CS_REPORTING -> CS_DESTROY
2011-04-01 19:55:17.204366 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:9999 at 192.168.0.18:58251 [BREAK]
2011-04-01 19:55:17.204366 [DEBUG] switch_core_session.c:1288 Session 79 (sofia/internal/sip:9999 at 192.168.0.18:58251) Locked, Waiting on external entities
2011-04-01 19:55:17.206396 [DEBUG] switch_ivr_originate.c:3506 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
2011-04-01 19:55:17.206396 [ERR] switch_ivr_originate.c:2640 Cannot create outgoing channel of type [user] cause: [RECOVERY_ON_TIMER_EXPIRE]
2011-04-01 19:55:17.206396 [NOTICE] switch_core_session.c:1306 Session 79 (sofia/internal/sip:9999 at 192.168.0.18:58251) Ended
2011-04-01 19:55:17.206396 [DEBUG] switch_ivr_originate.c:3506 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
2011-04-01 19:55:17.206396 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/sip:9999 at 192.168.0.18:58251 [CS_DESTROY]
2011-04-01 19:55:17.206396 [DEBUG] switch_core_state_machine.c:449 (sofia/internal/sip:9999 at 192.168.0.18:58251) Callstate Change HANGUP -> DOWN
2011-04-01 19:55:17.206396 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/sip:9999 at 192.168.0.18:58251) Running State Change CS_DESTROY
2011-04-01 19:55:17.206396 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/sip:9999 at 192.168.0.18:58251) State DESTROY
2011-04-01 19:55:17.206396 [DEBUG] mod_sofia.c:362 sofia/internal/sip:9999 at 192.168.0.18:58251 SOFIA DESTROY
2011-04-01 19:55:17.206396 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:9999 at 192.168.0.18:58251 Standard DESTROY
2011-04-01 19:55:17.206396 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/sip:9999 at 192.168.0.18:58251) State DESTROY going to sleep


thanks

  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Friday, April 01, 2011 5:30 PM
  Subject: Re: [Freeswitch-users] originate from cli


  It can also be that the other side sent FS a reply saying that *it* had timed out. siptrace will also show if that's the case.



  On 1 April 2011 22:29, Steven Ayre <steveayre at gmail.com> wrote:

    It means FS sent a message and didn't get a reply (timed out).

    As anthm says, look at the siptrace - that'll show you what's being sent / received.

    -Steve



    On 1 April 2011 21:17, Madovsky <infos at madovsky.org> wrote:

      I make some test with originate from cli.

      /usr/local/freeswitch/bin/fs_cli -x "originate user/9999 7777 XML public"
      -ERR RECOVERY_ON_TIMER_EXPIRE

      9999 and 7777 are ready to receive calls and I don't have any NAT.

      is anyone can explain what it means ?

      Thanks
       


      _______________________________________________
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      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From casteven at gmail.com  Sat Apr  2 06:40:35 2011
From: casteven at gmail.com (Campbell Steven)
Date: Sat, 02 Apr 2011 15:40:35 +1300
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <4D95E839.2070403@KennedySoftware.ie>
References: <4D94C282.1090903@KennedySoftware.ie>
	<4D951D45.5010005@utoronto.ca>  <4D95E839.2070403@KennedySoftware.ie>
Message-ID: <1301712035.18009.1048.camel@macmini>

The Snom 870 will do it with a USB Wifi dongle, but in my experience
don't go there, they are a diabolical handset from a usability
standpoint.

Campbell

On Fri, 2011-04-01 at 15:59 +0100, Michael Kennedy wrote:

> Victor,
> 
> > A bit off-topic but here are my 50 cents:
> 
> Oopppssss, my apologies - I thought it might be a common query for folks 
> thinking about FS - but maybe in another "list"?
> 
> > -Did you consider building a wireless bridge with a $40 WiFi router
> > running DD-WRT/Tomato/OpenWRT etc?
> 
> I did NOT - and I've deployed a lot of them to support "PC"s! THANK YOU!
> 
> > This way you can plug wired phones into LAN ports of the "bridge" and
> > the router will bridge them to your main access point.
> > Asus WL-520GU will work and is really cheap.
> 
> EXCELLENT suggestion!
> 
> (Maybe I'm drifting even more O-T, but... I'm also glad you did not 
> mention WiFi devices from Linksys - in my experience, some of these 
> boxes performed very poorly, but I seem to be the only one on the planet 
> with these experiences!).
> 
> > -If you go with WiFi you should only use WPA or WPA2.
> > Less secure options (WEP :-) ) make all conversations accessible to public.
> 
> Yes, I think all APs are currently running on WPA2.
> 
> Thank you VERY much, Victor!
>    - 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


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From fieldpeak at gmail.com  Sat Apr  2 06:49:51 2011
From: fieldpeak at gmail.com (Charles)
Date: Sat, 2 Apr 2011 10:49:51 +0800
Subject: [Freeswitch-users] How to realize -GIT pull latest
	versiontoalocal copy and work with prevoius changes
References: <4d953258.c4b3ec0a.5e9b.5c5a@mx.google.com>,
	<1301624326796-6229497.post@n2.nabble.com>,
	<4d953c7d.0a3fec0a.4e36.5a1e@mx.google.com>,
	<4d95648a.204b640a.6598.7377@mx.google.com>,
	<1301666188189-6230871.post@n2.nabble.com>
Message-ID: <4d968ed2.29d6e70a.5685.7f52@mx.google.com>

Thanks Jeff, understood.

Regards,
Charles

2011-04-02 



Charles 



???? Jeff Lenk 
????? 2011-04-01  21:57:37 
???? freeswitch-users 
??? 
??? Re: [Freeswitch-users] How to realize -GIT pull latest versiontoalocal copy and work with prevoius changes 
 
Hi Charles,
You sure can do that. Thats what git it for. Just be aware if you have local
changes and have problems it may be more difficult for "you" to determine
whether your modifications or the base code possibly is at fault. Potential
problems in the fs base code have to be reproduced with git head with no
other modifications or being able to demonstrate clearly what the problem is
and then submitting diffs against git head.
Jeff
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-realize-GIT-pull-latest-version-to-a-local-copy-and-work-with-prevoius-changes-tp6229486p6230871.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
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From infos at madovsky.org  Sat Apr  2 07:37:14 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 1 Apr 2011 23:37:14 -0400
Subject: [Freeswitch-users] originate from cli
References: <F38382DFFBE041AF8562F02269FE45EC@e1705><AANLkTiks_xJX8vyX-QWZ+ar+Bwhn0Mm6TPvcp4=_39DC@mail.gmail.com>
	<AANLkTik3or8JqMUmMwaYxgSumjD+Hm+HBtCh_WLusk6=@mail.gmail.com>
Message-ID: <C66F263892AE4887B47D8D6B0B3FA4D0@e1705>

sorry the sip trace :
after this command:

/usr/local/freeswitch/bin/fs_cli -x "originate user/9999999999 1111111111"

------------------------------------------------------------------------
2011-04-01 23:33:39.673581 [DEBUG] sofia.c:4754 Channel sofia/internal/sip:99999999999 at 11.22.33.44:52767 entering state [calling][0]
send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:40.673952:
------------------------------------------------------------------------
INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
Max-Forwards: 70
From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
To: <sip:99999999999 at 11.22.33.44:52767>
Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
CSeq: 10521545 INVITE
Contact: <sip:mod_sofia at 11.22.33.44:5080>
User-Agent: CiscoSystems-SIP-GW-UA
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 492
X-FS-Support: update_display
Remote-Party-ID: <sip:0000000000 at 11.22.33.44>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
s=FreeSWITCH
c=IN IP4 11.22.33.44
t=0 0
m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
a=rtpmap:98 SPEEX/16000
a=rtpmap:99 G726-16/8000
a=rtpmap:100 G726-24/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:103 G726-40/8000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 iLBC/8000
a=fmtp:105 mode=20
a=rtpmap:106 L16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
------------------------------------------------------------------------
send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:42.673971:
------------------------------------------------------------------------
INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
Max-Forwards: 70
From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
To: <sip:99999999999 at 11.22.33.44:52767>
Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
CSeq: 10521545 INVITE
Contact: <sip:mod_sofia at 11.22.33.44:5080>
User-Agent: CiscoSystems-SIP-GW-UA
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 492
X-FS-Support: update_display
Remote-Party-ID: <sip:0000000000 at 11.22.33.44>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
s=FreeSWITCH
c=IN IP4 11.22.33.44
t=0 0
m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
a=rtpmap:98 SPEEX/16000
a=rtpmap:99 G726-16/8000
a=rtpmap:100 G726-24/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:103 G726-40/8000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 iLBC/8000
a=fmtp:105 mode=20
a=rtpmap:106 L16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
------------------------------------------------------------------------
send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:46.673944:
------------------------------------------------------------------------
INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
Max-Forwards: 70
From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
To: <sip:99999999999 at 11.22.33.44:52767>
Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
CSeq: 10521545 INVITE
Contact: <sip:mod_sofia at 11.22.33.44:5080>
User-Agent: CiscoSystems-SIP-GW-UA
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 492
X-FS-Support: update_display
Remote-Party-ID: <sip:0000000000 at 11.22.33.44>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
s=FreeSWITCH
c=IN IP4 11.22.33.44
t=0 0
m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
a=rtpmap:98 SPEEX/16000
a=rtpmap:99 G726-16/8000
a=rtpmap:100 G726-24/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:103 G726-40/8000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 iLBC/8000
a=fmtp:105 mode=20
a=rtpmap:106 L16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
------------------------------------------------------------------------
send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:54.673950:
------------------------------------------------------------------------
INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
Max-Forwards: 70
From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
To: <sip:99999999999 at 11.22.33.44:52767>
Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
CSeq: 10521545 INVITE
Contact: <sip:mod_sofia at 11.22.33.44:5080>
User-Agent: CiscoSystems-SIP-GW-UA
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 492
X-FS-Support: update_display
Remote-Party-ID: <sip:0000000000 at 11.22.33.44>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
s=FreeSWITCH
c=IN IP4 11.22.33.44
t=0 0
m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
a=rtpmap:98 SPEEX/16000
a=rtpmap:99 G726-16/8000
a=rtpmap:100 G726-24/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:103 G726-40/8000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 iLBC/8000
a=fmtp:105 mode=20
a=rtpmap:106 L16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
------------------------------------------------------------------------
send 1385 bytes to udp/[11.22.33.44]:52767 at 03:34:10.674077:
------------------------------------------------------------------------
INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
Max-Forwards: 70
From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
To: <sip:99999999999 at 11.22.33.44:52767>
Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
CSeq: 10521545 INVITE
Contact: <sip:mod_sofia at 11.22.33.44:5080>
User-Agent: CiscoSystems-SIP-GW-UA
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 492
X-FS-Support: update_display
Remote-Party-ID: <sip:0000000000 at 11.22.33.44>;party=calling;screen=yes;privacy=off

v=0
o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
s=FreeSWITCH
c=IN IP4 11.22.33.44
t=0 0
m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
a=rtpmap:98 SPEEX/16000
a=rtpmap:99 G726-16/8000
a=rtpmap:100 G726-24/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:103 G726-40/8000
a=rtpmap:104 G7221/16000
a=fmtp:104 bitrate=32000
a=rtpmap:105 iLBC/8000
a=fmtp:105 mode=20
a=rtpmap:106 L16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
------------------------------------------------------------------------
2011-04-01 23:34:11.674022 [DEBUG] sofia.c:4754 Channel sofia/internal/sip:99999999999 at 11.22.33.44:52767 entering state [terminated][408]
2011-04-01 23:34:11.674022 [DEBUG] switch_channel.c:2563 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) Callstate Change RINGING -> HANGUP
2011-04-01 23:34:11.674022 [NOTICE] sofia.c:5394 Hangup sofia/internal/sip:99999999999 at 11.22.33.44:52767 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]
2011-04-01 23:34:11.674022 [DEBUG] switch_channel.c:2579 Send signal sofia/internal/sip:99999999999 at 11.22.33.44:52767 [KILL]
2011-04-01 23:34:11.674022 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:99999999999 at 11.22.33.44:52767 [BREAK]
2011-04-01 23:34:11.674022 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) Running State Change CS_HANGUP
2011-04-01 23:34:11.674022 [DEBUG] switch_ivr_originate.c:3506 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
2011-04-01 23:34:11.674022 [ERR] switch_ivr_originate.c:2640 Cannot create outgoing channel of type [user] cause: [RECOVERY_ON_TIMER_EXPIRE]
2011-04-01 23:34:11.674022 [DEBUG] switch_ivr_originate.c:3506 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
2011-04-01 23:34:11.675057 [DEBUG] switch_core_state_machine.c:560 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State HANGUP
2011-04-01 23:34:11.675057 [DEBUG] mod_sofia.c:451 sofia/internal/sip:99999999999 at 11.22.33.44:52767 Overriding SIP cause 504 with 408 from the other leg
2011-04-01 23:34:11.675057 [DEBUG] mod_sofia.c:457 Channel sofia/internal/sip:99999999999 at 11.22.33.44:52767 hanging up, cause: RECOVERY_ON_TIMER_EXPIRE
2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:99999999999 at 11.22.33.44:52767 Standard HANGUP, cause: RECOVERY_ON_TIMER_EXPIRE
2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:560 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State HANGUP going to sleep
2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State Change CS_HANGUP -> CS_REPORTING
2011-04-01 23:34:11.695371 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:99999999999 at 11.22.33.44:52767 [BREAK]
2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) Running State Change CS_REPORTING
2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State REPORTING
2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:99999999999 at 11.22.33.44:52767 Standard REPORTING, cause: RECOVERY_ON_TIMER_EXPIRE
2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State REPORTING going to sleep
2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:345 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State Change CS_REPORTING -> CS_DESTROY
2011-04-01 23:34:11.695371 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:99999999999 at 11.22.33.44:52767 [BREAK]
2011-04-01 23:34:11.695371 [DEBUG] switch_core_session.c:1288 Session 5 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) Locked, Waiting on external entities
2011-04-01 23:34:11.695371 [NOTICE] switch_core_session.c:1306 Session 5 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) Ended
2011-04-01 23:34:11.695371 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/sip:99999999999 at 11.22.33.44:52767 [CS_DESTROY]
2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:449 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) Callstate Change HANGUP -> DOWN
2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) Running State Change CS_DESTROY
2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State DESTROY
2011-04-01 23:34:11.696436 [DEBUG] mod_sofia.c:362 sofia/internal/sip:99999999999 at 11.22.33.44:52767 SOFIA DESTROY
2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:99999999999 at 11.22.33.44:52767 Standard DESTROY
2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State DESTROY going to sleep


a

  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Friday, April 01, 2011 5:30 PM
  Subject: Re: [Freeswitch-users] originate from cli


  It can also be that the other side sent FS a reply saying that *it* had timed out. siptrace will also show if that's the case.



  On 1 April 2011 22:29, Steven Ayre <steveayre at gmail.com> wrote:

    It means FS sent a message and didn't get a reply (timed out).

    As anthm says, look at the siptrace - that'll show you what's being sent / received.

    -Steve



    On 1 April 2011 21:17, Madovsky <infos at madovsky.org> wrote:

      I make some test with originate from cli.

      /usr/local/freeswitch/bin/fs_cli -x "originate user/9999 7777 XML public"
      -ERR RECOVERY_ON_TIMER_EXPIRE

      9999 and 7777 are ready to receive calls and I don't have any NAT.

      is anyone can explain what it means ?

      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
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From dujinfang at gmail.com  Sat Apr  2 11:44:32 2011
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 2 Apr 2011 15:44:32 +0800
Subject: [Freeswitch-users] video problem in conference with H264
In-Reply-To: <AD0F952609504D9D8E8AC968ADABE0C8@gmail.com>
References: <AD0F952609504D9D8E8AC968ADABE0C8@gmail.com>
Message-ID: <942671FE18024326A312ED29ACE4F5DD@gmail.com>

Answer myself.


I traced code and found that in line 1007 of mod_conference the frame data never matches 0x11

} else if (vid_frame->codec->implementation->ianacode == 99) { /* h.264 */
iframe = (*((int16_t *) vid_frame->data) >> 5 == 0x11);

I hardcoded to iframe = 1 and then it works.

As I said I don't have problem with Bria, but with the xtp8886 device I got the following data sequence. I'm not familiar with video encoding, so is my device broken or we need other methods to detect an i-frame or is it safe to just hard coded into 1?

Thanks.



*(int16_t *) vid_frame->data,  *((int16_t *) vid_frame->data) >> 5

ffffb465, fffffda3
165, b
165, b
65, 3
ffffd061, fffffe83
61, 3
ffffd061, fffffe83
161, b
4267, 213
4868, 243
ffffb465, fffffda3
165, b
165, b
65, 3
ffffd061, fffffe83
61, 3
ffffd061, fffffe83
161, b
61, 3
ffffd061, fffffe83
461, 23
161, b
61, 3
ffffd061, fffffe83
161, b
ffffd061, fffffe83
361, 1b
61, 3
ffffd061, fffffe83
161, b
ffffd061, fffffe83
361, 1b
161, b
ffffd061, fffffe83
61, 3
ffffd061, fffffe83
361, 1b
161, b
ffffd061, fffffe83
161, b
ffffd061, fffffe83
361, 1b
161, b
61, 3
ffffd061, fffffe83
161, b
ffffd061, fffffe83
361, 1b
161, b
61, 3
ffffd061, fffffe83
161, b
4267, 213
4868, 243
ffffb465, fffffda3
165, b
165, b
65, 3
ffffd061, fffffe83
ffffd061, fffffe83
61, 3
ffffd061, fffffe83
161, b
ffffd061, fffffe83
161, b
61, 3
ffffd061, fffffe83
161, b
ffffd061, fffffe83
161, b
61, 3
4267, 213
4868, 243
ffffb465, fffffda3
365, 1b
165, b
65, 3
ffffd061, fffffe83
ffffd061, fffffe83
61, 3
ffffd061, fffffe83
61, 3
ffffd061, fffffe83
161, b


On Wednesday, March 30, 2011 at 7:31 PM, Seven Du wrote: 
> I tested with default 3000 conference and it just OK. But I have problem on H264.
> 
> I tested with one Bria 3.1 on Mac and two XTP8886 hardware phones.
> 
> http://www.gvscusa.com/xtp8886.html
> 
> Bria 1003
> XTP 1011/1012
> 
> call from 1003 to 1011 and from 1011 to 1003 both ok with videos.
> 
> http://pastebin.freeswitch.org/15910
> http://pastebin.freeswitch.org/15911
> 
> When 3 phones calling into 3000(conference), Everyone call see Bria(1003), but no one can say 1011 and 1012. Even when I muted 1003. 
> 
> http://pastebin.freeswitch.org/15913
> 
> As I said there's no problem with similar test with h263.
> 
> Can anyone help take a look, thanks.
> 
> -- 
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj: http://www.freeswitch.org.cn
> 
> Sent with Sparrow
> 
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From infos at madovsky.org  Sat Apr  2 12:08:25 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 2 Apr 2011 04:08:25 -0400
Subject: [Freeswitch-users] incoming fax calls
References: <2AEA11608B0642348D4C867C5058F0AC@e1705>
	<AANLkTikFZHE_QD7na=uABeaGaUeX85cibpGjfUNcwCsr@mail.gmail.com>
Message-ID: <6FB02A6745904515B882E9A35BE14584@e1705>

I tried your solution

        <extension name="myphone">
                <condition field="${sip_to_uri}" expression="^(9999)@$${domain}$">
                        <action inline="true" application="export" data="dialed_extension=$1"/>
                        <action application="set" data="session_in_hangup_hook=true"/>
                        <action application="set" data="hangup_after_bridge=true"/>
                        <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,CALL_REJECTED,NORMAL_CLEARING,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
                        <action application="set" data="group_confirm_cancel_timeout=1"/>
                        <action application="ring_ready"/>
                        <action application="tone_detect" data="fax 1100 r +5000 transfer 'receivefax XML features' 1"/>
                        <action application="sleep" data="5000"/>
                        <action application="export" data="originate_timeout=25"/>
                        <action application="bridge" data="{sip_invite_domain=${sip_from_host},nibble_account=,nibble_rate=,origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number,}}user/${dialed_extension}"/>
                </condition>
         </extension>

but the tone_detect doesn't tranfer as long as the bridge answer

Thanks


  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Friday, April 01, 2011 12:44 AM
  Subject: Re: [Freeswitch-users] incoming fax calls


  Is this an incoming call? If so then why are you doing "execute_on_media"? Wouldn't you want to pre_answer the call, do the tone_detect and sleep for 5000ms or so, and then proceed on to the bridge?


  -MC


  On Thu, Mar 31, 2011 at 10:10 AM, Madovsky <infos at madovsky.org> wrote:

    I'm trying to find a way to dectect a fax or call from the same extension

            <extension name="myphone">
                    <condition field="${sip_to_uri}" expression="^(9999)@$${domain}$">
                            <action inline="true" application="export" data="dialed_extension=$1"/>
                            <action application="set" data="session_in_hangup_hook=true"/>
                            <action application="set" data="hangup_after_bridge=true"/>
                            <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,CALL_REJECTED,NORMAL_CLEARING,USER_NOT_REGISTERED,NO_ANSWER,NO_USER_RESPONSE,USER_BUSY"/>
                            <action application="set" data="group_confirm_cancel_timeout=1"/>
                            <action application="ring_ready"/>
                            <action application="set" data="execute_on_media=tone_detect fax 1100 r +5000 transfer 'receivefax XML features' 1"/>
                            <action application="export" data="originate_timeout=25"/>
                            <action application="bridge" data="{sip_invite_domain=${sip_from_host},nibble_account=,nibble_rate=,origination_caller_id_name=${caller_id_name},origination_caller_id_number=${caller_id_number,}}user/${dialed_extension}"/>
                    </condition>
             </extension>

    is there a way to detect a fax before answer (2 rings for example) and avoid 
    phone rings until no answer ?

    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






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From Info at KennedySoftware.ie  Sat Apr  2 14:10:49 2011
From: Info at KennedySoftware.ie (Michael Kennedy)
Date: Sat, 02 Apr 2011 11:10:49 +0100
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <1301712035.18009.1048.camel@macmini>
References: <4D94C282.1090903@KennedySoftware.ie>	<4D951D45.5010005@utoronto.ca>
	<4D95E839.2070403@KennedySoftware.ie>
	<1301712035.18009.1048.camel@macmini>
Message-ID: <4D96F629.2020005@KennedySoftware.ie>

Thank you very much, Campbell.

Knowing what to AVOID is frequently even MORE valuable that what to try/use!

   - Mike


On 02/04/2011 03:40, Campbell Steven wrote:
> The Snom 870 will do it with a USB Wifi dongle, but in my experience
> don't go there, they are a diabolical handset from a usability standpoint.
>
> Campbell


From lists at telefaks.de  Sat Apr  2 14:30:58 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Sat, 02 Apr 2011 12:30:58 +0200
Subject: [Freeswitch-users] Dingaling and sasl authentication failed
In-Reply-To: <4D9127CD.7060300@telefaks.de>
References: <4D9127CD.7060300@telefaks.de>
Message-ID: <4D96FAE2.2050608@telefaks.de>

Nobody has an idea?

Best regards
Peter

Peter Steinbach schrieb:
> Hello,
>
> I installed mod_dingaling and having problems with the registration
>
> freeswitch at internal> dingaling status
> --DingaLing status--
> login   |       connected
> my.account at googlemail.com/talk   |       UNCONNECTED
>
> 2011-03-29 02:02:48.756832 [DEBUG] libdingaling.c:1289 sasl
> authentication failed
> 2011-03-29 02:02:48.756832 [DEBUG] libdingaling.c:1607 io error 2 7
> retry in 47 second(s)
>
> Searching the mailing list led me to look at libgnutls26.
> But libgnutls26 and libgnutls-dev are both installed, and both the are
> newest version, and both were installed long before I configured and
> compiled mod_dingaling. And TLS for SIP is working sucessfully since a
> while.
>
> Anybody has a hint where to look further?
>
>   


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From Nabble at slickdeals.endjunk.com  Sat Apr  2 15:29:17 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 2 Apr 2011 04:29:17 -0700 (PDT)
Subject: [Freeswitch-users] Dingaling and sasl authentication failed
In-Reply-To: <4D9127CD.7060300@telefaks.de>
References: <4D9127CD.7060300@telefaks.de>
Message-ID: <1301743757340-6233670.post@n2.nabble.com>

I don't know if this will help or not. But, so far the only dingaling error
messages found in /var/log/freeswitch/freeswitch.log file on my FS (running
on FreeSWITCH Version 1.0.head (git-9795dd2 2011-03-26 11-07-34 -0500)) is
shown below:
2011-03-31 13:22:30.718490 [DEBUG] libdingaling.c:1610 io error 2 7 retry in
3 second(s)
2011-03-31 13:22:34.171096 [DEBUG] libdingaling.c:1297 XMPP server connected
2011-03-31 13:22:34.307809 [DEBUG] libdingaling.c:1309 XMPP authenticated


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dingaling-and-sasl-authentication-failed-tp6217329p6233670.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Nabble at slickdeals.endjunk.com  Sat Apr  2 17:23:07 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 2 Apr 2011 06:23:07 -0700 (PDT)
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <4D94C282.1090903@KennedySoftware.ie>
References: <4D94C282.1090903@KennedySoftware.ie>
Message-ID: <1301750587582-6233798.post@n2.nabble.com>


Michael Kennedy wrote:
> I'm hoping to roll out FS where some areas in a building are wired, and 
> other areas are on WiFi, and to deploy some SIP phones in both areas.If
> you can still find an inexpensive 
> http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
> Seagate FreeAgent DockStar  (used to be on sale for as low as
> $13.99/each), you certainly can use it to host your FS. It is an ARM
> platform clocked @1.2GHz with 128/256MB RAM/NAND, 4 USB2 ports, and a
> single Gigabit RJ-45 port. Unless you already have a NAT/Firewall WiFi
> router, all you need is an additional USB WiFi dongle to make it
> WiFi-able.


I expected that many phone suppliers would have handsets with EITHER 
> RJ45 or WiFi connectivity to the LAN, or even both! I've found only a 
> single device, a Cisco SPA525G2! Furthermore, searching the FS site, and 
> various VoIP sites, and running general searches, I've found no other 
> SIP WiFi phones that look like standard desktop handsets.
These days, more and more people are using smartphone.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Recommended-SIP-IP-Wifi-Handsets-tp6229405p6233798.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Info at KennedySoftware.ie  Sat Apr  2 18:01:07 2011
From: Info at KennedySoftware.ie (Michael Kennedy)
Date: Sat, 02 Apr 2011 15:01:07 +0100
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <1301750587582-6233798.post@n2.nabble.com>
References: <4D94C282.1090903@KennedySoftware.ie>
	<1301750587582-6233798.post@n2.nabble.com>
Message-ID: <4D972C23.60305@KennedySoftware.ie>

> I expected that many phone suppliers would have handsets with EITHER
>> RJ45 or WiFi connectivity to the LAN, or even both! I've found only a
>> single device, a Cisco SPA525G2! Furthermore, searching the FS site, and
>> various VoIP sites, and running general searches, I've found no other
>> SIP WiFi phones that look like standard desktop handsets.

> These days, more and more people are using smartphone.

Yes, I appreciate that.

However, *I*'m very old!, and probably much too conservative, as are 
some clients - though we're still using lots of Linux, IT, and, maybe, FS!

Some employees already use smartphones, but I believe the client is 
REDUCING the support and usage of these, because of the very high costs, 
and no proportional benefits, and opportunities to just "waste" lots of 
employee time on them, etc...

   - Mike.


From richocet2 at hotmail.com  Sat Apr  2 15:49:31 2011
From: richocet2 at hotmail.com (Dave Bracken)
Date: Sat, 2 Apr 2011 11:49:31 +0000
Subject: [Freeswitch-users] freeswitch-users@lists.freeswitch.org
Message-ID: <BLU0-SMTP32BB3CBC68FEC4D0E7B5FAF0A10@phx.gbl>

freeswitch-users at lists.freeswitch.org life has thrown plenty of obstacles my way people always look for an easy way out this turned my luck around http://j.mp/frSIba now I vacation four times a year youll get the hang of it


From curriegrad2004 at gmail.com  Sat Apr  2 20:07:42 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 2 Apr 2011 09:07:42 -0700
Subject: [Freeswitch-users] freeswitch-users@lists.freeswitch.org
In-Reply-To: <BLU0-SMTP32BB3CBC68FEC4D0E7B5FAF0A10@phx.gbl>
References: <BLU0-SMTP32BB3CBC68FEC4D0E7B5FAF0A10@phx.gbl>
Message-ID: <AANLkTikQn6b6M6WTTq=i=hVBHJF=agcE98zYAQK7Y=LH@mail.gmail.com>

Can somebody moderate this person already?

On Sat, Apr 2, 2011 at 4:49 AM, Dave Bracken <richocet2 at hotmail.com> wrote:
> freeswitch-users at lists.freeswitch.org life has thrown plenty of obstacles my way people always look for an easy way out this turned my luck around http://j.mp/frSIba now I vacation four times a year youll get the hang of 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 alexstdb at gmail.com  Sat Apr  2 05:29:36 2011
From: alexstdb at gmail.com (Alex)
Date: Fri, 1 Apr 2011 22:29:36 -0300
Subject: [Freeswitch-users] Cannot receive fax with T.38
Message-ID: <AANLkTimm5sCBj27A14ZHC0g+xL+_46LCfBQyi-Ys32Hf@mail.gmail.com>

Hi everybody,

I am stuck for several days trying to receive a fax using T.38

Based on the instructions in the wiki, I use the following dialplan:

<action application="answer" />
<action application="playback" data="silence_stream://2000"/>
<action application=set data="fax_enable_t38=true"/>
<action application=set data="fax_enable_t38_request=true"/>
<action application="rxfax" data="/tmp/rxfax-${uuid}.tif"/>
<action application="hangup"/>

When I do that, when rxfax is executed, the following appears on the
console:

2011-04-02 01:00:04.890173 [ERR] switch_core_session.c:1918 Invalid
Application

If I take out the t38 lines, rxfax behaves as expected, receiving the fax
using ulaw.

However, I would like to use t.38 instead, which is supported by my gateway.

My scenario is:

PSTN -> CISCO AS5400 from my provider -> Internet -> My Freeswitch.

I have also tried connecting a fax to a Grandstream HT-502, with the same
results. I get a INVITE (which don't mention anywhere T.38 in the sdp) and
Freeswitch never issues a reINVITE asking to switch to t.38

My version of FS is:
FreeSWITCH Version 1.0.head (git-24a9729 2011-03-11 13-00-55 -0600)

Anybody can give me a clue on what could I be doing wrong????

Thanks in advance!

Alex
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From boris at tagnet.ru  Sat Apr  2 20:32:00 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 02 Apr 2011 22:32:00 +0600
Subject: [Freeswitch-users] Cannot receive fax with T.38
In-Reply-To: <AANLkTimm5sCBj27A14ZHC0g+xL+_46LCfBQyi-Ys32Hf@mail.gmail.com>
References: <AANLkTimm5sCBj27A14ZHC0g+xL+_46LCfBQyi-Ys32Hf@mail.gmail.com>
Message-ID: <4D974F80.2090002@tagnet.ru>

Hello!

     Have You compiled and loaded mod_spandsp?

> Hi everybody,
>
> I am stuck for several days trying to receive a fax using T.38
>
> Based on the instructions in the wiki, I use the following dialplan:
>
> <action application="answer" />
> <action application="playback" data="silence_stream://2000"/>
> <action application=set data="fax_enable_t38=true"/>
> <action application=set data="fax_enable_t38_request=true"/>
> <action application="rxfax" data="/tmp/rxfax-${uuid}.tif"/>
> <action application="hangup"/>
>
> When I do that, when rxfax is executed, the following appears on the 
> console:
>
> 2011-04-02 01:00:04.890173 [ERR] switch_core_session.c:1918 Invalid 
> Application
>
> If I take out the t38 lines, rxfax behaves as expected, receiving the 
> fax using ulaw.
>
> However, I would like to use t.38 instead, which is supported by my 
> gateway.
>
> My scenario is:
>
> PSTN -> CISCO AS5400 from my provider -> Internet -> My Freeswitch.
>
> I have also tried connecting a fax to a Grandstream HT-502, with the 
> same results. I get a INVITE (which don't mention anywhere T.38 in the 
> sdp) and Freeswitch never issues a reINVITE asking to switch to t.38
>
> My version of FS is:
> FreeSWITCH Version 1.0.head (git-24a9729 2011-03-11 13-00-55 -0600)
>
> Anybody can give me a clue on what could I be doing wrong????
>
> Thanks in advance!
>
> Alex
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991


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From alexstdb at gmail.com  Sat Apr  2 20:52:46 2011
From: alexstdb at gmail.com (Alex)
Date: Sat, 2 Apr 2011 13:52:46 -0300
Subject: [Freeswitch-users] Cannot receive fax with T.38
In-Reply-To: <4D974F80.2090002@tagnet.ru>
References: <AANLkTimm5sCBj27A14ZHC0g+xL+_46LCfBQyi-Ys32Hf@mail.gmail.com>
	<4D974F80.2090002@tagnet.ru>
Message-ID: <BANLkTi=LwjhJ8RH8ddT=3w+QCKYKbb6w8A@mail.gmail.com>

Yes, mod_spandsp is loaded.

In fact rxfax works fine when I take out the two fax_enable_t38_xxxx lines.


On Sat, Apr 2, 2011 at 1:32 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

>  Hello!
>
>     Have You compiled and loaded mod_spandsp?
>
> Hi everybody,
>
> I am stuck for several days trying to receive a fax using T.38
>
> Based on the instructions in the wiki, I use the following dialplan:
>
> <action application="answer" />
> <action application="playback" data="silence_stream://2000"/>
> <action application=set data="fax_enable_t38=true"/>
> <action application=set data="fax_enable_t38_request=true"/>
> <action application="rxfax" data="/tmp/rxfax-${uuid}.tif"/>
> <action application="hangup"/>
>
> When I do that, when rxfax is executed, the following appears on the
> console:
>
> 2011-04-02 01:00:04.890173 [ERR] switch_core_session.c:1918 Invalid
> Application
>
> If I take out the t38 lines, rxfax behaves as expected, receiving the fax
> using ulaw.
>
> However, I would like to use t.38 instead, which is supported by my
> gateway.
>
> My scenario is:
>
> PSTN -> CISCO AS5400 from my provider -> Internet -> My Freeswitch.
>
> I have also tried connecting a fax to a Grandstream HT-502, with the same
> results. I get a INVITE (which don't mention anywhere T.38 in the sdp) and
> Freeswitch never issues a reINVITE asking to switch to t.38
>
> My version of FS is:
> FreeSWITCH Version 1.0.head (git-24a9729 2011-03-11 13-00-55 -0600)
>
> Anybody can give me a clue on what could I be doing wrong????
>
> Thanks in advance!
>
> Alex
>
>
> _______________________________________________
> 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
>
>
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   (3435) 494991
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From boris at tagnet.ru  Sat Apr  2 21:48:58 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 02 Apr 2011 23:48:58 +0600
Subject: [Freeswitch-users] Cannot receive fax with T.38
In-Reply-To: <BANLkTi=LwjhJ8RH8ddT=3w+QCKYKbb6w8A@mail.gmail.com>
References: <AANLkTimm5sCBj27A14ZHC0g+xL+_46LCfBQyi-Ys32Hf@mail.gmail.com>	<4D974F80.2090002@tagnet.ru>
	<BANLkTi=LwjhJ8RH8ddT=3w+QCKYKbb6w8A@mail.gmail.com>
Message-ID: <4D97618A.5050306@tagnet.ru>

Hello!

     Try to add:
<action application="export" data="execute_on_answer=t38_gateway peer"/>

> Yes, mod_spandsp is loaded.
>
> In fact rxfax works fine when I take out the two fax_enable_t38_xxxx 
> lines.
>
>
> On Sat, Apr 2, 2011 at 1:32 PM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         Have You compiled and loaded mod_spandsp?
>
>>     Hi everybody,
>>
>>     I am stuck for several days trying to receive a fax using T.38
>>
>>     Based on the instructions in the wiki, I use the following dialplan:
>>
>>     <action application="answer" />
>>     <action application="playback" data="silence_stream://2000"/>
>>     <action application=set data="fax_enable_t38=true"/>
>>     <action application=set data="fax_enable_t38_request=true"/>
>>     <action application="rxfax" data="/tmp/rxfax-${uuid}.tif"/>
>>     <action application="hangup"/>
>>
>>     When I do that, when rxfax is executed, the following appears on
>>     the console:
>>
>>     2011-04-02 01:00:04.890173 [ERR] switch_core_session.c:1918
>>     Invalid Application
>>
>>     If I take out the t38 lines, rxfax behaves as expected, receiving
>>     the fax using ulaw.
>>
>>     However, I would like to use t.38 instead, which is supported by
>>     my gateway.
>>
>>     My scenario is:
>>
>>     PSTN -> CISCO AS5400 from my provider -> Internet -> My Freeswitch.
>>
>>     I have also tried connecting a fax to a Grandstream HT-502, with
>>     the same results. I get a INVITE (which don't mention anywhere
>>     T.38 in the sdp) and Freeswitch never issues a reINVITE asking to
>>     switch to t.38
>>
>>     My version of FS is:
>>     FreeSWITCH Version 1.0.head (git-24a9729 2011-03-11 13-00-55 -0600)
>>
>>     Anybody can give me a clue on what could I be doing wrong????
>>
>>     Thanks in advance!
>>
>>     Alex
>>
>>
>>     _______________________________________________
>>     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
>
>
>     -- 
>     ? ?????????,
>        ????? ?????????
>        ??? "??????"
>        (3435) 494991
>
>
>
>     _______________________________________________
>     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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991


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From sos at sokhapkin.dyndns.org  Sat Apr  2 21:53:02 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sat, 2 Apr 2011 13:53:02 -0400
Subject: [Freeswitch-users] Cannot receive fax with T.38
In-Reply-To: <BANLkTi=LwjhJ8RH8ddT=3w+QCKYKbb6w8A@mail.gmail.com>
References: <AANLkTimm5sCBj27A14ZHC0g+xL+_46LCfBQyi-Ys32Hf@mail.gmail.com>
	<4D974F80.2090002@tagnet.ru>
	<BANLkTi=LwjhJ8RH8ddT=3w+QCKYKbb6w8A@mail.gmail.com>
Message-ID: <201104021353.02078.sos@sokhapkin.dyndns.org>

application="set". You missed quotes.

On Saturday 02 April 2011, Alex wrote:
> Yes, mod_spandsp is loaded.
> 
> In fact rxfax works fine when I take out the two fax_enable_t38_xxxx lines.
> 
> On Sat, Apr 2, 2011 at 1:32 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
> >  Hello!
> >  
> >     Have You compiled and loaded mod_spandsp?
> > 
> > Hi everybody,
> > 
> > I am stuck for several days trying to receive a fax using T.38
> > 
> > Based on the instructions in the wiki, I use the following dialplan:
> > 
> > <action application="answer" />
> > <action application="playback" data="silence_stream://2000"/>
> > <action application=set data="fax_enable_t38=true"/>
> > <action application=set data="fax_enable_t38_request=true"/>
> > <action application="rxfax" data="/tmp/rxfax-${uuid}.tif"/>
> > <action application="hangup"/>
> > 
> > When I do that, when rxfax is executed, the following appears on the
> > console:
> > 
> > 2011-04-02 01:00:04.890173 [ERR] switch_core_session.c:1918 Invalid
> > Application
> > 
> > If I take out the t38 lines, rxfax behaves as expected, receiving the fax
> > using ulaw.
> > 
> > However, I would like to use t.38 instead, which is supported by my
> > gateway.
> > 
> > My scenario is:
> > 
> > PSTN -> CISCO AS5400 from my provider -> Internet -> My Freeswitch.
> > 
> > I have also tried connecting a fax to a Grandstream HT-502, with the same
> > results. I get a INVITE (which don't mention anywhere T.38 in the sdp)
> > and Freeswitch never issues a reINVITE asking to switch to t.38
> > 
> > My version of FS is:
> > FreeSWITCH Version 1.0.head (git-24a9729 2011-03-11 13-00-55 -0600)
> > 
> > Anybody can give me a clue on what could I be doing wrong????
> > 
> > Thanks in advance!
> > 
> > Alex
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing
> > listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mai
> > lman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> > 
> > 
> > --
> > ? ?????????,
> > 
> >   ????? ?????????
> >   ??? "??????"
> >   (3435) 494991
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.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 Apr  2 23:31:10 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 2 Apr 2011 14:31:10 -0500
Subject: [Freeswitch-users] video problem in conference with H264
In-Reply-To: <942671FE18024326A312ED29ACE4F5DD@gmail.com>
References: <AD0F952609504D9D8E8AC968ADABE0C8@gmail.com>
	<942671FE18024326A312ED29ACE4F5DD@gmail.com>
Message-ID: <BANLkTinEbJhmLWFZ=HfjbF5+gvmJgFE9YA@mail.gmail.com>

If you can find a patch to properly tell full frames on various codec it
would be nice. That code is unfinished.
On Apr 2, 2011 2:46 AM, "Seven Du" <dujinfang at gmail.com> wrote:
> Answer myself.
>
>
> I traced code and found that in line 1007 of mod_conference the frame data
never matches 0x11
>
> } else if (vid_frame->codec->implementation->ianacode == 99) { /* h.264 */
> iframe = (*((int16_t *) vid_frame->data) >> 5 == 0x11);
>
> I hardcoded to iframe = 1 and then it works.
>
> As I said I don't have problem with Bria, but with the xtp8886 device I
got the following data sequence. I'm not familiar with video encoding, so is
my device broken or we need other methods to detect an i-frame or is it safe
to just hard coded into 1?
>
> Thanks.
>
>
>
> *(int16_t *) vid_frame->data, *((int16_t *) vid_frame->data) >> 5
>
> ffffb465, fffffda3
> 165, b
> 165, b
> 65, 3
> ffffd061, fffffe83
> 61, 3
> ffffd061, fffffe83
> 161, b
> 4267, 213
> 4868, 243
> ffffb465, fffffda3
> 165, b
> 165, b
> 65, 3
> ffffd061, fffffe83
> 61, 3
> ffffd061, fffffe83
> 161, b
> 61, 3
> ffffd061, fffffe83
> 461, 23
> 161, b
> 61, 3
> ffffd061, fffffe83
> 161, b
> ffffd061, fffffe83
> 361, 1b
> 61, 3
> ffffd061, fffffe83
> 161, b
> ffffd061, fffffe83
> 361, 1b
> 161, b
> ffffd061, fffffe83
> 61, 3
> ffffd061, fffffe83
> 361, 1b
> 161, b
> ffffd061, fffffe83
> 161, b
> ffffd061, fffffe83
> 361, 1b
> 161, b
> 61, 3
> ffffd061, fffffe83
> 161, b
> ffffd061, fffffe83
> 361, 1b
> 161, b
> 61, 3
> ffffd061, fffffe83
> 161, b
> 4267, 213
> 4868, 243
> ffffb465, fffffda3
> 165, b
> 165, b
> 65, 3
> ffffd061, fffffe83
> ffffd061, fffffe83
> 61, 3
> ffffd061, fffffe83
> 161, b
> ffffd061, fffffe83
> 161, b
> 61, 3
> ffffd061, fffffe83
> 161, b
> ffffd061, fffffe83
> 161, b
> 61, 3
> 4267, 213
> 4868, 243
> ffffb465, fffffda3
> 365, 1b
> 165, b
> 65, 3
> ffffd061, fffffe83
> ffffd061, fffffe83
> 61, 3
> ffffd061, fffffe83
> 61, 3
> ffffd061, fffffe83
> 161, b
>
>
> On Wednesday, March 30, 2011 at 7:31 PM, Seven Du wrote:
>> I tested with default 3000 conference and it just OK. But I have problem
on H264.
>>
>> I tested with one Bria 3.1 on Mac and two XTP8886 hardware phones.
>>
>> http://www.gvscusa.com/xtp8886.html
>>
>> Bria 1003
>> XTP 1011/1012
>>
>> call from 1003 to 1011 and from 1011 to 1003 both ok with videos.
>>
>> http://pastebin.freeswitch.org/15910
>> http://pastebin.freeswitch.org/15911
>>
>> When 3 phones calling into 3000(conference), Everyone call see
Bria(1003), but no one can say 1011 and 1012. Even when I muted 1003.
>>
>> http://pastebin.freeswitch.org/15913
>>
>> As I said there's no problem with similar test with h263.
>>
>> Can anyone help take a look, thanks.
>>
>> --
>> About: http://about.me/dujinfang
>> Blog: http://www.dujinfang.com
>> Proj: http://www.freeswitch.org.cn
>>
>> Sent with Sparrow
>>
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From anthony.minessale at gmail.com  Sat Apr  2 23:34:04 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 2 Apr 2011 14:34:04 -0500
Subject: [Freeswitch-users] originate from cli
In-Reply-To: <C66F263892AE4887B47D8D6B0B3FA4D0@e1705>
References: <F38382DFFBE041AF8562F02269FE45EC@e1705>
	<AANLkTiks_xJX8vyX-QWZ+ar+Bwhn0Mm6TPvcp4=_39DC@mail.gmail.com>
	<AANLkTik3or8JqMUmMwaYxgSumjD+Hm+HBtCh_WLusk6=@mail.gmail.com>
	<C66F263892AE4887B47D8D6B0B3FA4D0@e1705>
Message-ID: <BANLkTi=37Pv9Dyb4ughCA1_F4NapoZNt7A@mail.gmail.com>

You never get the reply to the invite trace from the thing you are calling
and see if they are getting any packets.  Maybe you have iptables or selinux
on on one host or the other.
On Apr 1, 2011 10:37 PM, "Madovsky" <infos at madovsky.org> wrote:
> sorry the sip trace :
> after this command:
>
> /usr/local/freeswitch/bin/fs_cli -x "originate user/9999999999 1111111111"
>
> ------------------------------------------------------------------------
> 2011-04-01 23:33:39.673581 [DEBUG] sofia.c:4754 Channel sofia/internal/
sip:99999999999 at 11.22.33.44:52767 entering state [calling][0]
> send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:40.673952:
> ------------------------------------------------------------------------
> INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
> Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
> Max-Forwards: 70
> From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
> To: <sip:99999999999 at 11.22.33.44:52767>
> Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
> CSeq: 10521545 INVITE
> Contact: <sip:mod_sofia at 11.22.33.44:5080>
> User-Agent: CiscoSystems-SIP-GW-UA
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 492
> X-FS-Support: update_display
> Remote-Party-ID: <sip:0000000000 at 11.22.33.44
>;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
> s=FreeSWITCH
> c=IN IP4 11.22.33.44
> t=0 0
> m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
> a=rtpmap:98 SPEEX/16000
> a=rtpmap:99 G726-16/8000
> a=rtpmap:100 G726-24/8000
> a=rtpmap:102 G726-32/8000
> a=rtpmap:103 G726-40/8000
> a=rtpmap:104 G7221/16000
> a=fmtp:104 bitrate=32000
> a=rtpmap:105 iLBC/8000
> a=fmtp:105 mode=20
> a=rtpmap:106 L16/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> ------------------------------------------------------------------------
> send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:42.673971:
> ------------------------------------------------------------------------
> INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
> Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
> Max-Forwards: 70
> From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
> To: <sip:99999999999 at 11.22.33.44:52767>
> Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
> CSeq: 10521545 INVITE
> Contact: <sip:mod_sofia at 11.22.33.44:5080>
> User-Agent: CiscoSystems-SIP-GW-UA
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 492
> X-FS-Support: update_display
> Remote-Party-ID: <sip:0000000000 at 11.22.33.44
>;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
> s=FreeSWITCH
> c=IN IP4 11.22.33.44
> t=0 0
> m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
> a=rtpmap:98 SPEEX/16000
> a=rtpmap:99 G726-16/8000
> a=rtpmap:100 G726-24/8000
> a=rtpmap:102 G726-32/8000
> a=rtpmap:103 G726-40/8000
> a=rtpmap:104 G7221/16000
> a=fmtp:104 bitrate=32000
> a=rtpmap:105 iLBC/8000
> a=fmtp:105 mode=20
> a=rtpmap:106 L16/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> ------------------------------------------------------------------------
> send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:46.673944:
> ------------------------------------------------------------------------
> INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
> Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
> Max-Forwards: 70
> From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
> To: <sip:99999999999 at 11.22.33.44:52767>
> Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
> CSeq: 10521545 INVITE
> Contact: <sip:mod_sofia at 11.22.33.44:5080>
> User-Agent: CiscoSystems-SIP-GW-UA
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 492
> X-FS-Support: update_display
> Remote-Party-ID: <sip:0000000000 at 11.22.33.44
>;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
> s=FreeSWITCH
> c=IN IP4 11.22.33.44
> t=0 0
> m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
> a=rtpmap:98 SPEEX/16000
> a=rtpmap:99 G726-16/8000
> a=rtpmap:100 G726-24/8000
> a=rtpmap:102 G726-32/8000
> a=rtpmap:103 G726-40/8000
> a=rtpmap:104 G7221/16000
> a=fmtp:104 bitrate=32000
> a=rtpmap:105 iLBC/8000
> a=fmtp:105 mode=20
> a=rtpmap:106 L16/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> ------------------------------------------------------------------------
> send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:54.673950:
> ------------------------------------------------------------------------
> INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
> Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
> Max-Forwards: 70
> From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
> To: <sip:99999999999 at 11.22.33.44:52767>
> Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
> CSeq: 10521545 INVITE
> Contact: <sip:mod_sofia at 11.22.33.44:5080>
> User-Agent: CiscoSystems-SIP-GW-UA
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 492
> X-FS-Support: update_display
> Remote-Party-ID: <sip:0000000000 at 11.22.33.44
>;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
> s=FreeSWITCH
> c=IN IP4 11.22.33.44
> t=0 0
> m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
> a=rtpmap:98 SPEEX/16000
> a=rtpmap:99 G726-16/8000
> a=rtpmap:100 G726-24/8000
> a=rtpmap:102 G726-32/8000
> a=rtpmap:103 G726-40/8000
> a=rtpmap:104 G7221/16000
> a=fmtp:104 bitrate=32000
> a=rtpmap:105 iLBC/8000
> a=fmtp:105 mode=20
> a=rtpmap:106 L16/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> ------------------------------------------------------------------------
> send 1385 bytes to udp/[11.22.33.44]:52767 at 03:34:10.674077:
> ------------------------------------------------------------------------
> INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
> Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
> Max-Forwards: 70
> From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
> To: <sip:99999999999 at 11.22.33.44:52767>
> Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
> CSeq: 10521545 INVITE
> Contact: <sip:mod_sofia at 11.22.33.44:5080>
> User-Agent: CiscoSystems-SIP-GW-UA
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER,
REFER, NOTIFY, PUBLISH, SUBSCRIBE
> Supported: timer, precondition, path, replaces
> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
> Content-Type: application/sdp
> Content-Disposition: session
> Content-Length: 492
> X-FS-Support: update_display
> Remote-Party-ID: <sip:0000000000 at 11.22.33.44
>;party=calling;screen=yes;privacy=off
>
> v=0
> o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
> s=FreeSWITCH
> c=IN IP4 11.22.33.44
> t=0 0
> m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
> a=rtpmap:98 SPEEX/16000
> a=rtpmap:99 G726-16/8000
> a=rtpmap:100 G726-24/8000
> a=rtpmap:102 G726-32/8000
> a=rtpmap:103 G726-40/8000
> a=rtpmap:104 G7221/16000
> a=fmtp:104 bitrate=32000
> a=rtpmap:105 iLBC/8000
> a=fmtp:105 mode=20
> a=rtpmap:106 L16/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> ------------------------------------------------------------------------
> 2011-04-01 23:34:11.674022 [DEBUG] sofia.c:4754 Channel sofia/internal/
sip:99999999999 at 11.22.33.44:52767 entering state [terminated][408]
> 2011-04-01 23:34:11.674022 [DEBUG] switch_channel.c:2563 (sofia/internal/
sip:99999999999 at 11.22.33.44:52767) Callstate Change RINGING -> HANGUP
> 2011-04-01 23:34:11.674022 [NOTICE] sofia.c:5394 Hangup sofia/internal/
sip:99999999999 at 11.22.33.44:52767 [CS_CONSUME_MEDIA]
[RECOVERY_ON_TIMER_EXPIRE]
> 2011-04-01 23:34:11.674022 [DEBUG] switch_channel.c:2579 Send signal
sofia/internal/sip:99999999999 at 11.22.33.44:52767 [KILL]
> 2011-04-01 23:34:11.674022 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/sip:99999999999 at 11.22.33.44:52767 [BREAK]
> 2011-04-01 23:34:11.674022 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/sip:99999999999 at 11.22.33.44:52767) Running State Change
CS_HANGUP
> 2011-04-01 23:34:11.674022 [DEBUG] switch_ivr_originate.c:3506 Originate
Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
> 2011-04-01 23:34:11.674022 [ERR] switch_ivr_originate.c:2640 Cannot create
outgoing channel of type [user] cause: [RECOVERY_ON_TIMER_EXPIRE]
> 2011-04-01 23:34:11.674022 [DEBUG] switch_ivr_originate.c:3506 Originate
Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
> 2011-04-01 23:34:11.675057 [DEBUG] switch_core_state_machine.c:560
(sofia/internal/sip:99999999999 at 11.22.33.44:52767) State HANGUP
> 2011-04-01 23:34:11.675057 [DEBUG] mod_sofia.c:451 sofia/internal/
sip:99999999999 at 11.22.33.44:52767 Overriding SIP cause 504 with 408 from the
other leg
> 2011-04-01 23:34:11.675057 [DEBUG] mod_sofia.c:457 Channel sofia/internal/
sip:99999999999 at 11.22.33.44:52767 hanging up, cause:
RECOVERY_ON_TIMER_EXPIRE
> 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:46
sofia/internal/sip:99999999999 at 11.22.33.44:52767 Standard HANGUP, cause:
RECOVERY_ON_TIMER_EXPIRE
> 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:560
(sofia/internal/sip:99999999999 at 11.22.33.44:52767) State HANGUP going to
sleep
> 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:351
(sofia/internal/sip:99999999999 at 11.22.33.44:52767) State Change CS_HANGUP ->
CS_REPORTING
> 2011-04-01 23:34:11.695371 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/sip:99999999999 at 11.22.33.44:52767 [BREAK]
> 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:320
(sofia/internal/sip:99999999999 at 11.22.33.44:52767) Running State Change
CS_REPORTING
> 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:620
(sofia/internal/sip:99999999999 at 11.22.33.44:52767) State REPORTING
> 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:53
sofia/internal/sip:99999999999 at 11.22.33.44:52767 Standard REPORTING, cause:
RECOVERY_ON_TIMER_EXPIRE
> 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:620
(sofia/internal/sip:99999999999 at 11.22.33.44:52767) State REPORTING going to
sleep
> 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:345
(sofia/internal/sip:99999999999 at 11.22.33.44:52767) State Change CS_REPORTING
-> CS_DESTROY
> 2011-04-01 23:34:11.695371 [DEBUG] switch_core_session.c:1116 Send signal
sofia/internal/sip:99999999999 at 11.22.33.44:52767 [BREAK]
> 2011-04-01 23:34:11.695371 [DEBUG] switch_core_session.c:1288 Session 5
(sofia/internal/sip:99999999999 at 11.22.33.44:52767) Locked, Waiting on
external entities
> 2011-04-01 23:34:11.695371 [NOTICE] switch_core_session.c:1306 Session 5
(sofia/internal/sip:99999999999 at 11.22.33.44:52767) Ended
> 2011-04-01 23:34:11.695371 [NOTICE] switch_core_session.c:1308 Close
Channel sofia/internal/sip:99999999999 at 11.22.33.44:52767 [CS_DESTROY]
> 2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:449
(sofia/internal/sip:99999999999 at 11.22.33.44:52767) Callstate Change HANGUP
-> DOWN
> 2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:452
(sofia/internal/sip:99999999999 at 11.22.33.44:52767) Running State Change
CS_DESTROY
> 2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:462
(sofia/internal/sip:99999999999 at 11.22.33.44:52767) State DESTROY
> 2011-04-01 23:34:11.696436 [DEBUG] mod_sofia.c:362 sofia/internal/
sip:99999999999 at 11.22.33.44:52767 SOFIA DESTROY
> 2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:60
sofia/internal/sip:99999999999 at 11.22.33.44:52767 Standard DESTROY
> 2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:462
(sofia/internal/sip:99999999999 at 11.22.33.44:52767) State DESTROY going to
sleep
>
>
> a
>
> ----- Original Message -----
> From: Steven Ayre
> To: FreeSWITCH Users Help
> Sent: Friday, April 01, 2011 5:30 PM
> Subject: Re: [Freeswitch-users] originate from cli
>
>
> It can also be that the other side sent FS a reply saying that *it* had
timed out. siptrace will also show if that's the case.
>
>
>
> On 1 April 2011 22:29, Steven Ayre <steveayre at gmail.com> wrote:
>
> It means FS sent a message and didn't get a reply (timed out).
>
> As anthm says, look at the siptrace - that'll show you what's being sent /
received.
>
> -Steve
>
>
>
> On 1 April 2011 21:17, Madovsky <infos at madovsky.org> wrote:
>
> I make some test with originate from cli.
>
> /usr/local/freeswitch/bin/fs_cli -x "originate user/9999 7777 XML public"
> -ERR RECOVERY_ON_TIMER_EXPIRE
>
> 9999 and 7777 are ready to receive calls and I don't have any NAT.
>
> is anyone can explain what it means ?
>
> Thanks
>
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> http://www.freeswitch.org
>
>
>
>
>
>
>
>
------------------------------------------------------------------------------
>
>
> _______________________________________________
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> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From infos at madovsky.org  Sat Apr  2 23:51:39 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 2 Apr 2011 15:51:39 -0400
Subject: [Freeswitch-users] originate from cli
References: <F38382DFFBE041AF8562F02269FE45EC@e1705><AANLkTiks_xJX8vyX-QWZ+ar+Bwhn0Mm6TPvcp4=_39DC@mail.gmail.com><AANLkTik3or8JqMUmMwaYxgSumjD+Hm+HBtCh_WLusk6=@mail.gmail.com><C66F263892AE4887B47D8D6B0B3FA4D0@e1705>
	<BANLkTi=37Pv9Dyb4ughCA1_F4NapoZNt7A@mail.gmail.com>
Message-ID: <8565DD398F9A447B840F92232A44AB6D@e1705>

Ok I'm trying to figure it out
thanks
  ----- Original Message ----- 
  From: Anthony Minessale 
  To: FreeSWITCH Users Help 
  Sent: Saturday, April 02, 2011 3:34 PM
  Subject: Re: [Freeswitch-users] originate from cli


  You never get the reply to the invite trace from the thing you are calling and see if they are getting any packets.  Maybe you have iptables or selinux on on one host or the other.

  On Apr 1, 2011 10:37 PM, "Madovsky" <infos at madovsky.org> wrote:
  > sorry the sip trace :
  > after this command:
  > 
  > /usr/local/freeswitch/bin/fs_cli -x "originate user/9999999999 1111111111"
  > 
  > ------------------------------------------------------------------------
  > 2011-04-01 23:33:39.673581 [DEBUG] sofia.c:4754 Channel sofia/internal/sip:99999999999 at 11.22.33.44:52767 entering state [calling][0]
  > send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:40.673952:
  > ------------------------------------------------------------------------
  > INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
  > Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
  > Max-Forwards: 70
  > From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
  > To: <sip:99999999999 at 11.22.33.44:52767>
  > Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
  > CSeq: 10521545 INVITE
  > Contact: <sip:mod_sofia at 11.22.33.44:5080>
  > User-Agent: CiscoSystems-SIP-GW-UA
  > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  > Supported: timer, precondition, path, replaces
  > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  > Content-Type: application/sdp
  > Content-Disposition: session
  > Content-Length: 492
  > X-FS-Support: update_display
  > Remote-Party-ID: <sip:0000000000 at 11.22.33.44>;party=calling;screen=yes;privacy=off
  > 
  > v=0
  > o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
  > s=FreeSWITCH
  > c=IN IP4 11.22.33.44
  > t=0 0
  > m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
  > a=rtpmap:98 SPEEX/16000
  > a=rtpmap:99 G726-16/8000
  > a=rtpmap:100 G726-24/8000
  > a=rtpmap:102 G726-32/8000
  > a=rtpmap:103 G726-40/8000
  > a=rtpmap:104 G7221/16000
  > a=fmtp:104 bitrate=32000
  > a=rtpmap:105 iLBC/8000
  > a=fmtp:105 mode=20
  > a=rtpmap:106 L16/8000
  > a=rtpmap:101 telephone-event/8000
  > a=fmtp:101 0-16
  > a=ptime:20
  > ------------------------------------------------------------------------
  > send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:42.673971:
  > ------------------------------------------------------------------------
  > INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
  > Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
  > Max-Forwards: 70
  > From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
  > To: <sip:99999999999 at 11.22.33.44:52767>
  > Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
  > CSeq: 10521545 INVITE
  > Contact: <sip:mod_sofia at 11.22.33.44:5080>
  > User-Agent: CiscoSystems-SIP-GW-UA
  > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  > Supported: timer, precondition, path, replaces
  > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  > Content-Type: application/sdp
  > Content-Disposition: session
  > Content-Length: 492
  > X-FS-Support: update_display
  > Remote-Party-ID: <sip:0000000000 at 11.22.33.44>;party=calling;screen=yes;privacy=off
  > 
  > v=0
  > o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
  > s=FreeSWITCH
  > c=IN IP4 11.22.33.44
  > t=0 0
  > m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
  > a=rtpmap:98 SPEEX/16000
  > a=rtpmap:99 G726-16/8000
  > a=rtpmap:100 G726-24/8000
  > a=rtpmap:102 G726-32/8000
  > a=rtpmap:103 G726-40/8000
  > a=rtpmap:104 G7221/16000
  > a=fmtp:104 bitrate=32000
  > a=rtpmap:105 iLBC/8000
  > a=fmtp:105 mode=20
  > a=rtpmap:106 L16/8000
  > a=rtpmap:101 telephone-event/8000
  > a=fmtp:101 0-16
  > a=ptime:20
  > ------------------------------------------------------------------------
  > send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:46.673944:
  > ------------------------------------------------------------------------
  > INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
  > Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
  > Max-Forwards: 70
  > From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
  > To: <sip:99999999999 at 11.22.33.44:52767>
  > Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
  > CSeq: 10521545 INVITE
  > Contact: <sip:mod_sofia at 11.22.33.44:5080>
  > User-Agent: CiscoSystems-SIP-GW-UA
  > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  > Supported: timer, precondition, path, replaces
  > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  > Content-Type: application/sdp
  > Content-Disposition: session
  > Content-Length: 492
  > X-FS-Support: update_display
  > Remote-Party-ID: <sip:0000000000 at 11.22.33.44>;party=calling;screen=yes;privacy=off
  > 
  > v=0
  > o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
  > s=FreeSWITCH
  > c=IN IP4 11.22.33.44
  > t=0 0
  > m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
  > a=rtpmap:98 SPEEX/16000
  > a=rtpmap:99 G726-16/8000
  > a=rtpmap:100 G726-24/8000
  > a=rtpmap:102 G726-32/8000
  > a=rtpmap:103 G726-40/8000
  > a=rtpmap:104 G7221/16000
  > a=fmtp:104 bitrate=32000
  > a=rtpmap:105 iLBC/8000
  > a=fmtp:105 mode=20
  > a=rtpmap:106 L16/8000
  > a=rtpmap:101 telephone-event/8000
  > a=fmtp:101 0-16
  > a=ptime:20
  > ------------------------------------------------------------------------
  > send 1385 bytes to udp/[11.22.33.44]:52767 at 03:33:54.673950:
  > ------------------------------------------------------------------------
  > INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
  > Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
  > Max-Forwards: 70
  > From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
  > To: <sip:99999999999 at 11.22.33.44:52767>
  > Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
  > CSeq: 10521545 INVITE
  > Contact: <sip:mod_sofia at 11.22.33.44:5080>
  > User-Agent: CiscoSystems-SIP-GW-UA
  > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  > Supported: timer, precondition, path, replaces
  > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  > Content-Type: application/sdp
  > Content-Disposition: session
  > Content-Length: 492
  > X-FS-Support: update_display
  > Remote-Party-ID: <sip:0000000000 at 11.22.33.44>;party=calling;screen=yes;privacy=off
  > 
  > v=0
  > o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
  > s=FreeSWITCH
  > c=IN IP4 11.22.33.44
  > t=0 0
  > m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
  > a=rtpmap:98 SPEEX/16000
  > a=rtpmap:99 G726-16/8000
  > a=rtpmap:100 G726-24/8000
  > a=rtpmap:102 G726-32/8000
  > a=rtpmap:103 G726-40/8000
  > a=rtpmap:104 G7221/16000
  > a=fmtp:104 bitrate=32000
  > a=rtpmap:105 iLBC/8000
  > a=fmtp:105 mode=20
  > a=rtpmap:106 L16/8000
  > a=rtpmap:101 telephone-event/8000
  > a=fmtp:101 0-16
  > a=ptime:20
  > ------------------------------------------------------------------------
  > send 1385 bytes to udp/[11.22.33.44]:52767 at 03:34:10.674077:
  > ------------------------------------------------------------------------
  > INVITE sip:99999999999 at 11.22.33.44:52767 SIP/2.0
  > Via: SIP/2.0/UDP 11.22.33.44:5080;rport;branch=z9hG4bK415Sta2FX5NrD
  > Max-Forwards: 70
  > From: "" <sip:0000000000 at 11.22.33.44>;tag=S16mjZ06UgQSe
  > To: <sip:99999999999 at 11.22.33.44:52767>
  > Call-ID: 02ac42f6-92d1-4f3b-bc98-aa00d5f01af5
  > CSeq: 10521545 INVITE
  > Contact: <sip:mod_sofia at 11.22.33.44:5080>
  > User-Agent: CiscoSystems-SIP-GW-UA
  > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
  > Supported: timer, precondition, path, replaces
  > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
  > Content-Type: application/sdp
  > Content-Disposition: session
  > Content-Length: 492
  > X-FS-Support: update_display
  > Remote-Party-ID: <sip:0000000000 at 11.22.33.44>;party=calling;screen=yes;privacy=off
  > 
  > v=0
  > o=FreeSWITCH 1301682069 1301682070 IN IP4 11.22.33.44
  > s=FreeSWITCH
  > c=IN IP4 11.22.33.44
  > t=0 0
  > m=audio 33150 RTP/AVP 98 0 8 3 99 100 102 103 104 9 105 5 106 101 13
  > a=rtpmap:98 SPEEX/16000
  > a=rtpmap:99 G726-16/8000
  > a=rtpmap:100 G726-24/8000
  > a=rtpmap:102 G726-32/8000
  > a=rtpmap:103 G726-40/8000
  > a=rtpmap:104 G7221/16000
  > a=fmtp:104 bitrate=32000
  > a=rtpmap:105 iLBC/8000
  > a=fmtp:105 mode=20
  > a=rtpmap:106 L16/8000
  > a=rtpmap:101 telephone-event/8000
  > a=fmtp:101 0-16
  > a=ptime:20
  > ------------------------------------------------------------------------
  > 2011-04-01 23:34:11.674022 [DEBUG] sofia.c:4754 Channel sofia/internal/sip:99999999999 at 11.22.33.44:52767 entering state [terminated][408]
  > 2011-04-01 23:34:11.674022 [DEBUG] switch_channel.c:2563 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) Callstate Change RINGING -> HANGUP
  > 2011-04-01 23:34:11.674022 [NOTICE] sofia.c:5394 Hangup sofia/internal/sip:99999999999 at 11.22.33.44:52767 [CS_CONSUME_MEDIA] [RECOVERY_ON_TIMER_EXPIRE]
  > 2011-04-01 23:34:11.674022 [DEBUG] switch_channel.c:2579 Send signal sofia/internal/sip:99999999999 at 11.22.33.44:52767 [KILL]
  > 2011-04-01 23:34:11.674022 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:99999999999 at 11.22.33.44:52767 [BREAK]
  > 2011-04-01 23:34:11.674022 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) Running State Change CS_HANGUP
  > 2011-04-01 23:34:11.674022 [DEBUG] switch_ivr_originate.c:3506 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
  > 2011-04-01 23:34:11.674022 [ERR] switch_ivr_originate.c:2640 Cannot create outgoing channel of type [user] cause: [RECOVERY_ON_TIMER_EXPIRE]
  > 2011-04-01 23:34:11.674022 [DEBUG] switch_ivr_originate.c:3506 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE]
  > 2011-04-01 23:34:11.675057 [DEBUG] switch_core_state_machine.c:560 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State HANGUP
  > 2011-04-01 23:34:11.675057 [DEBUG] mod_sofia.c:451 sofia/internal/sip:99999999999 at 11.22.33.44:52767 Overriding SIP cause 504 with 408 from the other leg
  > 2011-04-01 23:34:11.675057 [DEBUG] mod_sofia.c:457 Channel sofia/internal/sip:99999999999 at 11.22.33.44:52767 hanging up, cause: RECOVERY_ON_TIMER_EXPIRE
  > 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:46 sofia/internal/sip:99999999999 at 11.22.33.44:52767 Standard HANGUP, cause: RECOVERY_ON_TIMER_EXPIRE
  > 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:560 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State HANGUP going to sleep
  > 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:351 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State Change CS_HANGUP -> CS_REPORTING
  > 2011-04-01 23:34:11.695371 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:99999999999 at 11.22.33.44:52767 [BREAK]
  > 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:320 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) Running State Change CS_REPORTING
  > 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State REPORTING
  > 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:53 sofia/internal/sip:99999999999 at 11.22.33.44:52767 Standard REPORTING, cause: RECOVERY_ON_TIMER_EXPIRE
  > 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:620 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State REPORTING going to sleep
  > 2011-04-01 23:34:11.695371 [DEBUG] switch_core_state_machine.c:345 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State Change CS_REPORTING -> CS_DESTROY
  > 2011-04-01 23:34:11.695371 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:99999999999 at 11.22.33.44:52767 [BREAK]
  > 2011-04-01 23:34:11.695371 [DEBUG] switch_core_session.c:1288 Session 5 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) Locked, Waiting on external entities
  > 2011-04-01 23:34:11.695371 [NOTICE] switch_core_session.c:1306 Session 5 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) Ended
  > 2011-04-01 23:34:11.695371 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/sip:99999999999 at 11.22.33.44:52767 [CS_DESTROY]
  > 2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:449 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) Callstate Change HANGUP -> DOWN
  > 2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) Running State Change CS_DESTROY
  > 2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State DESTROY
  > 2011-04-01 23:34:11.696436 [DEBUG] mod_sofia.c:362 sofia/internal/sip:99999999999 at 11.22.33.44:52767 SOFIA DESTROY
  > 2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:60 sofia/internal/sip:99999999999 at 11.22.33.44:52767 Standard DESTROY
  > 2011-04-01 23:34:11.696436 [DEBUG] switch_core_state_machine.c:462 (sofia/internal/sip:99999999999 at 11.22.33.44:52767) State DESTROY going to sleep
  > 
  > 
  > a
  > 
  > ----- Original Message ----- 
  > From: Steven Ayre 
  > To: FreeSWITCH Users Help 
  > Sent: Friday, April 01, 2011 5:30 PM
  > Subject: Re: [Freeswitch-users] originate from cli
  > 
  > 
  > It can also be that the other side sent FS a reply saying that *it* had timed out. siptrace will also show if that's the case.
  > 
  > 
  > 
  > On 1 April 2011 22:29, Steven Ayre <steveayre at gmail.com> wrote:
  > 
  > It means FS sent a message and didn't get a reply (timed out).
  > 
  > As anthm says, look at the siptrace - that'll show you what's being sent / received.
  > 
  > -Steve
  > 
  > 
  > 
  > On 1 April 2011 21:17, Madovsky <infos at madovsky.org> wrote:
  > 
  > I make some test with originate from cli.
  > 
  > /usr/local/freeswitch/bin/fs_cli -x "originate user/9999 7777 XML public"
  > -ERR RECOVERY_ON_TIMER_EXPIRE
  > 
  > 9999 and 7777 are ready to receive calls and I don't have any NAT.
  > 
  > is anyone can explain what it means ?
  > 
  > 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



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From ibc at aliax.net  Sun Apr  3 00:57:12 2011
From: ibc at aliax.net (=?UTF-8?Q?I=C3=B1aki_Baz_Castillo?=)
Date: Sat, 2 Apr 2011 22:57:12 +0200
Subject: [Freeswitch-users] Why FS rewrites From header?
In-Reply-To: <AANLkTinyqyXadnhF77KJRS1iMcJR51Pkwi-W_nNomRKy@mail.gmail.com>
References: <538261301575539@web100.yandex.ru>
	<AANLkTin2X8FvBhPf65rDXqDXMv8O36FwUS_Par9iPR3s@mail.gmail.com>
	<BANLkTimm5nhyWEw33QeyRmpHgmhB-nmKig@mail.gmail.com>
	<BANLkTin8gut1aqo172QB=XznOGX-snx8Ww@mail.gmail.com>
	<4D955D71.90108@opensipstack.org>
	<BANLkTimPAxvssD3kVdDa131X3eqFuXPd0Q@mail.gmail.com>
	<1066491301670118@web9.yandex.ru>
	<BANLkTinMHRsqcKa=zoM6rZ+bjPF3nu69RQ@mail.gmail.com>
	<AANLkTinyqyXadnhF77KJRS1iMcJR51Pkwi-W_nNomRKy@mail.gmail.com>
Message-ID: <BANLkTimnMyxug05V2=VUA1JKr4Z25Unxxg@mail.gmail.com>

2011/4/1 Anthony Minessale <anthony.minessale at gmail.com>:
> Everyone wants the way they want it to work in their specific single
> use case to be the default.
>
> It's not a hack, it's the way you want it solved by a documented
> config option and its not any more ugly than a cisco dial-plan is it?
>
> FreeSWITCH can be mostly anything you want it to be, besides a proxy.
> It's your job to configure it how you would like.
>
> For your connivence, latest git has a new option you can specify in
> the from-domain param on a gateway xml to "auto-aleg" indicating you
> want this behavior that in now way should be the default.......

So, in case FS receives an INVITE with "From:
sip:alice at example.org;custom-param=abc" and the sofia profile has
"auto-aleg"="yes", would FS keep the original From URI in the outbound
leg?

If so, that's really good :)

Thanks a lot.


-- 
I?aki Baz Castillo
<ibc at aliax.net>


From anthony.minessale at gmail.com  Sun Apr  3 01:48:18 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 2 Apr 2011 16:48:18 -0500
Subject: [Freeswitch-users] Dingaling and sasl authentication failed
In-Reply-To: <1301743757340-6233670.post@n2.nabble.com>
References: <4D9127CD.7060300@telefaks.de>
	<1301743757340-6233670.post@n2.nabble.com>
Message-ID: <BANLkTimEczTy50PV2V2QZmuwk8GrZpP0Zg@mail.gmail.com>

The iksemel lib we use does not have support for srv records.  So if
the auth is really done to some remote server, you will have to
specify it manually in the server option.  See the default for gmail,
googlemail (the euro version may have a different alternate server"

Try doing a naptr or srv lookup on it.


On Sat, Apr 2, 2011 at 6:29 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
> I don't know if this will help or not. But, so far the only dingaling error
> messages found in /var/log/freeswitch/freeswitch.log file on my FS (running
> on FreeSWITCH Version 1.0.head (git-9795dd2 2011-03-26 11-07-34 -0500)) is
> shown below:
> 2011-03-31 13:22:30.718490 [DEBUG] libdingaling.c:1610 io error 2 7 retry in
> 3 second(s)
> 2011-03-31 13:22:34.171096 [DEBUG] libdingaling.c:1297 XMPP server connected
> 2011-03-31 13:22:34.307809 [DEBUG] libdingaling.c:1309 XMPP authenticated
>
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dingaling-and-sasl-authentication-failed-tp6217329p6233670.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
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From lists at telefaks.de  Sun Apr  3 03:55:20 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Sun, 03 Apr 2011 01:55:20 +0200
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <4D972C23.60305@KennedySoftware.ie>
References: <4D94C282.1090903@KennedySoftware.ie>	<1301750587582-6233798.post@n2.nabble.com>
	<4D972C23.60305@KennedySoftware.ie>
Message-ID: <4D97B768.1080006@telefaks.de>

For a desktop phone you may also use a powerline adapter when no copper
network is there.

Best regards
Peter


Michael Kennedy schrieb:
>> I expected that many phone suppliers would have handsets with EITHER
>>     
>>> RJ45 or WiFi connectivity to the LAN, or even both! I've found only a
>>> single device, a Cisco SPA525G2! Furthermore, searching the FS site, and
>>> various VoIP sites, and running general searches, I've found no other
>>> SIP WiFi phones that look like standard desktop handsets.
>>>       
>
>   
>> These days, more and more people are using smartphone.
>>     
>
> Yes, I appreciate that.
>
> However, *I*'m very old!, and probably much too conservative, as are 
> some clients - though we're still using lots of Linux, IT, and, maybe, FS!
>
> Some employees already use smartphones, but I believe the client is 
> REDUCING the support and usage of these, because of the very high costs, 
> and no proportional benefits, and opportunities to just "waste" lots of 
> employee time on them, etc...
>
>    - 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
>
>   


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From infos at madovsky.org  Sun Apr  3 07:40:03 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 2 Apr 2011 23:40:03 -0400
Subject: [Freeswitch-users] FS and Skype conference
Message-ID: <6CF06FA04FA5441EA931E7877EB06678@e1705>

I read on skype the article of conference call at
https://support.skype.com/en/faq/FA2831/How-do-I-start-a-conference-call

I'm curious to know with what technology they use to offer to hundred of 
thousands people
confernce with 25 people without big latency CPU load and bandwidth 
problem..

any spy ? 



From infos at madovsky.org  Sun Apr  3 08:30:36 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 3 Apr 2011 00:30:36 -0400
Subject: [Freeswitch-users] fax receive last git
Message-ID: <76E2E125F6BF41288935A3473C27F359@e1705>

just updated 1 hour ago and fax receive doesn't work anymore

my dialplan

                <extension name="receivefax">
                        <condition field="destination_number" expression="^receivefax$">
                                <action application="set" data="is_fax=FAX incoming..."/>
                                <action application="set" data="session_in_hangup_hook=true"/>
                                <action application="answer"/>
                                <action application="playback" data="silence_stream://2000"/>
                                <action application="set" data="fax_enable_t38_request=true"/>
                                <action application="set" data="fax_enable_t38=true"/>
                                <action application="rxfax" data="/path/files/${dialed_extension}/rxfax-${caller_id_number}-${uuid}.tiff"/>
                                <action application="hangup"/>
                        </condition>
                </extension>



2011-04-03 00:19:09.904843 [DEBUG] mod_spandsp_fax.c:1103 Raw read codec activation Success L16 20000
2011-04-03 00:19:09.904843 [DEBUG] switch_core_codec.c:116 sofia/external/9999 at domain.ltd Push codec L16:70
2011-04-03 00:19:09.905869 [DEBUG] mod_spandsp_fax.c:1119 Raw write codec activation Success L16
2011-04-03 00:19:10.193804 [NOTICE] switch_core_io.c:883 Deactivating write resampler
2011-04-03 00:19:10.554031 [DEBUG] switch_rtp.c:3082 Correct ip/port confirmed.
2011-04-03 00:19:10.593544 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:10.634032 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:10.673586 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:10.673586 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:10.693816 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:10.733338 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:10.793154 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:10.833815 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:10.853076 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:10.933105 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:11.033364 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:11.053605 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:11.073871 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:11.073871 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:11.093110 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:11.234012 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:11.253342 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:11.314084 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:11.394011 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:11.394011 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:11.413241 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:11.553164 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:11.553164 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:11.633239 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:11.653479 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:11.773073 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:11.773073 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:11.773073 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:12.273076 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:12.313701 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:12.353307 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:12.433831 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:12.473426 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:12.653861 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:12.813980 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:12.853491 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:12.873763 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:12.974095 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:13.093808 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:13.233665 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:13.273146 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:13.313686 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:13.413536 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:13.433820 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:13.613312 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:13.773450 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:14.053085 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:14.213363 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:14.493532 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:14.553322 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:14.613161 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:14.653775 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:14.893958 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:15.033711 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:15.093547 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:15.153355 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:15.193876 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:15.433555 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:15.553156 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:15.693274 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:15.733782 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:15.773272 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:15.873696 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:15.973118 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:16.113876 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:16.373909 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:16.433665 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:16.493443 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:16.813666 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:16.973216 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:17.033237 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:17.133924 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:17.253513 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:17.493360 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:17.573359 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:17.633177 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:17.753715 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:17.833839 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:17.833839 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:17.933132 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:17.953406 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:18.193689 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:18.253792 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:18.393098 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:18.593729 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:18.653484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:18.673705 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:18.713279 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:18.854018 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:18.893600 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:18.913875 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:19.193602 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:19.293926 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:19.373446 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:19.414013 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:19.554081 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:19.613097 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:19.613097 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:19.633465 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:19.674069 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:19.713603 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:19.833360 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:19.953201 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:20.333416 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:20.393534 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:20.493840 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:20.513116 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:20.593189 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:20.793800 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:20.873803 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:20.973126 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:20.973126 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:21.173743 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:21.213364 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:21.313608 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:21.413325 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:21.473102 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:21.553142 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:21.633148 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:21.694037 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:21.774069 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:21.914023 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:21.953610 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:21.973915 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:22.013481 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:22.013481 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:22.033714 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:22.193781 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:22.253560 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:22.533749 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:22.554000 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:22.713998 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:22.773659 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:22.813192 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:22.833482 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:22.953075 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:22.973382 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:22.973382 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:23.073710 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 18
2011-04-03 00:19:23.313959 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier up (-2) in state 18
2011-04-03 00:19:23.353550 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:23.413730 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:23.453242 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:23.514039 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:23.553547 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:23.613335 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:23.673163 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:23.713702 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:23.773443 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:23.813935 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:23.873822 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:23.933633 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:23.973201 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:24.033978 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:24.073535 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:24.133307 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:24.193140 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:24.233726 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18
2011-04-03 00:19:24.293513 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Abort (-8) in state 18





thanks



Franck
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From infos at madovsky.org  Sun Apr  3 11:46:29 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 3 Apr 2011 03:46:29 -0400
Subject: [Freeswitch-users] turn off cdr
Message-ID: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>

is it possible to deactivate completly any cdr without any side effect ?

Thanks
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From avi at avimarcus.net  Sun Apr  3 14:02:58 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 3 Apr 2011 13:02:58 +0300
Subject: [Freeswitch-users] turn off cdr
In-Reply-To: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>
References: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>
Message-ID: <BANLkTim-UV-S0-mYaur6wvoXZXTfbBOj6g@mail.gmail.com>

You can turn it off: http://wiki.freeswitch.org/wiki/Variable_process_cdr
 <http://wiki.freeswitch.org/wiki/Variable_process_cdr>No side effect? Well,
it won't affect the call flow (e.g. variables will still be set), but you
won't get any CDRs saved..
It's all modular, so the mod_xml_cdr or mod_cdr_csv not saving the CDRs
won't affect everything else..

(Oddly, you could even bill via mod_nibblebill but not keep CDRs. Sounds
like a bad idea though.)

-Avi

On Sun, Apr 3, 2011 at 10:46 AM, Madovsky <infos at madovsky.org> wrote:

>  is it possible to deactivate completly any cdr without any side effect ?
>
> 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
>
>
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From bearybeary7 at yahoo.com  Sun Apr  3 05:54:18 2011
From: bearybeary7 at yahoo.com (Beary Beary)
Date: Sat, 2 Apr 2011 18:54:18 -0700 (PDT)
Subject: [Freeswitch-users] T.38 faxing with Gafachi
Message-ID: <943031.33534.qm@web120903.mail.ne1.yahoo.com>

I'm trying to send a fax directly through CLI (no NAT present) "originate 
sofia/gateway/sip.gafachi.com/1234567890 &txfax(/tmp/file.tiff)"
I've followed directions on http://wiki.freeswitch.com/wiki/Mod_spandsp to set 
settings in my fax.conf:
<configuration name="fax.conf" description="FAX application configuration">
    <settings>
      <param name="use-ecm"value="false"/>
      <param name="verbose"value="true"/>
      <param name="disable-v17"value="false"/>
      <param name="enable-t38"        value="true"/>
      <param name="enable-t38-request" value="true"/>
      <param name="spool-dir"value="/tmp"/>
      <param name="file-prefix"value="faxrx"/>
    </settings>
</configuration>

My freeswitch server communicates with SIP-provider(Gafachi) using this profile:
<include>
  <gateway name="sip.gafachi.com">
    <param name="username" value="XXXX"/>
    <param name="password" value="XXXX"/>
    <param name="register" value="true"/>
    <param name="realm" value="67.216.35.162"/>
    <param name="proxy" value="67.216.35.162:5060"/>
  </gateway>
</include>


>From the logs I can't see that T.38 is used. SpanDSP says:

2011-03-29 23:13:39.101285 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ...0= 
Store and forward Internet fax (T.37): Not set
2011-03-29 23:13:39.101285 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... .0..= 
Real-time Internet fax (T.38): Not set
2011-03-29 23:13:39.101285 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... 0...= 
3G mobile network: Not set
2011-03-29 23:13:39.101285 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..1.= 
Receive fax: Set
2011-03-29 23:13:39.101285 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ..10 00..= 
Selected data signalling rate: V.17 14400bps
2011-03-29 23:13:39.101285 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .0.. ....= 
R8x7.7lines/mm and/or 200x200pels/25.4mm: Not set
2011-03-29 23:13:39.101285 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   0... ....= 
2-D coding: Not set
2011-03-29 23:13:39.101285 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..00= 
Recording width: 215mm +- 1%
2011-03-29 23:13:39.101285 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... 10..= 
Recording length: Unlimited
2011-03-29 23:13:39.101285 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .111 ....= 
Minimum scan line time: 0ms
2011-03-29 23:13:39.101285 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   0... ....= 
Extension indicator: Not set

In the end "fax is successfully sent" (via T4 to my understanding), but I only 
receive it on the other efax 1/20 times. What am I missing?
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From david.villasmil.work at gmail.com  Sun Apr  3 15:05:30 2011
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Sun, 3 Apr 2011 13:05:30 +0200
Subject: [Freeswitch-users] XML parser bug
In-Reply-To: <AANLkTimGoP07q3dzpvfEtdZNXCUo+acqJVB8EUswnj89@mail.gmail.com>
References: <AANLkTi=2zkLLKyLg9qaKDsF=y_oKqUEo_ErKwvEpgkvM@mail.gmail.com>
	<AANLkTimVxhaN4f3TU3Ni9vv0FxjRaYz=H1rGY9aZvFje@mail.gmail.com>
	<AANLkTinUhFVvbx364NVyNgC0_jApVyi9E+p8jb_59efw@mail.gmail.com>
	<AANLkTi=qJn+GL+4m6bXW2VSdMUDsWFcGqnh7WsY6P6it@mail.gmail.com>
	<AANLkTi=-yaOswXcV7iSt=r1QtAXPD7Eu8Za652XaZAez@mail.gmail.com>
	<AANLkTimGoP07q3dzpvfEtdZNXCUo+acqJVB8EUswnj89@mail.gmail.com>
Message-ID: <BANLkTik2KcQDS9k7V5BkziDueTu4LoN=vg@mail.gmail.com>

That's what I mean. (i am on 64bit, btw)

On Thu, Mar 31, 2011 at 2:35 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Which warnings?
>
> "WARNING: Wasting up to 8 megs of memory per thread." only appears if
> you're giving -waste
>
> "Error: stacksize x is too large" will only appear if you haven't set
> "ulimit -s" correctly. If you're on 64bit I don't think it appears at all.
>
> -Steve
>
>
>
>
> On 31 March 2011 11:36, David Villasmil <david.villasmil.work at gmail.com>wrote:
>
>> Hello,
>>
>> that's just testing :P i just don't like the warnings when testing
>> I don't run it like that for production.
>>
>> David
>>
>>
>> On Wed, Mar 30, 2011 at 11:55 PM, Eliot Gable <
>> egable+freeswitch at gmail.com> wrote:
>>
>>> On a side note, why are you running with -waste flag? You really should
>>> not be doing that unless you have very good and very specific reasons to do
>>> it and you know what that does and why you want to do it. Perhaps you do,
>>> but I would double check. Personally, I've run FS on several different
>>> versions of Linux without -waste for two years without ever needing it.
>>>
>>>
>>> On Wed, Mar 30, 2011 at 8:54 AM, David Villasmil <
>>> david.villasmil.work at gmail.com> wrote:
>>>
>>>> Hello Joao,
>>>>
>>>> Ok, thanks
>>>>
>>>> David
>>>>
>>>>
>>>> 2011/3/30 Jo?o Mesquita <jmesquita at freeswitch.org>
>>>>
>>>>> This is not a bug and has been discussed several times on this mailing
>>>>> list. You can't comment X-PRE-PROCESS tags like that. Make a quick google
>>>>> search and you'll find several discussions about that including an
>>>>> explanation from Tony on the subject.
>>>>>
>>>>> Regards,
>>>>> Jo?o Mesquita
>>>>>
>>>>>
>>>>>
>>>>> On Wed, Mar 30, 2011 at 9:21 AM, David Villasmil <
>>>>> david.villasmil.work at gmail.com> wrote:
>>>>>
>>>>>> Hello,
>>>>>>
>>>>>> I noticed the following:
>>>>>>
>>>>>> I have my sofia.conf.xml like this:
>>>>>>
>>>>>> <configuration name="sofia.conf" description="sofia Endpoint">
>>>>>>
>>>>>>   <global_settings>
>>>>>>     <param name="log-level" value="0"/>
>>>>>>
>>>>>>     <!-- <param name="auto-restart" value="false"/> -->
>>>>>>
>>>>>>     <param name="debug-presence" value="0"/>
>>>>>>   </global_settings>
>>>>>>
>>>>>>   <!--
>>>>>>            The rabbit hole goes deep.  This includes all the
>>>>>>            profiles in the sip_profiles directory that is up
>>>>>>            one level from this directory.
>>>>>>        <profiles>
>>>>>>          <X-PRE-PROCESS cmd="include" data="../sip_profiles/*.xml" />
>>>>>>        </profiles>
>>>>>>   -->
>>>>>>
>>>>>> </configuration>
>>>>>>
>>>>>> when I start FS, latest GIT:
>>>>>> freeswitch -version
>>>>>> FreeSWITCH version: 1.0.head (git-7e52acf 2011-03-28 22-18-47 -0500)
>>>>>>
>>>>>> I get the following output:
>>>>>>
>>>>>> ./freeswitch -waste
>>>>>> WARNING: Wasting up to 8 megs of memory per thread.
>>>>>> 2011-03-30 14:02:23.200097 [INFO] switch_event.c:615 Activate Eventing
>>>>>> Engine.
>>>>>> 2011-03-30 14:02:23.211052 [DEBUG] switch_event.c:594 Create event
>>>>>> dispatch thread 0
>>>>>> Cannot Initialize [[error near line 1521]: unclosed <!--]
>>>>>>
>>>>>> If i then change my sofia to:
>>>>>>
>>>>>> <configuration name="sofia.conf" description="sofia Endpoint">
>>>>>>
>>>>>>   <global_settings>
>>>>>>     <param name="log-level" value="0"/>
>>>>>>
>>>>>>     <!-- <param name="auto-restart" value="false"/> -->
>>>>>>
>>>>>>     <param name="debug-presence" value="0"/>
>>>>>>   </global_settings>
>>>>>>
>>>>>>   <!--
>>>>>>            The rabbit hole goes deep.  This includes all the
>>>>>>            profiles in the sip_profiles directory that is up
>>>>>>            one level from this directory.
>>>>>>        <profiles>
>>>>>>        </profiles>
>>>>>>   -->
>>>>>>
>>>>>> </configuration>
>>>>>>
>>>>>>
>>>>>> Please note the absence of:          <X-PRE-PROCESS cmd="include"
>>>>>> data="../sip_profiles/*.xml" />
>>>>>>
>>>>>>
>>>>>> FS Starts normally!
>>>>>>
>>>>>> Is this the correct behaviour? Isn't comments supposed NOT to be read?
>>>>>>
>>>>>> Thanks all.
>>>>>>
>>>>>>
>>>>>>
>>>>>> David
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Eliot Gable
>>>
>>> "We do not inherit the Earth from our ancestors: we borrow it from our
>>> children." ~David Brower
>>>
>>> "I decided the words were too conservative for me. We're not borrowing
>>> from our children, we're stealing from them--and it's not even considered to
>>> be a crime." ~David Brower
>>>
>>> "Esse oportet ut vivas, non vivere ut edas." (Thou shouldst eat to live;
>>> not live to eat.) ~Marcus Tullius Cicero
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble at slickdeals.endjunk.com  Sun Apr  3 15:17:37 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 3 Apr 2011 04:17:37 -0700 (PDT)
Subject: [Freeswitch-users] freeswitch-users@lists.freeswitch.org
In-Reply-To: <AANLkTikQn6b6M6WTTq=i=hVBHJF=agcE98zYAQK7Y=LH@mail.gmail.com>
References: <BLU0-SMTP32BB3CBC68FEC4D0E7B5FAF0A10@phx.gbl>
	<AANLkTikQn6b6M6WTTq=i=hVBHJF=agcE98zYAQK7Y=LH@mail.gmail.com>
Message-ID: <1301829457335-6234804.post@n2.nabble.com>


curriegrad2004 wrote:
> Can somebody moderate this person already?
Probably, his/her computer has been compromised and used to post spams by
spambots.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-users-lists-freeswitch-org-tp6234066p6234804.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From bwibowo at gmail.com  Sun Apr  3 15:28:44 2011
From: bwibowo at gmail.com (Budi wibowo)
Date: Sun, 3 Apr 2011 11:28:44 +0000
Subject: [Freeswitch-users] turn off cdr
In-Reply-To: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>
References: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>
Message-ID: <2016830847-1301830122-cardhu_decombobulator_blackberry.rim.net-502672334-@b4.c2.bise3.blackberry>

unload module for cdr may be?

-----Original Message-----
From: "Madovsky" <infos at madovsky.org>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Sun, 3 Apr 2011 03:46:29 
To: <freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] turn off cdr

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




From ejay.greeves at yahoo.com  Sun Apr  3 15:33:00 2011
From: ejay.greeves at yahoo.com (Ejay Greeves)
Date: Sun, 3 Apr 2011 12:33:00 +0100 (BST)
Subject: [Freeswitch-users] gateway port
Message-ID: <5160.32088.qm@web132305.mail.ird.yahoo.com>

I want to connect to a gateway which is configured on port 8081

What is the gateway param that sets the port on which to connect on the gateway
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From richocet2 at hotmail.com  Sun Apr  3 12:00:37 2011
From: richocet2 at hotmail.com (Dave Bracken)
Date: Sun, 3 Apr 2011 08:00:37 +0000
Subject: [Freeswitch-users] freeswitch-users@lists.freeswitch.org
Message-ID: <BLU0-SMTP115839926C081A1B79CD4BAF0A00@phx.gbl>

freeswitch-users at lists.freeswitch.org I have overcome many of lifes obstacles so many people were concerned about me this completely exceeded my expectations http://bit.ly/gvuOgB now im on the way to the top just looking out for you


From Info at KennedySoftware.ie  Sun Apr  3 16:55:52 2011
From: Info at KennedySoftware.ie (Michael Kennedy)
Date: Sun, 03 Apr 2011 13:55:52 +0100
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <4D97B768.1080006@telefaks.de>
References: <4D94C282.1090903@KennedySoftware.ie>	<1301750587582-6233798.post@n2.nabble.com>	<4D972C23.60305@KennedySoftware.ie>
	<4D97B768.1080006@telefaks.de>
Message-ID: <4D986E58.1090106@KennedySoftware.ie>

Thank you, Peter.

I also had not considered that option!

About 6-7-8+ years ago, we used these devices for some clients, and they 
worked reasonably well.

However, (for the sake of lurkers here!):
   - they were slow-ish,
   - they were very problematic in buildings where 3-phase was used, and 
in old buildings with mysterious/multiple mains runs,
   - they were very problematic if some desktop PCs had PSUs from 
specific manufacturers (presumably cheap-and-nasty devices). This was a 
huge problem, and required the replacement of apparently working PSUs is 
the PCs. No other office devices (faxes, monitors, photocopiers, 
kettles, microwaves even!, etc, etc, presented these problems).

So, we've not used them in the past few years - even though their 
performance might have improved significantly.

We'll certainly take another look at them, especially where the mains 
cabling is new/simple. Thank you.

   - Mike


On 03/04/2011 00:55, Peter Steinbach wrote:
> For a desktop phone you may also use a powerline adapter when no copper
> network is there.
>
> Best regards
> Peter


From richocet2 at hotmail.com  Sun Apr  3 15:37:02 2011
From: richocet2 at hotmail.com (Dave Bracken)
Date: Sun, 3 Apr 2011 11:37:02 +0000
Subject: [Freeswitch-users] freeswitch-users@lists.freeswitch.org
Message-ID: <BLU0-SMTP172FCCF01BA66884836C3A5F0A00@phx.gbl>

freeswitch-users at lists.freeswitch.org I hated being broke all the time I had tried everything I cant believe how much this exceeded expectations http://j.mp/fEj6bH now im headed straight for the top im telling you this is the real deal


From infos at madovsky.org  Sun Apr  3 20:13:13 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 3 Apr 2011 12:13:13 -0400
Subject: [Freeswitch-users] turn off cdr
References: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>
	<BANLkTim-UV-S0-mYaur6wvoXZXTfbBOj6g@mail.gmail.com>
Message-ID: <9B13E3A9A42F4156BBC841ADA4237EE7@e1705>

ok I wanted to be sure that turn off cdr was not a problem.for other modules (nibble bill for example)

> Oddly, you could even bill via mod_nibblebill but not keep CDRs. Sounds like a bad idea though.)
not so bad when you care of HD activity ;)
  ----- Original Message ----- 
  From: Avi Marcus 
  To: FreeSWITCH Users Help 
  Sent: Sunday, April 03, 2011 6:02 AM
  Subject: Re: [Freeswitch-users] turn off cdr


  You can turn it off: http://wiki.freeswitch.org/wiki/Variable_process_cdr
  No side effect? Well, it won't affect the call flow (e.g. variables will still be set), but you won't get any CDRs saved..
  It's all modular, so the mod_xml_cdr or mod_cdr_csv not saving the CDRs won't affect everything else..


  (Oddly, you could even bill via mod_nibblebill but not keep CDRs. Sounds like a bad idea though.)


  -Avi


  On Sun, Apr 3, 2011 at 10:46 AM, Madovsky <infos at madovsky.org> wrote:

    is it possible to deactivate completly any cdr without any side effect ?

    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






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


  _______________________________________________
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  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From anthony.minessale at gmail.com  Sun Apr  3 20:51:01 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sun, 3 Apr 2011 11:51:01 -0500
Subject: [Freeswitch-users] FS and Skype conference
In-Reply-To: <6CF06FA04FA5441EA931E7877EB06678@e1705>
References: <6CF06FA04FA5441EA931E7877EB06678@e1705>
Message-ID: <BANLkTimFnqvi13XTmPmK5eRE_jcEpYreew@mail.gmail.com>

its skype, they use your, and everyone else's PC as their network.


On Sat, Apr 2, 2011 at 10:40 PM, Madovsky <infos at madovsky.org> wrote:
> I read on skype the article of conference call at
> https://support.skype.com/en/faq/FA2831/How-do-I-start-a-conference-call
>
> I'm curious to know with what technology they use to offer to hundred of
> thousands people
> confernce with 25 people without big latency CPU load and bandwidth
> problem..
>
> any spy ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From infos at madovsky.org  Sun Apr  3 21:16:42 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 3 Apr 2011 13:16:42 -0400
Subject: [Freeswitch-users] turn off cdr
Message-ID: <53CE7EAEBA934FB1BDDECC22F7BB5B34@e1705>

apparently all cdr modules are commented in the module.con.xml.
but there's still log/cdr_csv and xml_cdr logs in logs folder... how to turn off ?
  ----- Original Message ----- 
  From: Madovsky 
  To: FreeSWITCH Users Help 
  Sent: Sunday, April 03, 2011 12:13 PM
  Subject: Re: [Freeswitch-users] turn off cdr


  ok I wanted to be sure that turn off cdr was not a problem.for other modules (nibble bill for example)

  > Oddly, you could even bill via mod_nibblebill but not keep CDRs. Sounds like a bad idea though.)
  not so bad when you care of HD activity ;)
    ----- Original Message ----- 
    From: Avi Marcus 
    To: FreeSWITCH Users Help 
    Sent: Sunday, April 03, 2011 6:02 AM
    Subject: Re: [Freeswitch-users] turn off cdr


    You can turn it off: http://wiki.freeswitch.org/wiki/Variable_process_cdr 
    No side effect? Well, it won't affect the call flow (e.g. variables will still be set), but you won't get any CDRs saved..
    It's all modular, so the mod_xml_cdr or mod_cdr_csv not saving the CDRs won't affect everything else..


    (Oddly, you could even bill via mod_nibblebill but not keep CDRs. Sounds like a bad idea though.)


    -Avi


    On Sun, Apr 3, 2011 at 10:46 AM, Madovsky <infos at madovsky.org> wrote:

      is it possible to deactivate completly any cdr without any side effect ?

      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






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


    _______________________________________________
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    FreeSWITCH-users at lists.freeswitch.org
    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From infos at madovsky.org  Sun Apr  3 21:18:45 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 3 Apr 2011 13:18:45 -0400
Subject: [Freeswitch-users] simultaneous voice conference question
Message-ID: <963D2073207646E2BAFF12EC8F8D0656@e1705>

When at least 3 persons are in conference I noticed
that we can't hear the 3 voices in same time, only one voice at a time.
is it normal ? 

thanks
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From infos at madovsky.org  Sun Apr  3 21:20:46 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 3 Apr 2011 13:20:46 -0400
Subject: [Freeswitch-users] turn off cdr
Message-ID: <68DB8FF6AD9245DDBBAA0BA6B1182C97@e1705>

sorry my bad it was old log

thanks
  ----- Original Message ----- 
  From: Madovsky 
  To: Madovsky ; FreeSWITCH Users Help 
  Sent: Sunday, April 03, 2011 1:16 PM
  Subject: Re: [Freeswitch-users] turn off cdr


  apparently all cdr modules are commented in the module.con.xml.
  but there's still log/cdr_csv and xml_cdr logs in logs folder... how to turn off ?
    ----- Original Message ----- 
    From: Madovsky 
    To: FreeSWITCH Users Help 
    Sent: Sunday, April 03, 2011 12:13 PM
    Subject: Re: [Freeswitch-users] turn off cdr


    ok I wanted to be sure that turn off cdr was not a problem.for other modules (nibble bill for example)

    > Oddly, you could even bill via mod_nibblebill but not keep CDRs. Sounds like a bad idea though.)
    not so bad when you care of HD activity ;)
      ----- Original Message ----- 
      From: Avi Marcus 
      To: FreeSWITCH Users Help 
      Sent: Sunday, April 03, 2011 6:02 AM
      Subject: Re: [Freeswitch-users] turn off cdr


      You can turn it off: http://wiki.freeswitch.org/wiki/Variable_process_cdr 
      No side effect? Well, it won't affect the call flow (e.g. variables will still be set), but you won't get any CDRs saved..
      It's all modular, so the mod_xml_cdr or mod_cdr_csv not saving the CDRs won't affect everything else..


      (Oddly, you could even bill via mod_nibblebill but not keep CDRs. Sounds like a bad idea though.)


      -Avi


      On Sun, Apr 3, 2011 at 10:46 AM, Madovsky <infos at madovsky.org> wrote:

        is it possible to deactivate completly any cdr without any side effect ?

        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






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


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      FreeSWITCH-users at lists.freeswitch.org
      http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
      UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From philippe at ppmt.org  Sun Apr  3 20:54:03 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Sun, 03 Apr 2011 12:54:03 -0400
Subject: [Freeswitch-users] incoming call stop working after a few minutes
Message-ID: <4D98A62B.5060303@ppmt.org>

Hi,

I have the latest FS installed on my Guruplug and it was working fine
overall

Last Friday I decided to rename the hostname of the guruplug and since
then I have problem with
incoming calls no longer going through. I can't see how the 2 can
related but that is the only thing I
can think of.

Also if I make a outgoing call then incoming will start working again
for a few minutes and stops again

The only thing I can see in the debug is that message:

2011-04-03 12:39:00.820680 [WARNING] sofia_reg.c:1246 SIP auth challenge
(REGISTER) on sofia profile 'internal' for [8926 at 172.21.0.20] from ip
172.21.0.50

But I was seeing it before as well so it can't be only that.

Is there some other trace I can activate to see more?

Regards

/Philippe

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From ovvenkatesan at gmail.com  Mon Apr  4 00:29:48 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Mon, 4 Apr 2011 01:59:48 +0530
Subject: [Freeswitch-users] How to set call max time
Message-ID: <BANLkTikLsnvE7AEEAi551bWxpmmCsCgHeg@mail.gmail.com>

Hi to all,

I dont know, how to set max time for a call.
for example,

call is landing from mobile to freeswitch.
I need to disconnect the call after 500 seconds if still connnected.

Which parameter I need to set to accomplish this situation .


Thanks and Regards
Venkat.
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From avi at avimarcus.net  Mon Apr  4 01:03:49 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 4 Apr 2011 00:03:49 +0300
Subject: [Freeswitch-users] How to set call max time
In-Reply-To: <BANLkTikLsnvE7AEEAi551bWxpmmCsCgHeg@mail.gmail.com>
References: <BANLkTikLsnvE7AEEAi551bWxpmmCsCgHeg@mail.gmail.com>
Message-ID: <BANLkTina3-Jc9whVnfOcZk-VdA9oH9cRKg@mail.gmail.com>

Try:  http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup
 <http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup>-Avi

On Sun, Apr 3, 2011 at 11:29 PM, ovvenkat <ovvenkatesan at gmail.com> wrote:

> Hi to all,
>
> I dont know, how to set max time for a call.
> for example,
>
> call is landing from mobile to freeswitch.
> I need to disconnect the call after 500 seconds if still connnected.
>
> Which parameter I need to set to accomplish this situation .
>
>
> Thanks and Regards
> Venkat.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sc_zhangming at sina.com  Sat Apr  2 18:38:37 2011
From: sc_zhangming at sina.com (=?gb2312?B?1cXD9w==?=)
Date: Sat, 2 Apr 2011 22:38:37 +0800
Subject: [Freeswitch-users] uuid_hold is not send hold message
Message-ID: <8slt5a$angvq4@irxd5-187.sinamail.sina.com.cn>

freeswitch-users????

	uuid_hold command , freeswitch is not send HOLD  message. who know it.

?????????
??
 				

??????????
????????sc_zhangming at sina.com
??????????2011-04-02

From rgelfand2 at gmail.com  Mon Apr  4 06:08:32 2011
From: rgelfand2 at gmail.com (Roman Gelfand)
Date: Sun, 3 Apr 2011 22:08:32 -0400
Subject: [Freeswitch-users] Vestec Connector
Message-ID: <BANLkTi=Q5p04bDr6OC34ToQsrhSMPoG52g@mail.gmail.com>

Is the vestec connector compatible with freeswitch 1.05?

Thanks in advance


From max.clark at gmail.com  Mon Apr  4 08:20:22 2011
From: max.clark at gmail.com (Max Clark)
Date: Sun, 3 Apr 2011 21:20:22 -0700
Subject: [Freeswitch-users] PRI Test Equipment
In-Reply-To: <AANLkTi=MbzOfMbuT9ePvFPo6rw+JwxmR=HFXAQU6DiqR@mail.gmail.com>
References: <AANLkTi=txKH5Yj3MHaMrX1uHhnqMpuS=PuLgGsxpR=H8@mail.gmail.com>
	<AANLkTi=emvxczkxopd4+OYLdY7gJjyd7qgONO8sK-PKT@mail.gmail.com>
	<AANLkTinexeb64gsV_ih8fwub1F=K8V+fKEJtEPUeWuwQ@mail.gmail.com>
	<2E80FBC43F4F464AB3730923CA7CC754@dell9400>
	<AANLkTi=t418uxQL37eaD-HCmky=7UOCPsH0NyLbkgybQ@mail.gmail.com>
	<AANLkTi=MbzOfMbuT9ePvFPo6rw+JwxmR=HFXAQU6DiqR@mail.gmail.com>
Message-ID: <BANLkTimtL=fzfQCjHYRQkvTQCoyJGyPHiw@mail.gmail.com>

Thank you - that looks perfect.

On Thu, Mar 31, 2011 at 10:48 AM, shouldbe q931 <shouldbeq931 at gmail.com> wrote:
>
>
> On Thu, Mar 31, 2011 at 4:41 PM, Max Clark <max.clark at gmail.com> wrote:
>>
>> Thanks Jan I'll check this out.
>>
>
> If you want to purchase a dedicated tester, the Trend Aurora is the one that
> I used to
> use?http://www.trendcomms.com/web2/pages.nsf/vlCookie/global$aurora%20sonata?opendocument&cc=true


From ayhkor at gmail.com  Mon Apr  4 08:31:00 2011
From: ayhkor at gmail.com (deniro)
Date: Mon, 4 Apr 2011 00:31:00 -0400
Subject: [Freeswitch-users] conf max-members is not enforced
Message-ID: <BANLkTindtKK==ZuC9YAqt1KNMem69vqV+w@mail.gmail.com>

I've been experimenting this
When phone callers  *first*  join to the conference "max-members" is not
enforced (more members can join than max-members value)
below is my piece of code.
(it works when web audio callers first  join to conference)
analysed freeswitch logs but couldn't get much clue.
any idea what might be causes for this behavior?



  $session->execute("conference","${PIN}@test_profile");

conference.conf.xml

 <profile name="test_profile">
 <param name="max-members" value="2"/>
      <param name="max-members-sound"
value="conference/conf-is-locked.wav"/>
........
........

thx
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From djbinter at gmail.com  Mon Apr  4 09:40:06 2011
From: djbinter at gmail.com (DJB International)
Date: Sun, 3 Apr 2011 22:40:06 -0700
Subject: [Freeswitch-users] conf max-members is not enforced
In-Reply-To: <BANLkTindtKK==ZuC9YAqt1KNMem69vqV+w@mail.gmail.com>
References: <BANLkTindtKK==ZuC9YAqt1KNMem69vqV+w@mail.gmail.com>
Message-ID: <BANLkTi=EP2SKxHBjvRrJUv=+LsVcESraqA@mail.gmail.com>

You should see something similar to this:  [NOTICE] mod_conference.c:5900
Conference 12345-1.2.3.4 is full.  I had it set up on git-6eba56d 2011-04-03
17-55-07 -0500 with no problem.

-djbinter


On Sun, Apr 3, 2011 at 9:31 PM, deniro <ayhkor at gmail.com> wrote:

> I've been experimenting this
> When phone callers  *first*  join to the conference "max-members" is not
> enforced (more members can join than max-members value)
> below is my piece of code.
> (it works when web audio callers first  join to conference)
> analysed freeswitch logs but couldn't get much clue.
> any idea what might be causes for this behavior?
>
>
>
>   $session->execute("conference","${PIN}@test_profile");
>
> conference.conf.xml
>
>  <profile name="test_profile">
>  <param name="max-members" value="2"/>
>       <param name="max-members-sound"
> value="conference/conf-is-locked.wav"/>
> ........
> ........
>
> 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
>
>
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From thomas at chaschperli.ch  Mon Apr  4 12:48:37 2011
From: thomas at chaschperli.ch (Thomas Mueller)
Date: Mon, 04 Apr 2011 10:48:37 +0200
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <4D94C282.1090903@KennedySoftware.ie>
References: <4D94C282.1090903@KennedySoftware.ie>
Message-ID: <4D9985E5.2060102@chaschperli.ch>


> I expected that many phone suppliers would have handsets with EITHER
> RJ45 or WiFi connectivity to the LAN, or even both! I've found only a
> single device, a Cisco SPA525G2! Furthermore, searching the FS site, and
> various VoIP sites, and running general searches, I've found no other
> SIP WiFi phones that look like standard desktop handsets.
>
> I'd appreciate any pointers to WiFi devices that are recommended with
> FS. Preferably "standard-looking" desktop units, and better still, if
> they had wired "sisters" - in appearance and functionality!

If your clients have "smart"-phones with WLAN :

http://www.counterpath.com/bria-android-edition.html (offers G.729, 
haven't used it)

http://www.sipdroid.com/ (using this for myself with FS)

For sure there are more solutions out there.

- Thomas



From freeswitch at priv.de  Mon Apr  4 13:07:38 2011
From: freeswitch at priv.de (Markus Mueller)
Date: Mon, 04 Apr 2011 11:07:38 +0200
Subject: [Freeswitch-users] BUG FIX: "Buffer size sanity check failed!"
	drops FAX receiving unneeded
Message-ID: <4D998A5A.6080901@priv.de>

Hello FreeSwitch users and programmers,

I found a problem on receiving faxes and want to share a working patch 
for this. The problem is that on receiving a fax, it is unneeded aborted 
by a sanity check. Sanity checks are fine, but a unneeded abort instead 
of a warning is in productive versions not the best solution.

The message apearing is:

2011-04-04 10:44:52.060860 [CRIT] switch_core_codec.c:660 Buffer size 
sanity check failed!

which is normaly aborting in receiving the fax. Simply decreasing this 
fault to a warning let the server receive the fax without any problems. 
After the patch the message apears up to five times per fax before the 
fax is beeing accepted. I am using libpri with the three HFC ISDN Cards 
and the DAHDI from Debian Squeeze 6.0. For more informations about my 
hardware just write me an email.

Regards,
Markus Mueller
http://projekte.priv.de/

root at sip:/usr/local/src/freeswitch/src# diff -U 4 switch_core_codec.c*
--- switch_core_codec.c 2011-03-14 10:49:17.000000000 +0100
+++ switch_core_codec.c.org     2011-03-14 10:47:02.000000000 +0100
@@ -657,9 +657,9 @@
                 uint32_t frames = encoded_data_len / 
codec->implementation->encoded_bytes_per_packet;

                 if (frames && 
codec->implementation->decoded_bytes_per_packet * frames > 
*decoded_data_len) {
                         switch_log_printf(SWITCH_CHANNEL_LOG, 
SWITCH_LOG_CRIT, "Buffer size sanity check failed!\n");
-                       // return SWITCH_STATUS_GENERR;
+                       return SWITCH_STATUS_GENERR;
                 }
         }

         if (codec->mutex) switch_mutex_lock(codec->mutex);
root at sip:/usr/local/src/freeswitch/src#



From Info at KennedySoftware.ie  Mon Apr  4 14:38:43 2011
From: Info at KennedySoftware.ie (Michael Kennedy)
Date: Mon, 04 Apr 2011 11:38:43 +0100
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <4D9985E5.2060102@chaschperli.ch>
References: <4D94C282.1090903@KennedySoftware.ie>
	<4D9985E5.2060102@chaschperli.ch>
Message-ID: <4D999FB3.5070300@KennedySoftware.ie>

> If your clients have "smart"-phones with WLAN :
>
> http://www.counterpath.com/bria-android-edition.html (offers G.729,
> haven't used it)
>
> http://www.sipdroid.com/ (using this for myself with FS)

These are most interesting links/products, Thomas. Thank you very much.

I knew nothing about either, and have spent the past 2 hours reading up 
about them - and have "only started"!

Very many thanks.
   - Mike


From frank at telonium.com  Mon Apr  4 19:53:24 2011
From: frank at telonium.com (Frank Park)
Date: Mon, 4 Apr 2011 11:53:24 -0400
Subject: [Freeswitch-users] xml_curl response for voicemail_inject
Message-ID: <BANLkTi=AwsfhJw7j6yLBBmy7hAUwVegkbA@mail.gmail.com>

I had a quick question.
I am trying to figure out what directory response is expected when a user
wants to forward a voicemail to another extension.
Looking at the post request of xml_curl, I see that it's invoking
voicemail_inject. I've tried the same response as other voicemail function,
which is similar to the authorization response, but that didn't seem to do
it. Anybody care to share an example response when voicemail_inject is
requested?

Thanks,
Frank

-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From jeff at jefflenk.com  Mon Apr  4 19:55:39 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 4 Apr 2011 08:55:39 -0700 (PDT)
Subject: [Freeswitch-users] BUG FIX: "Buffer size sanity check failed!"
 drops FAX receiving unneeded
In-Reply-To: <4D998A5A.6080901@priv.de>
References: <4D998A5A.6080901@priv.de>
Message-ID: <1301932539390-6239109.post@n2.nabble.com>

Please report this issue to Jira. http://jira.freeswitch.org

Include all relevent information including how to produce, debug logs and
any patch you may have with git diff

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/BUG-FIX-Buffer-size-sanity-check-failed-drops-FAX-receiving-unneeded-tp6237813p6239109.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brad at tritelcomm.com  Mon Apr  4 20:25:13 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Mon, 4 Apr 2011 09:25:13 -0700
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <4D999FB3.5070300@KennedySoftware.ie>
References: <4D94C282.1090903@KennedySoftware.ie>
	<4D9985E5.2060102@chaschperli.ch>
	<4D999FB3.5070300@KennedySoftware.ie>
Message-ID: <BANLkTin_rQFWV-D56GzxV4dXdvL=bJE+yA@mail.gmail.com>

I've had some decent success with a couple Unidata phones. My only qualms
were lack of transfer buttons, which you can get around by using specified
star codes.
http://www.udcsystems.com/product/sq3000.php
I believe the Snom M3 has WiFi support as well, which might be a great
option if these clients are of higher importance.

<http://www.udcsystems.com/product/sq3000.php>As for smartphone sip clients,
I've a strong love for Bria on the iPhone (aside from the battery kill while
letting it background).
cSipSimple on the Android market is very nice, integrates with the OS and
native phone system quite well without such a dramatic hit to the battery as
far as I can tell.


On Mon, Apr 4, 2011 at 3:38 AM, Michael Kennedy <Info at kennedysoftware.ie>wrote:

> > If your clients have "smart"-phones with WLAN :
> >
> > http://www.counterpath.com/bria-android-edition.html (offers G.729,
> > haven't used it)
> >
> > http://www.sipdroid.com/ (using this for myself with FS)
>
> These are most interesting links/products, Thomas. Thank you very much.
>
> I knew nothing about either, and have spent the past 2 hours reading up
> about them - and have "only started"!
>
> Very many thanks.
>   - 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
>
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From wstephen80 at gmail.com  Mon Apr  4 20:49:16 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 4 Apr 2011 18:49:16 +0200
Subject: [Freeswitch-users] My tone_detect doesn't works
Message-ID: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>

I'm trying to do a tone_detect in a bridged session but when the tone is
detected, the specified action is not performed.

My dialplan is:

<action application="set" data="execute_on_media=tone_detect mytone 820 wo
+30000 set mytone=true 2" />
<action application="bridge" data="${dialstring}" />

In the log I see the following rows:

2011-04-04 18:34:24.604908 [DEBUG] mod_dptools.c:1059
sofia/external/xxx at yyySET [execute_on_media]=[tone_detect mytone 820
wo +30000 set mytone=true 2]

EXECUTE sofia/external/xxx at yyy tone_detect(mytone 820 wo +30000 set
mytone=true 2)

2011-04-04 18:34:24.647971 [NOTICE] mod_dptools.c:1591 Enabling tone
detection 'mytone' '820'

2011-04-04 18:34:28.163751 [DEBUG] switch_ivr_async.c:2475 TONE mytone HIT
1/2

2011-04-04 18:34:28.759034 [DEBUG] switch_ivr_async.c:2475 TONE mytone HIT
2/2

2011-04-04 18:34:28.759034 [DEBUG] switch_ivr_async.c:2481 TONE mytone
DETECTED

but the "set mytone=true" is never executed.

Any suggestion?

Stephen
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From anthony.minessale at gmail.com  Mon Apr  4 21:36:00 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Apr 2011 12:36:00 -0500
Subject: [Freeswitch-users] xml_curl response for voicemail_inject
In-Reply-To: <BANLkTi=AwsfhJw7j6yLBBmy7hAUwVegkbA@mail.gmail.com>
References: <BANLkTi=AwsfhJw7j6yLBBmy7hAUwVegkbA@mail.gmail.com>
Message-ID: <BANLkTikvfBZEYeA=UsoKRhXOJdrnMVgJGQ@mail.gmail.com>

Its just looking for the user record so I can get the params and
variables from it.


On Mon, Apr 4, 2011 at 10:53 AM, Frank Park <frank at telonium.com> wrote:
> I had a quick question.
> I am trying to figure out what directory response is expected when a user
> wants to forward a voicemail to another extension.
> Looking at the post request of xml_curl, I see that it's invoking
> voicemail_inject. I've tried the same response as other voicemail function,
> which is similar to the authorization response, but that didn't seem to do
> it. Anybody care to share an example response when voicemail_inject is
> requested?
> Thanks,
> Frank
>
> --
>
> ----=======================----
> Frank Park
> Telonium Communications, LLC
> frank at telonium.com
> http://www.telonium.com
> Follow Us on Twitter: @GetTelonium
> 404-566-8888 x1001 Office
> 404-939-4242 Cell
> ----=======================----
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From frank at telonium.com  Mon Apr  4 21:52:57 2011
From: frank at telonium.com (Frank Park)
Date: Mon, 4 Apr 2011 13:52:57 -0400
Subject: [Freeswitch-users] xml_curl response for voicemail_inject
In-Reply-To: <BANLkTikvfBZEYeA=UsoKRhXOJdrnMVgJGQ@mail.gmail.com>
References: <BANLkTi=AwsfhJw7j6yLBBmy7hAUwVegkbA@mail.gmail.com>
	<BANLkTikvfBZEYeA=UsoKRhXOJdrnMVgJGQ@mail.gmail.com>
Message-ID: <BANLkTimCsmMSagdMQBPhv=DQo0_ChVNMbA@mail.gmail.com>

Yeah.. the current response to voicemail_inject is identical to any
directory lookup, which looks something like this:

...
      <user id="1001">
        <params>
          <param name="a1-hash" value="xxxx"/>
          <param name="vm-a1-hash" value="xxxx"/>
          <param name="inbound-bypass-media" value="true"/>
          <param name="vm-email-all-messages" value="true"/>
          <param name="vm-attach-file" value="true"/>
          <param name="vm-mailto" value="xxxxxx"/>
          <param name="vm-keep-local-after-email" value="true"/>
        </params>
        <variables>
          <variable name="toll_allow" value="domestic"/>
          <variable name="accountcode" value="xxxx"/>
          <variable name="effective_caller_id_name" value="xxxxx"/>
          <variable name="effective_caller_id_number" value="xxxx"/>
          <variable name="user_context" value="xxxx"/>
          <variable name="domain_name" value="xxxxx"/>
        </variables>
...

Shouldn't this be enough?

Frank






On Mon, Apr 4, 2011 at 1:36 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Its just looking for the user record so I can get the params and
> variables from it.
>
>
> On Mon, Apr 4, 2011 at 10:53 AM, Frank Park <frank at telonium.com> wrote:
> > I had a quick question.
> > I am trying to figure out what directory response is expected when a user
> > wants to forward a voicemail to another extension.
> > Looking at the post request of xml_curl, I see that it's invoking
> > voicemail_inject. I've tried the same response as other voicemail
> function,
> > which is similar to the authorization response, but that didn't seem to
> do
> > it. Anybody care to share an example response when voicemail_inject is
> > requested?
> > Thanks,
> > Frank
> >
> > --
> >
> > ----=======================----
> > Frank Park
> > Telonium Communications, LLC
> > frank at telonium.com
> > http://www.telonium.com
> > Follow Us on Twitter: @GetTelonium
> > 404-566-8888 x1001 Office
> > 404-939-4242 Cell
> > ----=======================----
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> 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
>



-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From msc at freeswitch.org  Mon Apr  4 22:40:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Apr 2011 11:40:37 -0700
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
Message-ID: <BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>

On Mon, Apr 4, 2011 at 9:49 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> I'm trying to do a tone_detect in a bridged session but when the tone is
> detected, the specified action is not performed.
>
> My dialplan is:
>
> <action application="set" data="execute_on_media=tone_detect mytone 820 wo
> +30000 set mytone=true 2" />
> <action application="bridge" data="${dialstring}" />
>
> In the log I see the following rows:
>
> 2011-04-04 18:34:24.604908 [DEBUG] mod_dptools.c:1059
> sofia/external/xxx at yyy SET [execute_on_media]=[tone_detect mytone 820 wo
> +30000 set mytone=true 2]
>
> EXECUTE sofia/external/xxx at yyy tone_detect(mytone 820 wo +30000 set
> mytone=true 2)
>
> 2011-04-04 18:34:24.647971 [NOTICE] mod_dptools.c:1591 Enabling tone
> detection 'mytone' '820'
>
> 2011-04-04 18:34:28.163751 [DEBUG] switch_ivr_async.c:2475 TONE mytone HIT
> 1/2
>
> 2011-04-04 18:34:28.759034 [DEBUG] switch_ivr_async.c:2475 TONE mytone HIT
> 2/2
>
> 2011-04-04 18:34:28.759034 [DEBUG] switch_ivr_async.c:2481 TONE mytone
> DETECTED
>
> but the "set mytone=true" is never executed.
>
> Any suggestion?
>

Where are you looking to see if "mytone" is set to 'true'? Be sure that you
are looking on the A leg and not the B leg...
-MC
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From lon at kickasspixels.com  Mon Apr  4 22:43:09 2011
From: lon at kickasspixels.com (Lon Baker)
Date: Mon, 4 Apr 2011 11:43:09 -0700
Subject: [Freeswitch-users] Transcoding with Manual Redirect
Message-ID: <ED257D518489452BAA55D8EA69FF897A@kickasspixels.com>

I'm trying to force transcoding from PCMU/A to G722 or fallback. I have it working through a normal bridge dialplan. 

Another scenario I'm working on is when I receive a call, bridge it to another server which issues a redirect. Using the manual redirect settings and dialplan context, the following is not working.

<context name="redirected">
<extension name="redirected">
<condition field="${sip_redirect_dialstring}" expression="^sofia/internal/sip:(.*)$">
<action application="export" data="nolocal:absolute_codec_string=G722,PCMU,PCMA"/> 
<action application="bridge" data="sofia/external/$1" />
</condition>
</extension>
</context>


It appears to be losing the absolute codec string.

Any ideas?

-- 
Lon Baker
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From wstephen80 at gmail.com  Mon Apr  4 23:12:40 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 4 Apr 2011 21:12:40 +0200
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
	<BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
Message-ID: <BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>

Thank you Michael, yes, I expect that the variable will be in legA.

Stephen


On Mon, Apr 4, 2011 at 8:40 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Apr 4, 2011 at 9:49 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> I'm trying to do a tone_detect in a bridged session but when the tone is
>> detected, the specified action is not performed.
>>
>> My dialplan is:
>>
>> <action application="set" data="execute_on_media=tone_detect mytone 820 wo
>> +30000 set mytone=true 2" />
>> <action application="bridge" data="${dialstring}" />
>>
>> In the log I see the following rows:
>>
>> 2011-04-04 18:34:24.604908 [DEBUG] mod_dptools.c:1059
>> sofia/external/xxx at yyy SET [execute_on_media]=[tone_detect mytone 820 wo
>> +30000 set mytone=true 2]
>>
>> EXECUTE sofia/external/xxx at yyy tone_detect(mytone 820 wo +30000 set
>> mytone=true 2)
>>
>> 2011-04-04 18:34:24.647971 [NOTICE] mod_dptools.c:1591 Enabling tone
>> detection 'mytone' '820'
>>
>> 2011-04-04 18:34:28.163751 [DEBUG] switch_ivr_async.c:2475 TONE mytone HIT
>> 1/2
>>
>> 2011-04-04 18:34:28.759034 [DEBUG] switch_ivr_async.c:2475 TONE mytone HIT
>> 2/2
>>
>> 2011-04-04 18:34:28.759034 [DEBUG] switch_ivr_async.c:2481 TONE mytone
>> DETECTED
>>
>> but the "set mytone=true" is never executed.
>>
>> Any suggestion?
>>
>
> Where are you looking to see if "mytone" is set to 'true'? Be sure that you
> are looking on the A leg and not the B leg...
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Apr  4 23:23:09 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Apr 2011 12:23:09 -0700
Subject: [Freeswitch-users] uuid_hold is not send hold message
In-Reply-To: <8slt5a$angvq4@irxd5-187.sinamail.sina.com.cn>
References: <8slt5a$angvq4@irxd5-187.sinamail.sina.com.cn>
Message-ID: <BANLkTimVHcm+0oe4Ekhxq8hvx7V_39vQ5g@mail.gmail.com>

I just tried this on latest git and it worked fine for me. Can you pastebin
the console debug output when you use it?
-MC

2011/4/2 ?? <sc_zhangming at sina.com>

> freeswitch-users????
>
>        uuid_hold command , freeswitch is not send HOLD  message. who know
> it.
>
> ?
> ??
>
>
> ??
> sc_zhangming at sina.com
> 2011-04-02
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Apr  4 23:26:35 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Apr 2011 12:26:35 -0700
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
	<BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
	<BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>
Message-ID: <BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com>

On Mon, Apr 4, 2011 at 12:12 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> Thank you Michael, yes, I expect that the variable will be in legA.
>
> Stephen
>

Try putting the app argument in single quotes:
<action application="set" data="execute_on_media=tone_detect mytone 820 wo
+30000 set 'mytone=true' 2" />

-MC
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From lists at telefaks.de  Mon Apr  4 23:58:27 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Mon, 04 Apr 2011 21:58:27 +0200
Subject: [Freeswitch-users] Dingaling and sasl authentication failed
In-Reply-To: <BANLkTimEczTy50PV2V2QZmuwk8GrZpP0Zg@mail.gmail.com>
References: <4D9127CD.7060300@telefaks.de>	<1301743757340-6233670.post@n2.nabble.com>
	<BANLkTimEczTy50PV2V2QZmuwk8GrZpP0Zg@mail.gmail.com>
Message-ID: <4D9A22E3.5070803@telefaks.de>

Hello Anthony,

that did the trick. In Germany I now use "talk.google.com" for the
server and "googlemail.com" for the login.

Thanks

Best regards
Peter

Anthony Minessale schrieb:
> The iksemel lib we use does not have support for srv records.  So if
> the auth is really done to some remote server, you will have to
> specify it manually in the server option.  See the default for gmail,
> googlemail (the euro version may have a different alternate server"
>
> Try doing a naptr or srv lookup on it.
>
>
> On Sat, Apr 2, 2011 at 6:29 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>   
>> I don't know if this will help or not. But, so far the only dingaling error
>> messages found in /var/log/freeswitch/freeswitch.log file on my FS (running
>> on FreeSWITCH Version 1.0.head (git-9795dd2 2011-03-26 11-07-34 -0500)) is
>> shown below:
>> 2011-03-31 13:22:30.718490 [DEBUG] libdingaling.c:1610 io error 2 7 retry in
>> 3 second(s)
>> 2011-03-31 13:22:34.171096 [DEBUG] libdingaling.c:1297 XMPP server connected
>> 2011-03-31 13:22:34.307809 [DEBUG] libdingaling.c:1309 XMPP authenticated
>>
>>
>> -----
>> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
>> --
>> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Dingaling-and-sasl-authentication-failed-tp6217329p6233670.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
>>
>>     
>
>
>
>   


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de


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From wstephen80 at gmail.com  Tue Apr  5 00:07:57 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 4 Apr 2011 22:07:57 +0200
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
	<BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
	<BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>
	<BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com>
Message-ID: <BANLkTikpyu3kNa9cUue95_Krp4i7E19tWA@mail.gmail.com>

I have tried with single quotas: no change, the tone is detected but the set
is not executed.

stephen

On Mon, Apr 4, 2011 at 9:26 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Apr 4, 2011 at 12:12 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> Thank you Michael, yes, I expect that the variable will be in legA.
>>
>> Stephen
>>
>
> Try putting the app argument in single quotes:
> <action application="set" data="execute_on_media=tone_detect mytone 820 wo
> +30000 set 'mytone=true' 2" />
>
> -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
>
>
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From msc at freeswitch.org  Tue Apr  5 00:12:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Apr 2011 13:12:38 -0700
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTikpyu3kNa9cUue95_Krp4i7E19tWA@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
	<BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
	<BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>
	<BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com>
	<BANLkTikpyu3kNa9cUue95_Krp4i7E19tWA@mail.gmail.com>
Message-ID: <BANLkTikZ5avXAHzB6CzR2Fc06e6q4wTzUQ@mail.gmail.com>

Try doing an execute_extension or some other app, just to see if you can
narrow down where the issue is.
-MC

On Mon, Apr 4, 2011 at 1:07 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> I have tried with single quotas: no change, the tone is detected but the
> set is not executed.
>
> stephen
>
> On Mon, Apr 4, 2011 at 9:26 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Apr 4, 2011 at 12:12 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> Thank you Michael, yes, I expect that the variable will be in legA.
>>>
>>> Stephen
>>>
>>
>> Try putting the app argument in single quotes:
>> <action application="set" data="execute_on_media=tone_detect mytone 820 wo
>> +30000 set 'mytone=true' 2" />
>>
>> -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
>
>
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From wstephen80 at gmail.com  Tue Apr  5 00:35:31 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 4 Apr 2011 22:35:31 +0200
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTikZ5avXAHzB6CzR2Fc06e6q4wTzUQ@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
	<BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
	<BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>
	<BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com>
	<BANLkTikpyu3kNa9cUue95_Krp4i7E19tWA@mail.gmail.com>
	<BANLkTikZ5avXAHzB6CzR2Fc06e6q4wTzUQ@mail.gmail.com>
Message-ID: <BANLkTika9PtUpPa6BRyALQWm1dsG4F6Ruw@mail.gmail.com>

Ok, I have first tried with:

<action application="set" data="execute_on_media=tone_detect mytone 820 wo
+30000 execute_extension 'myextension XML myextension' 2" />

and myextension is NOT executed after tone detection.

Stephen



On Mon, Apr 4, 2011 at 10:12 PM, Michael Collins <msc at freeswitch.org> wrote:

> Try doing an execute_extension or some other app, just to see if you can
> narrow down where the issue is.
> -MC
>
>
> On Mon, Apr 4, 2011 at 1:07 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> I have tried with single quotas: no change, the tone is detected but the
>> set is not executed.
>>
>> stephen
>>
>> On Mon, Apr 4, 2011 at 9:26 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Mon, Apr 4, 2011 at 12:12 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>
>>>> Thank you Michael, yes, I expect that the variable will be in legA.
>>>>
>>>> Stephen
>>>>
>>>
>>> Try putting the app argument in single quotes:
>>> <action application="set" data="execute_on_media=tone_detect mytone 820
>>> wo +30000 set 'mytone=true' 2" />
>>>
>>> -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
>
>
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From msc at freeswitch.org  Tue Apr  5 01:21:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Apr 2011 14:21:13 -0700
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTika9PtUpPa6BRyALQWm1dsG4F6Ruw@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
	<BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
	<BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>
	<BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com>
	<BANLkTikpyu3kNa9cUue95_Krp4i7E19tWA@mail.gmail.com>
	<BANLkTikZ5avXAHzB6CzR2Fc06e6q4wTzUQ@mail.gmail.com>
	<BANLkTika9PtUpPa6BRyALQWm1dsG4F6Ruw@mail.gmail.com>
Message-ID: <BANLkTimjzTzuomK-RbsBXZahkJKvb-DU6g@mail.gmail.com>

what version of FS and what OS?
-MC

On Mon, Apr 4, 2011 at 1:35 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> Ok, I have first tried with:
>
> <action application="set" data="execute_on_media=tone_detect mytone 820 wo
> +30000 execute_extension 'myextension XML myextension' 2" />
>
> and myextension is NOT executed after tone detection.
>
> Stephen
>
>
>
> On Mon, Apr 4, 2011 at 10:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Try doing an execute_extension or some other app, just to see if you can
>> narrow down where the issue is.
>> -MC
>>
>>
>> On Mon, Apr 4, 2011 at 1:07 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> I have tried with single quotas: no change, the tone is detected but the
>>> set is not executed.
>>>
>>> stephen
>>>
>>> On Mon, Apr 4, 2011 at 9:26 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>>
>>>>
>>>> On Mon, Apr 4, 2011 at 12:12 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>
>>>>> Thank you Michael, yes, I expect that the variable will be in legA.
>>>>>
>>>>> Stephen
>>>>>
>>>>
>>>> Try putting the app argument in single quotes:
>>>> <action application="set" data="execute_on_media=tone_detect mytone 820
>>>> wo +30000 set 'mytone=true' 2" />
>>>>
>>>> -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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Apr  5 01:26:38 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 4 Apr 2011 22:26:38 +0100
Subject: [Freeswitch-users] Transcoding with Manual Redirect
In-Reply-To: <ED257D518489452BAA55D8EA69FF897A@kickasspixels.com>
References: <ED257D518489452BAA55D8EA69FF897A@kickasspixels.com>
Message-ID: <BANLkTikSyoRQz1WLLy5bbqbLN2c0ZwZtrg@mail.gmail.com>

Siptrace?

On 4 April 2011 19:43, Lon Baker <lon at kickasspixels.com> wrote:

>  I'm trying to force transcoding from PCMU/A to G722 or fallback. I have
> it working through a normal bridge dialplan.
>
> Another scenario I'm working on is when I receive a call, bridge it to
> another server which issues a redirect. Using the manual redirect settings
> and dialplan context, the following is not working.
>
> <context name="redirected">
>  <extension name="redirected">
>   <condition field="${sip_redirect_dialstring}"
> expression="^sofia/internal/sip:(.*)$">
>  <action application="export"
> data="nolocal:absolute_codec_string=G722,PCMU,PCMA"/>
>    <action application="bridge" data="sofia/external/$1" />
>   </condition>
>  </extension>
> </context>
>
> It appears to be losing the absolute codec string.
>
> Any ideas?
>
> --
> Lon Baker
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From wstephen80 at gmail.com  Tue Apr  5 01:29:44 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Mon, 4 Apr 2011 23:29:44 +0200
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTimjzTzuomK-RbsBXZahkJKvb-DU6g@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
	<BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
	<BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>
	<BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com>
	<BANLkTikpyu3kNa9cUue95_Krp4i7E19tWA@mail.gmail.com>
	<BANLkTikZ5avXAHzB6CzR2Fc06e6q4wTzUQ@mail.gmail.com>
	<BANLkTika9PtUpPa6BRyALQWm1dsG4F6Ruw@mail.gmail.com>
	<BANLkTimjzTzuomK-RbsBXZahkJKvb-DU6g@mail.gmail.com>
Message-ID: <BANLkTi=Se6k5Ly8f_NpD8s-yfZjE66poBw@mail.gmail.com>

FS commit 57b6255b17e8934a99f07c7467fb3ceaf822a5b4, Tue Mar 22 15:15:09 2011
-0500

CentOS 5.5 64bit



On Mon, Apr 4, 2011 at 11:21 PM, Michael Collins <msc at freeswitch.org> wrote:

> what version of FS and what OS?
> -MC
>
>
> On Mon, Apr 4, 2011 at 1:35 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> Ok, I have first tried with:
>>
>> <action application="set" data="execute_on_media=tone_detect mytone 820 wo
>> +30000 execute_extension 'myextension XML myextension' 2" />
>>
>> and myextension is NOT executed after tone detection.
>>
>> Stephen
>>
>>
>>
>> On Mon, Apr 4, 2011 at 10:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Try doing an execute_extension or some other app, just to see if you can
>>> narrow down where the issue is.
>>> -MC
>>>
>>>
>>> On Mon, Apr 4, 2011 at 1:07 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>
>>>> I have tried with single quotas: no change, the tone is detected but the
>>>> set is not executed.
>>>>
>>>> stephen
>>>>
>>>> On Mon, Apr 4, 2011 at 9:26 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>>
>>>>>
>>>>> On Mon, Apr 4, 2011 at 12:12 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>>
>>>>>> Thank you Michael, yes, I expect that the variable will be in legA.
>>>>>>
>>>>>> Stephen
>>>>>>
>>>>>
>>>>> Try putting the app argument in single quotes:
>>>>> <action application="set" data="execute_on_media=tone_detect mytone 820
>>>>> wo +30000 set 'mytone=true' 2" />
>>>>>
>>>>> -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
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Apr  5 01:33:34 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Apr 2011 14:33:34 -0700
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTi=Se6k5Ly8f_NpD8s-yfZjE66poBw@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
	<BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
	<BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>
	<BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com>
	<BANLkTikpyu3kNa9cUue95_Krp4i7E19tWA@mail.gmail.com>
	<BANLkTikZ5avXAHzB6CzR2Fc06e6q4wTzUQ@mail.gmail.com>
	<BANLkTika9PtUpPa6BRyALQWm1dsG4F6Ruw@mail.gmail.com>
	<BANLkTimjzTzuomK-RbsBXZahkJKvb-DU6g@mail.gmail.com>
	<BANLkTi=Se6k5Ly8f_NpD8s-yfZjE66poBw@mail.gmail.com>
Message-ID: <BANLkTimLCOgsk_XHCdJ_uTONjRtnmooMQA@mail.gmail.com>

Pastebin the whole extension and the entire debug output of the call. Also,
pastebin the a-leg xml cdr record.

-MC

On Mon, Apr 4, 2011 at 2:29 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> FS commit 57b6255b17e8934a99f07c7467fb3ceaf822a5b4, Tue Mar 22 15:15:09
> 2011 -0500
>
> CentOS 5.5 64bit
>
>
>
> On Mon, Apr 4, 2011 at 11:21 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> what version of FS and what OS?
>> -MC
>>
>>
>> On Mon, Apr 4, 2011 at 1:35 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>
>>> Ok, I have first tried with:
>>>
>>> <action application="set" data="execute_on_media=tone_detect mytone 820
>>> wo +30000 execute_extension 'myextension XML myextension' 2" />
>>>
>>> and myextension is NOT executed after tone detection.
>>>
>>> Stephen
>>>
>>>
>>>
>>> On Mon, Apr 4, 2011 at 10:12 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Try doing an execute_extension or some other app, just to see if you can
>>>> narrow down where the issue is.
>>>> -MC
>>>>
>>>>
>>>> On Mon, Apr 4, 2011 at 1:07 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>
>>>>> I have tried with single quotas: no change, the tone is detected but
>>>>> the set is not executed.
>>>>>
>>>>> stephen
>>>>>
>>>>> On Mon, Apr 4, 2011 at 9:26 PM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> On Mon, Apr 4, 2011 at 12:12 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>>>>>>
>>>>>>> Thank you Michael, yes, I expect that the variable will be in legA.
>>>>>>>
>>>>>>> Stephen
>>>>>>>
>>>>>>
>>>>>> Try putting the app argument in single quotes:
>>>>>> <action application="set" data="execute_on_media=tone_detect mytone
>>>>>> 820 wo +30000 set 'mytone=true' 2" />
>>>>>>
>>>>>> -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
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Tue Apr  5 01:53:18 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 4 Apr 2011 17:53:18 -0400
Subject: [Freeswitch-users] My tone_detect doesn't works
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com><BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com><BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com><BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com><BANLkTikpyu3kNa9cUue95_Krp4i7E19tWA@mail.gmail.com><BANLkTikZ5avXAHzB6CzR2Fc06e6q4wTzUQ@mail.gmail.com><BANLkTika9PtUpPa6BRyALQWm1dsG4F6Ruw@mail.gmail.com>
	<BANLkTimjzTzuomK-RbsBXZahkJKvb-DU6g@mail.gmail.com>
Message-ID: <DB8FD56D6C0C4FFA982CC379F2DF8DCC@e1705>

Suggestion for FS team :
why not integrate an automatic core dump/ log or whatever
in FS like other software that sends to your email all what you need ?
at the FS instal a kind of "please provide your email in case of bug/crash to send log automatically to our server thank you"....
it's only ann idea....
  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Monday, April 04, 2011 5:21 PM
  Subject: Re: [Freeswitch-users] My tone_detect doesn't works


  what version of FS and what OS?
  -MC


  On Mon, Apr 4, 2011 at 1:35 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

    Ok, I have first tried with:


    <action application="set" data="execute_on_media=tone_detect mytone 820 wo +30000 execute_extension 'myextension XML myextension' 2" />


    and myextension is NOT executed after tone detection.


    Stephen





    On Mon, Apr 4, 2011 at 10:12 PM, Michael Collins <msc at freeswitch.org> wrote:

      Try doing an execute_extension or some other app, just to see if you can narrow down where the issue is.
      -MC



      On Mon, Apr 4, 2011 at 1:07 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

        I have tried with single quotas: no change, the tone is detected but the set is not executed.


        stephen


        On Mon, Apr 4, 2011 at 9:26 PM, Michael Collins <msc at freeswitch.org> wrote:




          On Mon, Apr 4, 2011 at 12:12 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

            Thank you Michael, yes, I expect that the variable will be in legA.


            Stephen


          Try putting the app argument in single quotes:
          <action application="set" data="execute_on_media=tone_detect mytone 820 wo +30000 set 'mytone=true' 2" />


          -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





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






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


  _______________________________________________
  FreeSWITCH-users mailing list
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  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mario_fs at mgtech.com  Tue Apr  5 01:57:48 2011
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 4 Apr 2011 14:57:48 -0700
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <4D999FB3.5070300@KennedySoftware.ie>
References: <4D94C282.1090903@KennedySoftware.ie>
	<4D9985E5.2060102@chaschperli.ch>
	<4D999FB3.5070300@KennedySoftware.ie>
Message-ID: <F5586FB8-7CC2-4EDE-B4BB-2C8824CA218D@mgtech.com>

And if they are using iPhones we have been very happy with Acrobits Softwphone and Groundwite from http://www.acrobits.cz/27/acrobits-mobile-voip-solutions, also available for android now.

On Apr 4, 2011, at 3:38 AM, Michael Kennedy wrote:

>> If your clients have "smart"-phones with WLAN :
>> 
>> http://www.counterpath.com/bria-android-edition.html (offers G.729,
>> haven't used it)
>> 
>> http://www.sipdroid.com/ (using this for myself with FS)
> 
> These are most interesting links/products, Thomas. Thank you very much.
> 
> I knew nothing about either, and have spent the past 2 hours reading up 
> about them - and have "only started"!
> 
> Very many thanks.
>   - 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 mario_fs at mgtech.com  Tue Apr  5 02:00:36 2011
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 4 Apr 2011 15:00:36 -0700
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <4D9985E5.2060102@chaschperli.ch>
References: <4D94C282.1090903@KennedySoftware.ie>
	<4D9985E5.2060102@chaschperli.ch>
Message-ID: <C503C55F-C746-4F73-AEA8-DD7512D2C799@mgtech.com>

We have the SPA962s and if they are the same as they 525... searches stink! No way to lookup number by alpha. Also, paging function only activates if it finds a SPA9000 which we replaced with FreeSwitch. YMMV.

On Apr 4, 2011, at 1:48 AM, Thomas Mueller wrote:

> 
>> I expected that many phone suppliers would have handsets with EITHER
>> RJ45 or WiFi connectivity to the LAN, or even both! I've found only a
>> single device, a Cisco SPA525G2! Furthermore, searching the FS site, and
>> various VoIP sites, and running general searches, I've found no other
>> SIP WiFi phones that look like standard desktop handsets.
>> 
>> I'd appreciate any pointers to WiFi devices that are recommended with
>> FS. Preferably "standard-looking" desktop units, and better still, if
>> they had wired "sisters" - in appearance and functionality!
> 
> If your clients have "smart"-phones with WLAN :
> 
> http://www.counterpath.com/bria-android-edition.html (offers G.729, 
> haven't used it)
> 
> http://www.sipdroid.com/ (using this for myself with FS)
> 
> For sure there are more solutions out there.
> 
> - Thomas
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mario_fs at mgtech.com  Tue Apr  5 02:01:54 2011
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 4 Apr 2011 15:01:54 -0700
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <1301712035.18009.1048.camel@macmini>
References: <4D94C282.1090903@KennedySoftware.ie>
	<4D951D45.5010005@utoronto.ca>
	<4D95E839.2070403@KennedySoftware.ie>
	<1301712035.18009.1048.camel@macmini>
Message-ID: <2369DA46-4A45-431B-98F2-4122295C8776@mgtech.com>

I was planning on replacing our Cisco SPA962s with snom 870s. Could you please tell me why they are diabolical? Thanks.

On Apr 1, 2011, at 7:40 PM, Campbell Steven wrote:

> The Snom 870 will do it with a USB Wifi dongle, but in my experience don't go there, they are a diabolical handset from a usability standpoint.
> 
> Campbell
> 
> On Fri, 2011-04-01 at 15:59 +0100, Michael Kennedy wrote:
>> 
>> Victor,
>> 
>> > A bit off-topic but here are my 50 cents:
>> 
>> Oopppssss, my apologies - I thought it might be a common query for folks 
>> thinking about FS - but maybe in another "list"?
>> 
>> > -Did you consider building a wireless bridge with a $40 WiFi router
>> > running DD-WRT/Tomato/OpenWRT etc?
>> 
>> I did NOT - and I've deployed a lot of them to support "PC"s! THANK YOU!
>> 
>> > This way you can plug wired phones into LAN ports of the "bridge" and
>> > the router will bridge them to your main access point.
>> > Asus WL-520GU will work and is really cheap.
>> 
>> EXCELLENT suggestion!
>> 
>> (Maybe I'm drifting even more O-T, but... I'm also glad you did not 
>> mention WiFi devices from Linksys - in my experience, some of these 
>> boxes performed very poorly, but I seem to be the only one on the planet 
>> with these experiences!).
>> 
>> > -If you go with WiFi you should only use WPA or WPA2.
>> > Less secure options (WEP :-) ) make all conversations accessible to public.
>> 
>> Yes, I think all APs are currently running on WPA2.
>> 
>> Thank you VERY much, Victor!
>>    - 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Tue Apr  5 02:16:15 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Apr 2011 17:16:15 -0500
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTi=Se6k5Ly8f_NpD8s-yfZjE66poBw@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
	<BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
	<BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>
	<BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com>
	<BANLkTikpyu3kNa9cUue95_Krp4i7E19tWA@mail.gmail.com>
	<BANLkTikZ5avXAHzB6CzR2Fc06e6q4wTzUQ@mail.gmail.com>
	<BANLkTika9PtUpPa6BRyALQWm1dsG4F6Ruw@mail.gmail.com>
	<BANLkTimjzTzuomK-RbsBXZahkJKvb-DU6g@mail.gmail.com>
	<BANLkTi=Se6k5Ly8f_NpD8s-yfZjE66poBw@mail.gmail.com>
Message-ID: <BANLkTi=0kASBQM7gub7T6NG4kiNQwfXtCg@mail.gmail.com>

Please try latest git.

As I say about 100 times a week.  Please don't report issues on the
mailing list.
I almost missed this thread.  By forcing me to scan the mailing list
for bug reports, I do more work and run more of a risk of missing
things.



On Mon, Apr 4, 2011 at 4:29 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> FS?commit 57b6255b17e8934a99f07c7467fb3ceaf822a5b4,?Tue Mar 22 15:15:09 2011
> -0500
> CentOS 5.5 64bit
>
>
>
> On Mon, Apr 4, 2011 at 11:21 PM, Michael Collins <msc at freeswitch.org> wrote:
>>
>> what version of FS and what OS?
>> -MC
>>
>> On Mon, Apr 4, 2011 at 1:35 PM, Stephen Wilde <wstephen80 at gmail.com>
>> wrote:
>>>
>>> Ok, I have first tried with:
>>> <action application="set" data="execute_on_media=tone_detect mytone 820
>>> wo +30000 execute_extension 'myextension XML myextension' 2" />
>>> and myextension is NOT executed after tone detection.
>>> Stephen
>>>
>>>
>>> On Mon, Apr 4, 2011 at 10:12 PM, Michael Collins <msc at freeswitch.org>
>>> wrote:
>>>>
>>>> Try doing an execute_extension or some other app, just to see if you can
>>>> narrow down where the issue is.
>>>> -MC
>>>>
>>>> On Mon, Apr 4, 2011 at 1:07 PM, Stephen Wilde <wstephen80 at gmail.com>
>>>> wrote:
>>>>>
>>>>> I have tried with single quotas: no change, the tone is detected but
>>>>> the set is not executed.
>>>>> stephen
>>>>>
>>>>> On Mon, Apr 4, 2011 at 9:26 PM, Michael Collins <msc at freeswitch.org>
>>>>> wrote:
>>>>>>
>>>>>>
>>>>>> On Mon, Apr 4, 2011 at 12:12 PM, Stephen Wilde <wstephen80 at gmail.com>
>>>>>> wrote:
>>>>>>>
>>>>>>> Thank you Michael, yes, I expect that the variable will be in legA.
>>>>>>> Stephen
>>>>>>
>>>>>> Try putting the app argument in single quotes:
>>>>>> <action application="set" data="execute_on_media=tone_detect mytone
>>>>>> 820 wo +30000 set 'mytone=true' 2" />
>>>>>> -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
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Tue Apr  5 02:17:46 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Apr 2011 17:17:46 -0500
Subject: [Freeswitch-users] Reminder: Do NOT post bugs HERE! post them to
	http://jira.freeswitch.org
Message-ID: <BANLkTikcf048NiiBwR+RPDH8BMUy-V-k6A@mail.gmail.com>

We do not have enough staff to look for bug reports in mailing lists
and it takes just as much time to do it on JIRA.


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From andrew.keil at askinteractive.net  Tue Apr  5 02:21:48 2011
From: andrew.keil at askinteractive.net (Andrew Keil)
Date: Tue, 5 Apr 2011 08:21:48 +1000
Subject: [Freeswitch-users] Demo Service (5000) Press 5 to listen screaming
 monkeys causes exception from latest Freeswitch build on Windows
Message-ID: <C42A22554C0CF44DAC0AA1F132B86DE65716763FCD@ASKSERVER.asksbs.local>

To Freeswitch developers,

FYI: I am running the latest Git HEAD build (downloaded today) on Windows XP SP3 with Visual C++ 2010 Express.

I am just running through the test services and found an issue with the Demo IVR service (5000) when I press 5 to listen screaming monkeys.  This causes an exception.

Exception: Unhandled exception at 0x0144d92c (mod_enum.dll) in Freeswitchconsole.exe: 0xC0000005: Access violation reading location 0xccccccd0.

Since this one of the main demo services I thought it best to report it.

Looking forward to your response.

Andrew Keil

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From msc at freeswitch.org  Tue Apr  5 02:25:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Apr 2011 15:25:10 -0700
Subject: [Freeswitch-users] Demo Service (5000) Press 5 to listen
 screaming monkeys causes exception from latest Freeswitch build on Windows
In-Reply-To: <C42A22554C0CF44DAC0AA1F132B86DE65716763FCD@ASKSERVER.asksbs.local>
References: <C42A22554C0CF44DAC0AA1F132B86DE65716763FCD@ASKSERVER.asksbs.local>
Message-ID: <BANLkTikEpmOMXN+j_9Boa-37aaG3c88DjQ@mail.gmail.com>

You must have clicked "send" before reading Tony's email. Please report this
to Jira.

Thanks,
MC

On Mon, Apr 4, 2011 at 3:21 PM, Andrew Keil
<andrew.keil at askinteractive.net>wrote:

> To Freeswitch developers,
>
>
>
> FYI: I am running the latest Git HEAD build (downloaded today) on Windows
> XP SP3 with Visual C++ 2010 Express.
>
>
>
> I am just running through the test services and found an issue with the
> Demo IVR service (5000) when I press 5 to listen screaming monkeys.  This
> causes an exception.
>
>
>
> Exception: Unhandled exception at 0x0144d92c (mod_enum.dll) in
> Freeswitchconsole.exe: 0xC0000005: Access violation reading location
> 0xccccccd0.
>
>
>
> Since this one of the main demo services I thought it best to report it.
>
>
>
> Looking forward to your response.
>
>
>
> Andrew Keil
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From andrew.keil at askinteractive.net  Tue Apr  5 02:43:32 2011
From: andrew.keil at askinteractive.net (Andrew Keil)
Date: Tue, 5 Apr 2011 08:43:32 +1000
Subject: [Freeswitch-users] Demo Service (5000) Press 5 to listen
 screaming monkeys causes exception from latest Freeswitch build on Windows
In-Reply-To: <BANLkTikEpmOMXN+j_9Boa-37aaG3c88DjQ@mail.gmail.com>
References: <C42A22554C0CF44DAC0AA1F132B86DE65716763FCD@ASKSERVER.asksbs.local>
	<BANLkTikEpmOMXN+j_9Boa-37aaG3c88DjQ@mail.gmail.com>
Message-ID: <C42A22554C0CF44DAC0AA1F132B86DE65716763FCE@ASKSERVER.asksbs.local>

Exactly what happened.  Reported to Jira just now.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, 5 April 2011 8:25 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Demo Service (5000) Press 5 to listen screaming monkeys causes exception from latest Freeswitch build on Windows

You must have clicked "send" before reading Tony's email. Please report this to Jira.

Thanks,
MC
On Mon, Apr 4, 2011 at 3:21 PM, Andrew Keil <andrew.keil at askinteractive.net<mailto:andrew.keil at askinteractive.net>> wrote:
To Freeswitch developers,

FYI: I am running the latest Git HEAD build (downloaded today) on Windows XP SP3 with Visual C++ 2010 Express.

I am just running through the test services and found an issue with the Demo IVR service (5000) when I press 5 to listen screaming monkeys.  This causes an exception.

Exception: Unhandled exception at 0x0144d92c (mod_enum.dll) in Freeswitchconsole.exe: 0xC0000005: Access violation reading location 0xccccccd0.

Since this one of the main demo services I thought it best to report it.

Looking forward to your response.

Andrew Keil


_______________________________________________
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



__________ Information from ESET NOD32 Antivirus, version of virus signature database 6015 (20110404) __________


The message was checked by ESET NOD32 Antivirus.


http://www.eset.com
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From msc at freeswitch.org  Tue Apr  5 02:45:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Apr 2011 15:45:41 -0700
Subject: [Freeswitch-users] Vestec Connector
In-Reply-To: <BANLkTi=Q5p04bDr6OC34ToQsrhSMPoG52g@mail.gmail.com>
References: <BANLkTi=Q5p04bDr6OC34ToQsrhSMPoG52g@mail.gmail.com>
Message-ID: <BANLkTimuVG1tf0CgJMcppDC9jjZT43DBNA@mail.gmail.com>

I don't believe so. Besides, the new version 2.x of Vestec doesn't even use
the connector (IIRC), it uses uniMRCP. I'd advise getting on a non-ancient
version of FreeSWITCH if you're going to use anything in production.

-MC

On Sun, Apr 3, 2011 at 7:08 PM, Roman Gelfand <rgelfand2 at gmail.com> wrote:

> Is the vestec connector compatible with freeswitch 1.05?
>
> 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
>
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From wstephen80 at gmail.com  Tue Apr  5 02:54:22 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 5 Apr 2011 00:54:22 +0200
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTi=0kASBQM7gub7T6NG4kiNQwfXtCg@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
	<BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
	<BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>
	<BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com>
	<BANLkTikpyu3kNa9cUue95_Krp4i7E19tWA@mail.gmail.com>
	<BANLkTikZ5avXAHzB6CzR2Fc06e6q4wTzUQ@mail.gmail.com>
	<BANLkTika9PtUpPa6BRyALQWm1dsG4F6Ruw@mail.gmail.com>
	<BANLkTimjzTzuomK-RbsBXZahkJKvb-DU6g@mail.gmail.com>
	<BANLkTi=Se6k5Ly8f_NpD8s-yfZjE66poBw@mail.gmail.com>
	<BANLkTi=0kASBQM7gub7T6NG4kiNQwfXtCg@mail.gmail.com>
Message-ID: <BANLkTi=mmBPoLLjjY2uBkEOAsBpYawezTQ@mail.gmail.com>

I have tried with latest git without success.

I have created an issue on jira: http://jira.freeswitch.org/browse/FS-3229

Stephen


On Tue, Apr 5, 2011 at 12:16 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Please try latest git.
>
> As I say about 100 times a week.  Please don't report issues on the
> mailing list.
> I almost missed this thread.  By forcing me to scan the mailing list
> for bug reports, I do more work and run more of a risk of missing
> things.
>
>
>
> On Mon, Apr 4, 2011 at 4:29 PM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > FS commit 57b6255b17e8934a99f07c7467fb3ceaf822a5b4, Tue Mar 22 15:15:09
> 2011
> > -0500
> > CentOS 5.5 64bit
> >
> >
> >
> > On Mon, Apr 4, 2011 at 11:21 PM, Michael Collins <msc at freeswitch.org>
> wrote:
> >>
> >> what version of FS and what OS?
> >> -MC
> >>
> >> On Mon, Apr 4, 2011 at 1:35 PM, Stephen Wilde <wstephen80 at gmail.com>
> >> wrote:
> >>>
> >>> Ok, I have first tried with:
> >>> <action application="set" data="execute_on_media=tone_detect mytone 820
> >>> wo +30000 execute_extension 'myextension XML myextension' 2" />
> >>> and myextension is NOT executed after tone detection.
> >>> Stephen
> >>>
> >>>
> >>> On Mon, Apr 4, 2011 at 10:12 PM, Michael Collins <msc at freeswitch.org>
> >>> wrote:
> >>>>
> >>>> Try doing an execute_extension or some other app, just to see if you
> can
> >>>> narrow down where the issue is.
> >>>> -MC
> >>>>
> >>>> On Mon, Apr 4, 2011 at 1:07 PM, Stephen Wilde <wstephen80 at gmail.com>
> >>>> wrote:
> >>>>>
> >>>>> I have tried with single quotas: no change, the tone is detected but
> >>>>> the set is not executed.
> >>>>> stephen
> >>>>>
> >>>>> On Mon, Apr 4, 2011 at 9:26 PM, Michael Collins <msc at freeswitch.org>
> >>>>> wrote:
> >>>>>>
> >>>>>>
> >>>>>> On Mon, Apr 4, 2011 at 12:12 PM, Stephen Wilde <
> wstephen80 at gmail.com>
> >>>>>> wrote:
> >>>>>>>
> >>>>>>> Thank you Michael, yes, I expect that the variable will be in legA.
> >>>>>>> Stephen
> >>>>>>
> >>>>>> Try putting the app argument in single quotes:
> >>>>>> <action application="set" data="execute_on_media=tone_detect mytone
> >>>>>> 820 wo +30000 set 'mytone=true' 2" />
> >>>>>> -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
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> 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
>
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From msc at freeswitch.org  Tue Apr  5 03:32:20 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Apr 2011 16:32:20 -0700
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTi=mmBPoLLjjY2uBkEOAsBpYawezTQ@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
	<BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
	<BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>
	<BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com>
	<BANLkTikpyu3kNa9cUue95_Krp4i7E19tWA@mail.gmail.com>
	<BANLkTikZ5avXAHzB6CzR2Fc06e6q4wTzUQ@mail.gmail.com>
	<BANLkTika9PtUpPa6BRyALQWm1dsG4F6Ruw@mail.gmail.com>
	<BANLkTimjzTzuomK-RbsBXZahkJKvb-DU6g@mail.gmail.com>
	<BANLkTi=Se6k5Ly8f_NpD8s-yfZjE66poBw@mail.gmail.com>
	<BANLkTi=0kASBQM7gub7T6NG4kiNQwfXtCg@mail.gmail.com>
	<BANLkTi=mmBPoLLjjY2uBkEOAsBpYawezTQ@mail.gmail.com>
Message-ID: <BANLkTimeRo3hcay+DmUG4OTsbP89wmxZdw@mail.gmail.com>

On Mon, Apr 4, 2011 at 3:54 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> I have tried with latest git without success.
>
> I have created an issue on jira: http://jira.freeswitch.org/browse/FS-3229
>
> Stephen
>
>
Thanks, we'll check it out.
-MC
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From devel at omninet.eu  Tue Apr  5 03:14:34 2011
From: devel at omninet.eu (Anestis Mavro)
Date: Tue, 5 Apr 2011 02:14:34 +0300
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <F5586FB8-7CC2-4EDE-B4BB-2C8824CA218D@mgtech.com>
References: <4D94C282.1090903@KennedySoftware.ie><4D9985E5.2060102@chaschperli.ch><4D999FB3.5070300@KennedySoftware.ie>
	<F5586FB8-7CC2-4EDE-B4BB-2C8824CA218D@mgtech.com>
Message-ID: <4CBE0BA4A3DA492299ACFA4C41EA2214@omni1.local>

I have a lot of customers with iPhones and installed Media5Fone
(http://www.media5corp.com/en/softphones/media5-fone-iphone). They use it
even on UMTS and GPRS EDGE (with g729) networks beside WiFi without big
problems. This solution works well, even with TLS and SRTP!

 

Anestis

 

 

 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mario G
Sent: Tuesday, April 05, 2011 12:58 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Recommended SIP IP Wifi Handsets?

 

And if they are using iPhones we have been very happy with Acrobits
Softwphone and Groundwite from
http://www.acrobits.cz/27/acrobits-mobile-voip-solutions, also available for
android now.

 

On Apr 4, 2011, at 3:38 AM, Michael Kennedy wrote:

 

>> If your clients have "smart"-phones with WLAN :

>> 

>> http://www.counterpath.com/bria-android-edition.html (offers G.729,

>> haven't used it)

>> 

>> http://www.sipdroid.com/ (using this for myself with FS)

> 

> These are most interesting links/products, Thomas. Thank you very much.

> 

> I knew nothing about either, and have spent the past 2 hours reading up 

> about them - and have "only started"!

> 

> Very many thanks.

>   - 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

 

 

_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at 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 5054 (20100423) __________

 

The message was checked by ESET NOD32 Antivirus.

 

http://www.eset.com

 

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From lloydie.t at gmail.com  Tue Apr  5 03:35:07 2011
From: lloydie.t at gmail.com (lloyd thomas)
Date: Tue, 5 Apr 2011 00:35:07 +0100
Subject: [Freeswitch-users] PRI Test Equipment
In-Reply-To: <BANLkTimtL=fzfQCjHYRQkvTQCoyJGyPHiw@mail.gmail.com>
References: <AANLkTi=txKH5Yj3MHaMrX1uHhnqMpuS=PuLgGsxpR=H8@mail.gmail.com>
	<AANLkTi=emvxczkxopd4+OYLdY7gJjyd7qgONO8sK-PKT@mail.gmail.com>
	<AANLkTinexeb64gsV_ih8fwub1F=K8V+fKEJtEPUeWuwQ@mail.gmail.com>
	<2E80FBC43F4F464AB3730923CA7CC754@dell9400>
	<AANLkTi=t418uxQL37eaD-HCmky=7UOCPsH0NyLbkgybQ@mail.gmail.com>
	<AANLkTi=MbzOfMbuT9ePvFPo6rw+JwxmR=HFXAQU6DiqR@mail.gmail.com>
	<BANLkTimtL=fzfQCjHYRQkvTQCoyJGyPHiw@mail.gmail.com>
Message-ID: <BANLkTim94+vgYBh6ZAz7zk=a0VtvdS5T2A@mail.gmail.com>

At least in the UK the Trend Aurora was the one most field engineers went
for. Mine is a bit old now,
but when I have an ISDN fault you find that most telco engineers wont argue
against it.
It also supports the different ISDN protocols we have in the UK including
Q.931, Q.sig, DASS2, DPNSS, etc
Try ebay
http://cgi.ebay.co.uk/TREND-COMMUNICATIONS-AURORA-DUET-/320677254927?pt=LH_DefaultDomain_3&hash=item4aa9da970f
Which is half what I paid for mine and I paid a quarter of if's value when I
got it, so bargin.

On 4 April 2011 05:20, Max Clark <max.clark at gmail.com> wrote:

> Thank you - that looks perfect.
>
> On Thu, Mar 31, 2011 at 10:48 AM, shouldbe q931 <shouldbeq931 at gmail.com>
> wrote:
> >
> >
> > On Thu, Mar 31, 2011 at 4:41 PM, Max Clark <max.clark at gmail.com> wrote:
> >>
> >> Thanks Jan I'll check this out.
> >>
> >
> > If you want to purchase a dedicated tester, the Trend Aurora is the one
> that
> > I used to
> > use
> http://www.trendcomms.com/web2/pages.nsf/vlCookie/global$aurora%20sonata?opendocument&cc=true
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anthony.minessale at gmail.com  Tue Apr  5 04:14:09 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Apr 2011 19:14:09 -0500
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTi=mmBPoLLjjY2uBkEOAsBpYawezTQ@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
	<BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
	<BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>
	<BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com>
	<BANLkTikpyu3kNa9cUue95_Krp4i7E19tWA@mail.gmail.com>
	<BANLkTikZ5avXAHzB6CzR2Fc06e6q4wTzUQ@mail.gmail.com>
	<BANLkTika9PtUpPa6BRyALQWm1dsG4F6Ruw@mail.gmail.com>
	<BANLkTimjzTzuomK-RbsBXZahkJKvb-DU6g@mail.gmail.com>
	<BANLkTi=Se6k5Ly8f_NpD8s-yfZjE66poBw@mail.gmail.com>
	<BANLkTi=0kASBQM7gub7T6NG4kiNQwfXtCg@mail.gmail.com>
	<BANLkTi=mmBPoLLjjY2uBkEOAsBpYawezTQ@mail.gmail.com>
Message-ID: <BANLkTikL68EiJ9S2f7YeD+ktYFpmfLKVMA@mail.gmail.com>

see the updated jira.

I am willing to bet you are causing this problem on yourself with a
test that is not sending any media to FS because it works for me.
Nevertheless, I have fixed it for good with some new syntax.

MC, can you update the wiki?


On Mon, Apr 4, 2011 at 5:54 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> I have tried with latest git without success.
> I have created an issue on jira: http://jira.freeswitch.org/browse/FS-3229
> Stephen
>
> On Tue, Apr 5, 2011 at 12:16 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> Please try latest git.
>>
>> As I say about 100 times a week. ?Please don't report issues on the
>> mailing list.
>> I almost missed this thread. ?By forcing me to scan the mailing list
>> for bug reports, I do more work and run more of a risk of missing
>> things.
>>
>>
>>
>> On Mon, Apr 4, 2011 at 4:29 PM, Stephen Wilde <wstephen80 at gmail.com>
>> wrote:
>> > FS?commit 57b6255b17e8934a99f07c7467fb3ceaf822a5b4,?Tue Mar 22 15:15:09
>> > 2011
>> > -0500
>> > CentOS 5.5 64bit
>> >
>> >
>> >
>> > On Mon, Apr 4, 2011 at 11:21 PM, Michael Collins <msc at freeswitch.org>
>> > wrote:
>> >>
>> >> what version of FS and what OS?
>> >> -MC
>> >>
>> >> On Mon, Apr 4, 2011 at 1:35 PM, Stephen Wilde <wstephen80 at gmail.com>
>> >> wrote:
>> >>>
>> >>> Ok, I have first tried with:
>> >>> <action application="set" data="execute_on_media=tone_detect mytone
>> >>> 820
>> >>> wo +30000 execute_extension 'myextension XML myextension' 2" />
>> >>> and myextension is NOT executed after tone detection.
>> >>> Stephen
>> >>>
>> >>>
>> >>> On Mon, Apr 4, 2011 at 10:12 PM, Michael Collins <msc at freeswitch.org>
>> >>> wrote:
>> >>>>
>> >>>> Try doing an execute_extension or some other app, just to see if you
>> >>>> can
>> >>>> narrow down where the issue is.
>> >>>> -MC
>> >>>>
>> >>>> On Mon, Apr 4, 2011 at 1:07 PM, Stephen Wilde <wstephen80 at gmail.com>
>> >>>> wrote:
>> >>>>>
>> >>>>> I have tried with single quotas: no change, the tone is detected but
>> >>>>> the set is not executed.
>> >>>>> stephen
>> >>>>>
>> >>>>> On Mon, Apr 4, 2011 at 9:26 PM, Michael Collins <msc at freeswitch.org>
>> >>>>> wrote:
>> >>>>>>
>> >>>>>>
>> >>>>>> On Mon, Apr 4, 2011 at 12:12 PM, Stephen Wilde
>> >>>>>> <wstephen80 at gmail.com>
>> >>>>>> wrote:
>> >>>>>>>
>> >>>>>>> Thank you Michael, yes, I expect that the variable will be in
>> >>>>>>> legA.
>> >>>>>>> Stephen
>> >>>>>>
>> >>>>>> Try putting the app argument in single quotes:
>> >>>>>> <action application="set" data="execute_on_media=tone_detect mytone
>> >>>>>> 820 wo +30000 set 'mytone=true' 2" />
>> >>>>>> -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
>> >>>>
>> >>>
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>
>> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> 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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From msc at freeswitch.org  Tue Apr  5 04:41:35 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Apr 2011 17:41:35 -0700
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTikL68EiJ9S2f7YeD+ktYFpmfLKVMA@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
	<BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
	<BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>
	<BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com>
	<BANLkTikpyu3kNa9cUue95_Krp4i7E19tWA@mail.gmail.com>
	<BANLkTikZ5avXAHzB6CzR2Fc06e6q4wTzUQ@mail.gmail.com>
	<BANLkTika9PtUpPa6BRyALQWm1dsG4F6Ruw@mail.gmail.com>
	<BANLkTimjzTzuomK-RbsBXZahkJKvb-DU6g@mail.gmail.com>
	<BANLkTi=Se6k5Ly8f_NpD8s-yfZjE66poBw@mail.gmail.com>
	<BANLkTi=0kASBQM7gub7T6NG4kiNQwfXtCg@mail.gmail.com>
	<BANLkTi=mmBPoLLjjY2uBkEOAsBpYawezTQ@mail.gmail.com>
	<BANLkTikL68EiJ9S2f7YeD+ktYFpmfLKVMA@mail.gmail.com>
Message-ID: <BANLkTimU2=KUY725JjDHyS4h2kRaqWEA+A@mail.gmail.com>

On Mon, Apr 4, 2011 at 5:14 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> see the updated jira.
>
> I am willing to bet you are causing this problem on yourself with a
> test that is not sending any media to FS because it works for me.
> Nevertheless, I have fixed it for good with some new syntax.
>
> MC, can you update the wiki?
>
Done.
-MC
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From anthony.minessale at gmail.com  Tue Apr  5 04:42:57 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Apr 2011 19:42:57 -0500
Subject: [Freeswitch-users] xml_curl response for voicemail_inject
In-Reply-To: <BANLkTimCsmMSagdMQBPhv=DQo0_ChVNMbA@mail.gmail.com>
References: <BANLkTi=AwsfhJw7j6yLBBmy7hAUwVegkbA@mail.gmail.com>
	<BANLkTikvfBZEYeA=UsoKRhXOJdrnMVgJGQ@mail.gmail.com>
	<BANLkTimCsmMSagdMQBPhv=DQo0_ChVNMbA@mail.gmail.com>
Message-ID: <BANLkTik+OyUkwX6i3k+9i9HYpRuPg7AVqA@mail.gmail.com>

yes should be.  Is that use enclosed in the domain?  maybe there is
some hard to find typo in the xml, try saving it to disk and parsing
it perhaps?


On Mon, Apr 4, 2011 at 12:52 PM, Frank Park <frank at telonium.com> wrote:
> Yeah.. the current response to voicemail_inject is identical to any
> directory lookup, which looks something like this:
> ...
> ?? ? ?<user id="1001">
> ?? ? ? ?<params>
> ?? ? ? ? ?<param name="a1-hash" value="xxxx"/>
> ?? ? ? ? ?<param name="vm-a1-hash" value="xxxx"/>
> ?? ? ? ? ?<param name="inbound-bypass-media" value="true"/>
> ?? ? ? ? ?<param name="vm-email-all-messages" value="true"/>
> ?? ? ? ? ?<param name="vm-attach-file" value="true"/>
> ?? ? ? ? ?<param name="vm-mailto" value="xxxxxx"/>
> ?? ? ? ? ?<param name="vm-keep-local-after-email" value="true"/>
> ?? ? ? ?</params>
> ?? ? ? ?<variables>
> ?? ? ? ? ?<variable name="toll_allow" value="domestic"/>
> ?? ? ? ? ?<variable name="accountcode" value="xxxx"/>
> ?? ? ? ? ?<variable name="effective_caller_id_name" value="xxxxx"/>
> ?? ? ? ? ?<variable name="effective_caller_id_number" value="xxxx"/>
> ?? ? ? ? ?<variable name="user_context" value="xxxx"/>
> ?? ? ? ? ?<variable name="domain_name" value="xxxxx"/>
> ?? ? ? ?</variables>
> ...
> Shouldn't this be enough?
> Frank
>
>
>
>
>
> On Mon, Apr 4, 2011 at 1:36 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> Its just looking for the user record so I can get the params and
>> variables from it.
>>
>>
>> On Mon, Apr 4, 2011 at 10:53 AM, Frank Park <frank at telonium.com> wrote:
>> > I had a quick question.
>> > I am trying to figure out what directory response is expected when a
>> > user
>> > wants to forward a voicemail to another extension.
>> > Looking at the post request of xml_curl, I see that it's invoking
>> > voicemail_inject. I've tried the same response as other voicemail
>> > function,
>> > which is similar to the authorization response, but that didn't seem to
>> > do
>> > it. Anybody care to share an example response when voicemail_inject is
>> > requested?
>> > Thanks,
>> > Frank
>> >
>> > --
>> >
>> > ----=======================----
>> > Frank Park
>> > Telonium Communications, LLC
>> > frank at telonium.com
>> > http://www.telonium.com
>> > Follow Us on Twitter: @GetTelonium
>> > 404-566-8888 x1001 Office
>> > 404-939-4242 Cell
>> > ----=======================----
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> 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
>
>
>
> --
>
> ----=======================----
> Frank Park
> Telonium Communications, LLC
> frank at telonium.com
> http://www.telonium.com
> Follow Us on Twitter: @GetTelonium
> 404-566-8888 x1001 Office
> 404-939-4242 Cell
> ----=======================----
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Tue Apr  5 05:53:39 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 4 Apr 2011 20:53:39 -0500
Subject: [Freeswitch-users] Transcoding with Manual Redirect
In-Reply-To: <ED257D518489452BAA55D8EA69FF897A@kickasspixels.com>
References: <ED257D518489452BAA55D8EA69FF897A@kickasspixels.com>
Message-ID: <BANLkTikUQnmDD-5MQcyZ-hXd8zF0yK48+g@mail.gmail.com>

if the call has already received a 183 or 200, the absolute codec
string will not work.
you may need to enable late-negotiation in your sofia profile.


On Mon, Apr 4, 2011 at 1:43 PM, Lon Baker <lon at kickasspixels.com> wrote:
> I'm trying to force transcoding from PCMU/A to G722 or fallback. I have it
> working through a normal bridge dialplan.
> Another scenario I'm working on is when I receive a call, bridge it to
> another server which issues a redirect. Using the manual redirect settings
> and dialplan context, the following is not working.
> <context name="redirected">
> ?<extension name="redirected">
> ??<condition field="${sip_redirect_dialstring}"
> expression="^sofia/internal/sip:(.*)$">
> <action application="export"
> data="nolocal:absolute_codec_string=G722,PCMU,PCMA"/>
> ?? <action application="bridge" data="sofia/external/$1" />
> ??</condition>
> ?</extension>
> </context>
> It appears to be losing the absolute codec string.
> Any ideas?
> --
> Lon Baker
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From wstephen80 at gmail.com  Tue Apr  5 07:40:56 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 5 Apr 2011 05:40:56 +0200
Subject: [Freeswitch-users] My tone_detect doesn't works
In-Reply-To: <BANLkTikL68EiJ9S2f7YeD+ktYFpmfLKVMA@mail.gmail.com>
References: <BANLkTikGDWxExUy4aHy8rbXB-eeA7SaOEQ@mail.gmail.com>
	<BANLkTin1fbyVL6+O6ij4UQEutONu2=_Rtw@mail.gmail.com>
	<BANLkTinuXsDVfjGjeN-9RvZ0KFWtkQKHUA@mail.gmail.com>
	<BANLkTimQ-hicN0ivsmm75kQKRarVNZz7wA@mail.gmail.com>
	<BANLkTikpyu3kNa9cUue95_Krp4i7E19tWA@mail.gmail.com>
	<BANLkTikZ5avXAHzB6CzR2Fc06e6q4wTzUQ@mail.gmail.com>
	<BANLkTika9PtUpPa6BRyALQWm1dsG4F6Ruw@mail.gmail.com>
	<BANLkTimjzTzuomK-RbsBXZahkJKvb-DU6g@mail.gmail.com>
	<BANLkTi=Se6k5Ly8f_NpD8s-yfZjE66poBw@mail.gmail.com>
	<BANLkTi=0kASBQM7gub7T6NG4kiNQwfXtCg@mail.gmail.com>
	<BANLkTi=mmBPoLLjjY2uBkEOAsBpYawezTQ@mail.gmail.com>
	<BANLkTikL68EiJ9S2f7YeD+ktYFpmfLKVMA@mail.gmail.com>
Message-ID: <BANLkTimT7maD7jf=JWHjKYir0nafQcDQew@mail.gmail.com>

Thank you for this commit, now works also in this case.
I have closed the issue on jira.

Stephen

On Tue, Apr 5, 2011 at 2:14 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> see the updated jira.
>
> I am willing to bet you are causing this problem on yourself with a
> test that is not sending any media to FS because it works for me.
> Nevertheless, I have fixed it for good with some new syntax.
>
> MC, can you update the wiki?
>
>
> On Mon, Apr 4, 2011 at 5:54 PM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > I have tried with latest git without success.
> > I have created an issue on jira:
> http://jira.freeswitch.org/browse/FS-3229
> > Stephen
> >
> > On Tue, Apr 5, 2011 at 12:16 AM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> Please try latest git.
> >>
> >> As I say about 100 times a week.  Please don't report issues on the
> >> mailing list.
> >> I almost missed this thread.  By forcing me to scan the mailing list
> >> for bug reports, I do more work and run more of a risk of missing
> >> things.
> >>
> >>
> >>
> >> On Mon, Apr 4, 2011 at 4:29 PM, Stephen Wilde <wstephen80 at gmail.com>
> >> wrote:
> >> > FS commit 57b6255b17e8934a99f07c7467fb3ceaf822a5b4, Tue Mar 22
> 15:15:09
> >> > 2011
> >> > -0500
> >> > CentOS 5.5 64bit
> >> >
> >> >
> >> >
> >> > On Mon, Apr 4, 2011 at 11:21 PM, Michael Collins <msc at freeswitch.org>
> >> > wrote:
> >> >>
> >> >> what version of FS and what OS?
> >> >> -MC
> >> >>
> >> >> On Mon, Apr 4, 2011 at 1:35 PM, Stephen Wilde <wstephen80 at gmail.com>
> >> >> wrote:
> >> >>>
> >> >>> Ok, I have first tried with:
> >> >>> <action application="set" data="execute_on_media=tone_detect mytone
> >> >>> 820
> >> >>> wo +30000 execute_extension 'myextension XML myextension' 2" />
> >> >>> and myextension is NOT executed after tone detection.
> >> >>> Stephen
> >> >>>
> >> >>>
> >> >>> On Mon, Apr 4, 2011 at 10:12 PM, Michael Collins <
> msc at freeswitch.org>
> >> >>> wrote:
> >> >>>>
> >> >>>> Try doing an execute_extension or some other app, just to see if
> you
> >> >>>> can
> >> >>>> narrow down where the issue is.
> >> >>>> -MC
> >> >>>>
> >> >>>> On Mon, Apr 4, 2011 at 1:07 PM, Stephen Wilde <
> wstephen80 at gmail.com>
> >> >>>> wrote:
> >> >>>>>
> >> >>>>> I have tried with single quotas: no change, the tone is detected
> but
> >> >>>>> the set is not executed.
> >> >>>>> stephen
> >> >>>>>
> >> >>>>> On Mon, Apr 4, 2011 at 9:26 PM, Michael Collins <
> msc at freeswitch.org>
> >> >>>>> wrote:
> >> >>>>>>
> >> >>>>>>
> >> >>>>>> On Mon, Apr 4, 2011 at 12:12 PM, Stephen Wilde
> >> >>>>>> <wstephen80 at gmail.com>
> >> >>>>>> wrote:
> >> >>>>>>>
> >> >>>>>>> Thank you Michael, yes, I expect that the variable will be in
> >> >>>>>>> legA.
> >> >>>>>>> Stephen
> >> >>>>>>
> >> >>>>>> Try putting the app argument in single quotes:
> >> >>>>>> <action application="set" data="execute_on_media=tone_detect
> mytone
> >> >>>>>> 820 wo +30000 set 'mytone=true' 2" />
> >> >>>>>> -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
> >> >>>>
> >> >>>
> >> >>>
> >> >>> _______________________________________________
> >> >>> FreeSWITCH-users mailing list
> >> >>> FreeSWITCH-users at lists.freeswitch.org
> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>>
> >> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >>> http://www.freeswitch.org
> >> >>>
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >>
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> 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
> >> 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
> 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
>
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From joegen at opensipstack.org  Tue Apr  5 08:27:46 2011
From: joegen at opensipstack.org (Joegen E. Baclor)
Date: Tue, 05 Apr 2011 12:27:46 +0800
Subject: [Freeswitch-users] Transfer attempt for a previously a replaced
	call fails
Message-ID: <4D9A9A42.2070804@opensipstack.org>

Hi List,

I have a scenario where a bridged call has been replaced due to a 
consultative transfer.  This works pretty well and audio is 
bidirectional.  I have the original uuid of the call in a var 
somewhere.  The trouble begins when I uuid_deflect the bridged call once 
again to attempt another transfer.  Sofia disconnects the channel.  I am 
using the original uuid of the call (uuid prior to replaces).  Is this 
the right way of doing it?

Joegen


From msc at freeswitch.org  Tue Apr  5 08:35:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 4 Apr 2011 21:35:06 -0700
Subject: [Freeswitch-users] Transfer attempt for a previously a replaced
 call fails
In-Reply-To: <4D9A9A42.2070804@opensipstack.org>
References: <4D9A9A42.2070804@opensipstack.org>
Message-ID: <BANLkTimsX4oJStWYDvz2yhPvf1e9F7kezw@mail.gmail.com>

What do you see on the console when you try this? A console debug log with
siptrace would go a long way toward figuring out what is happening.

-MC

On Mon, Apr 4, 2011 at 9:27 PM, Joegen E. Baclor <joegen at opensipstack.org>wrote:

> Hi List,
>
> I have a scenario where a bridged call has been replaced due to a
> consultative transfer.  This works pretty well and audio is
> bidirectional.  I have the original uuid of the call in a var
> somewhere.  The trouble begins when I uuid_deflect the bridged call once
> again to attempt another transfer.  Sofia disconnects the channel.  I am
> using the original uuid of the call (uuid prior to replaces).  Is this
> the right way of doing it?
>
> Joegen
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From joegen at opensipstack.org  Tue Apr  5 09:51:39 2011
From: joegen at opensipstack.org (Joegen E. Baclor)
Date: Tue, 05 Apr 2011 13:51:39 +0800
Subject: [Freeswitch-users] Transfer attempt for a previously a replaced
 call fails
In-Reply-To: <BANLkTimsX4oJStWYDvz2yhPvf1e9F7kezw@mail.gmail.com>
References: <4D9A9A42.2070804@opensipstack.org>
	<BANLkTimsX4oJStWYDvz2yhPvf1e9F7kezw@mail.gmail.com>
Message-ID: <4D9AADEB.7040803@opensipstack.org>

Hi Michael,

I have pasted both working and none working logs on pastebin.

FreeSWITCH Version 1.0.7 (hacked-20110326T123355Z)
working:  http://pastebin.freeswitch.org/16008
not working:  http://pastebin.freeswitch.org/16009

The call flow for the working call is
UA1 ->  (FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp)
FSIVRApp knows the uuid of the bridge call.  Pressing # on the IVR 
results to a uuid_deflect on the bridged channel.  This works and call 
successfully transfers to the new destination.

The call flow for the none working call is

1.  UA1 -> UA2  is in conversation
2.  UA1 puts UA2 on hold

-- start of FS interaction here --

3.  UA1 ->  (FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp)  (on line 2)
4.  UA1 sends REFER (replacing its call with UA2) to FSBridgeDialPlan.
5.  Flow is now UA2 ->  ([REPLACED]FSBridgeDialPlan) -> (SIP-Loopback) 
-> (FSIVRApp)
6.  UA2 presses #.
7.  IVRApp performs uuid_deflect on FSBridgeDialPlan.
8. FSBridgeDialPlan drops call (no REFER is done)

Thanks for your help.

Joegen

On 04/05/2011 12:35 PM, Michael Collins wrote:
> What do you see on the console when you try this? A console debug log 
> with siptrace would go a long way toward figuring out what is happening.
>
> -MC
>
> On Mon, Apr 4, 2011 at 9:27 PM, Joegen E. Baclor 
> <joegen at opensipstack.org <mailto:joegen at opensipstack.org>> wrote:
>
>     Hi List,
>
>     I have a scenario where a bridged call has been replaced due to a
>     consultative transfer.  This works pretty well and audio is
>     bidirectional.  I have the original uuid of the call in a var
>     somewhere.  The trouble begins when I uuid_deflect the bridged
>     call once
>     again to attempt another transfer.  Sofia disconnects the channel.
>      I am
>     using the original uuid of the call (uuid prior to replaces).  Is this
>     the right way of doing it?
>
>     Joegen
>
>     _______________________________________________
>     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

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From casteven at gmail.com  Tue Apr  5 11:36:30 2011
From: casteven at gmail.com (Campbell Steven)
Date: Tue, 05 Apr 2011 19:36:30 +1200
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <2369DA46-4A45-431B-98F2-4122295C8776@mgtech.com>
References: <4D94C282.1090903@KennedySoftware.ie>
	<4D951D45.5010005@utoronto.ca> <4D95E839.2070403@KennedySoftware.ie>
	<1301712035.18009.1048.camel@macmini>
	<2369DA46-4A45-431B-98F2-4122295C8776@mgtech.com>
Message-ID: <1301988990.18009.1068.camel@macmini>

I kind of feel with the 870 that they decided they were going to make a
touch screen phone first before they decided how they were going to
implement it or how it was going to be useful on the phone. There are
alot of ways of doing the same thing on them which makes them rather
confusing to teach especially with transferring calls, the method of
"dragging" calls in coloured bubbles around the screen is error prone
and just plain difficult in practice given the orientation of the phone.
There is also almost a disconnect between some of the hard keys on the
handset and touch screen and they behave in different ways, especially
the speaker phone key which doesn't work in some scenarios but does in
others. The Vision sidecar that matches has only just in a beta release
got BLF functionality, nearly 12 months after it's release is not great
either (BLF support on a side car is pretty critical I think).  Anyway,
before you jump in I would suggest you buy one and see what you think.

Campbell

On Mon, 2011-04-04 at 15:01 -0700, Mario G wrote:

> I was planning on replacing our Cisco SPA962s with snom 870s. Could
> you please tell me why they are diabolical? Thanks.
> 
> 
> 
> On Apr 1, 2011, at 7:40 PM, Campbell Steven wrote:
> 
> 
> 
> > The Snom 870 will do it with a USB Wifi dongle, but in my experience
> > don't go there, they are a diabolical handset from a usability
> > standpoint.
> > 
> > Campbell
> > 
> > On Fri, 2011-04-01 at 15:59 +0100, Michael Kennedy wrote: 
> > 
> > > Victor,
> > > 
> > > > A bit off-topic but here are my 50 cents:
> > > 
> > > Oopppssss, my apologies - I thought it might be a common query for folks 
> > > thinking about FS - but maybe in another "list"?
> > > 
> > > > -Did you consider building a wireless bridge with a $40 WiFi router
> > > > running DD-WRT/Tomato/OpenWRT etc?
> > > 
> > > I did NOT - and I've deployed a lot of them to support "PC"s! THANK YOU!
> > > 
> > > > This way you can plug wired phones into LAN ports of the "bridge" and
> > > > the router will bridge them to your main access point.
> > > > Asus WL-520GU will work and is really cheap.
> > > 
> > > EXCELLENT suggestion!
> > > 
> > > (Maybe I'm drifting even more O-T, but... I'm also glad you did not 
> > > mention WiFi devices from Linksys - in my experience, some of these 
> > > boxes performed very poorly, but I seem to be the only one on the planet 
> > > with these experiences!).
> > > 
> > > > -If you go with WiFi you should only use WPA or WPA2.
> > > > Less secure options (WEP :-) ) make all conversations accessible to public.
> > > 
> > > Yes, I think all APs are currently running on WPA2.
> > > 
> > > Thank you VERY much, Victor!
> > >    - 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
> > 
> > 
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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From Info at KennedySoftware.ie  Tue Apr  5 15:03:51 2011
From: Info at KennedySoftware.ie (Michael Kennedy)
Date: Tue, 05 Apr 2011 12:03:51 +0100
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <BANLkTin_rQFWV-D56GzxV4dXdvL=bJE+yA@mail.gmail.com>
References: <4D94C282.1090903@KennedySoftware.ie>	<4D9985E5.2060102@chaschperli.ch>	<4D999FB3.5070300@KennedySoftware.ie>
	<BANLkTin_rQFWV-D56GzxV4dXdvL=bJE+yA@mail.gmail.com>
Message-ID: <4D9AF717.6090105@KennedySoftware.ie>

Very many thanks, Brad.

> I've had some decent success with a couple Unidata phones. My only
> qualms were lack of transfer buttons, which you can get around by using
> specified star codes.
> http://www.udcsystems.com/product/sq3000.php
> I believe the Snom M3 has WiFi support as well, which might be a great
> option if these clients are of higher importance.
>
> As for smartphone sip
> clients, I've a strong love for Bria on the iPhone (aside from the
> battery kill while letting it background).
> cSipSimple on the Android market is very nice, integrates with the OS
> and native phone system quite well without such a dramatic hit to the
> battery as far as I can tell.

Very valuable info, and all neatly filed away already!

   - Mike.


From Info at KennedySoftware.ie  Tue Apr  5 15:09:42 2011
From: Info at KennedySoftware.ie (Michael Kennedy)
Date: Tue, 05 Apr 2011 12:09:42 +0100
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <F5586FB8-7CC2-4EDE-B4BB-2C8824CA218D@mgtech.com>
References: <4D94C282.1090903@KennedySoftware.ie>	<4D9985E5.2060102@chaschperli.ch>	<4D999FB3.5070300@KennedySoftware.ie>
	<F5586FB8-7CC2-4EDE-B4BB-2C8824CA218D@mgtech.com>
Message-ID: <4D9AF876.9070301@KennedySoftware.ie>

> And if they are using iPhones we have been very happy with Acrobits Softwphone and Groundwite from http://www.acrobits.cz/27/acrobits-mobile-voip-solutions, also available for android now.

Thank you, Mario.

I think a few employees have iPhones, but, as I mentioned previously, I 
think the penetration of smartphones might reduce rather than increase! 
I don't know if any have Androids.

This info is MOST valuable - thank you.
   - Mike.


From Info at KennedySoftware.ie  Tue Apr  5 15:37:09 2011
From: Info at KennedySoftware.ie (Michael Kennedy)
Date: Tue, 05 Apr 2011 12:37:09 +0100
Subject: [Freeswitch-users] Recommended SIP IP Wifi Handsets?
In-Reply-To: <1301750587582-6233798.post@n2.nabble.com>
References: <4D94C282.1090903@KennedySoftware.ie>
	<1301750587582-6233798.post@n2.nabble.com>
Message-ID: <4D9AFEE5.90100@KennedySoftware.ie>

Mazilo,

My apologies for not acknowledging your comments on the Dockstar: when I 
looked at your email initially, your Dockstar comments appeared in the 
"quoted" text section, and I did not even read them - until a few 
moments ago when I was reviewing all the posts here...

>> I'm hoping to roll out FS where some areas in a building are wired, and
>> other areas are on WiFi, and to deploy some SIP phones in both areas.If
>> you can still find an inexpensive
>> http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
>> Seagate FreeAgent DockStar  (used to be on sale for as low as
>> $13.99/each), you certainly can use it to host your FS. It is an ARM
>> platform clocked @1.2GHz with 128/256MB RAM/NAND, 4 USB2 ports, and a
>> single Gigabit RJ-45 port. Unless you already have a NAT/Firewall WiFi
>> router, all you need is an additional USB WiFi dongle to make it
>> WiFi-able.

I'll certainly look at that - thank you. Each office already has a Linux 
server (Ubuntu 8.04/10.04, mainly as a file- and comms-server, FW, 
router, etc), which is mostly just "relaxing" all day, and I'm thinking 
of putting FS on these boxes, and use the existing APs where WiFi is active.

However... that Dockstar looks extremely interesting, and incredibly 
cheap! I'll certainly investigate it, and see where it might fit in. 
Thank you.

   - Mike


From dujinfang at gmail.com  Tue Apr  5 16:21:45 2011
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 5 Apr 2011 20:21:45 +0800
Subject: [Freeswitch-users] video problem in conference with H264
In-Reply-To: <BANLkTinEbJhmLWFZ=HfjbF5+gvmJgFE9YA@mail.gmail.com>
References: <AD0F952609504D9D8E8AC968ADABE0C8@gmail.com>
	<942671FE18024326A312ED29ACE4F5DD@gmail.com>
	<BANLkTinEbJhmLWFZ=HfjbF5+gvmJgFE9YA@mail.gmail.com>
Message-ID: <020B55848F7D44ACB95027A6713FE74B@gmail.com>

Thank you Anthony, I will work more on video and will keep this in mind.
On Sunday, April 3, 2011 at 3:31 AM, Anthony Minessale wrote:
If you can find a patch to properly tell full frames on various codec it would be nice. That code is unfinished.
> On Apr 2, 2011 2:46 AM, "Seven Du" <dujinfang at gmail.com> wrote:
> > Answer myself.
> > 
> > 
> > I traced code and found that in line 1007 of mod_conference the frame data never matches 0x11
> > 
> > } else if (vid_frame->codec->implementation->ianacode == 99) { /* h.264 */
> > iframe = (*((int16_t *) vid_frame->data) >> 5 == 0x11);
> > 
> > I hardcoded to iframe = 1 and then it works.
> > 
> > As I said I don't have problem with Bria, but with the xtp8886 device I got the following data sequence. I'm not familiar with video encoding, so is my device broken or we need other methods to detect an i-frame or is it safe to just hard coded into 1?
> > 
> > Thanks.
> > 
> > 
> > 
> > *(int16_t *) vid_frame->data, *((int16_t *) vid_frame->data) >> 5
> > 
> > ffffb465, fffffda3
> > 165, b
> > 165, b
> > 65, 3
> > ffffd061, fffffe83
> > 61, 3
> > ffffd061, fffffe83
> > 161, b
> > 4267, 213
> > 4868, 243
> > ffffb465, fffffda3
> > 165, b
> > 165, b
> > 65, 3
> > ffffd061, fffffe83
> > 61, 3
> > ffffd061, fffffe83
> > 161, b
> > 61, 3
> > ffffd061, fffffe83
> > 461, 23
> > 161, b
> > 61, 3
> > ffffd061, fffffe83
> > 161, b
> > ffffd061, fffffe83
> > 361, 1b
> > 61, 3
> > ffffd061, fffffe83
> > 161, b
> > ffffd061, fffffe83
> > 361, 1b
> > 161, b
> > ffffd061, fffffe83
> > 61, 3
> > ffffd061, fffffe83
> > 361, 1b
> > 161, b
> > ffffd061, fffffe83
> > 161, b
> > ffffd061, fffffe83
> > 361, 1b
> > 161, b
> > 61, 3
> > ffffd061, fffffe83
> > 161, b
> > ffffd061, fffffe83
> > 361, 1b
> > 161, b
> > 61, 3
> > ffffd061, fffffe83
> > 161, b
> > 4267, 213
> > 4868, 243
> > ffffb465, fffffda3
> > 165, b
> > 165, b
> > 65, 3
> > ffffd061, fffffe83
> > ffffd061, fffffe83
> > 61, 3
> > ffffd061, fffffe83
> > 161, b
> > ffffd061, fffffe83
> > 161, b
> > 61, 3
> > ffffd061, fffffe83
> > 161, b
> > ffffd061, fffffe83
> > 161, b
> > 61, 3
> > 4267, 213
> > 4868, 243
> > ffffb465, fffffda3
> > 365, 1b
> > 165, b
> > 65, 3
> > ffffd061, fffffe83
> > ffffd061, fffffe83
> > 61, 3
> > ffffd061, fffffe83
> > 61, 3
> > ffffd061, fffffe83
> > 161, b
> > 
> > 
> > On Wednesday, March 30, 2011 at 7:31 PM, Seven Du wrote: 
> >> I tested with default 3000 conference and it just OK. But I have problem on H264.
> >> 
> >> I tested with one Bria 3.1 on Mac and two XTP8886 hardware phones.
> >> 
> >> http://www.gvscusa.com/xtp8886.html
> >> 
> >> Bria 1003
> >> XTP 1011/1012
> >> 
> >> call from 1003 to 1011 and from 1011 to 1003 both ok with videos.
> >> 
> >> http://pastebin.freeswitch.org/15910
> >> http://pastebin.freeswitch.org/15911
> >> 
> >> When 3 phones calling into 3000(conference), Everyone call see Bria(1003), but no one can say 1011 and 1012. Even when I muted 1003. 
> >> 
> >> http://pastebin.freeswitch.org/15913
> >> 
> >> As I said there's no problem with similar test with h263.
> >> 
> >> Can anyone help take a look, thanks.
> >> 
> >> -- 
> >> About: http://about.me/dujinfang
> >> Blog: http://www.dujinfang.com
> >> Proj: http://www.freeswitch.org.cn
> >> 
> >> Sent with Sparrow
> >> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
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From frank at telonium.com  Tue Apr  5 18:01:45 2011
From: frank at telonium.com (Frank Park)
Date: Tue, 5 Apr 2011 10:01:45 -0400
Subject: [Freeswitch-users] xml_curl response for voicemail_inject
In-Reply-To: <BANLkTik+OyUkwX6i3k+9i9HYpRuPg7AVqA@mail.gmail.com>
References: <BANLkTi=AwsfhJw7j6yLBBmy7hAUwVegkbA@mail.gmail.com>
	<BANLkTikvfBZEYeA=UsoKRhXOJdrnMVgJGQ@mail.gmail.com>
	<BANLkTimCsmMSagdMQBPhv=DQo0_ChVNMbA@mail.gmail.com>
	<BANLkTik+OyUkwX6i3k+9i9HYpRuPg7AVqA@mail.gmail.com>
Message-ID: <BANLkTi=CHYXEE=hQmM9qE16U96u=7mMPdw@mail.gmail.com>

Yes, this is enclosed in a domain and works very well for authorization of
extensions and other voicemail functions, but when forward option is pressed
in the IVR and when a user enters the extensions, it responds by "invalid
extension".

Frank



On Mon, Apr 4, 2011 at 8:42 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> yes should be.  Is that use enclosed in the domain?  maybe there is
> some hard to find typo in the xml, try saving it to disk and parsing
> it perhaps?
>
>
> On Mon, Apr 4, 2011 at 12:52 PM, Frank Park <frank at telonium.com> wrote:
> > Yeah.. the current response to voicemail_inject is identical to any
> > directory lookup, which looks something like this:
> > ...
> >       <user id="1001">
> >         <params>
> >           <param name="a1-hash" value="xxxx"/>
> >           <param name="vm-a1-hash" value="xxxx"/>
> >           <param name="inbound-bypass-media" value="true"/>
> >           <param name="vm-email-all-messages" value="true"/>
> >           <param name="vm-attach-file" value="true"/>
> >           <param name="vm-mailto" value="xxxxxx"/>
> >           <param name="vm-keep-local-after-email" value="true"/>
> >         </params>
> >         <variables>
> >           <variable name="toll_allow" value="domestic"/>
> >           <variable name="accountcode" value="xxxx"/>
> >           <variable name="effective_caller_id_name" value="xxxxx"/>
> >           <variable name="effective_caller_id_number" value="xxxx"/>
> >           <variable name="user_context" value="xxxx"/>
> >           <variable name="domain_name" value="xxxxx"/>
> >         </variables>
> > ...
> > Shouldn't this be enough?
> > Frank
> >
> >
> >
> >
> >
> > On Mon, Apr 4, 2011 at 1:36 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> Its just looking for the user record so I can get the params and
> >> variables from it.
> >>
> >>
> >> On Mon, Apr 4, 2011 at 10:53 AM, Frank Park <frank at telonium.com> wrote:
> >> > I had a quick question.
> >> > I am trying to figure out what directory response is expected when a
> >> > user
> >> > wants to forward a voicemail to another extension.
> >> > Looking at the post request of xml_curl, I see that it's invoking
> >> > voicemail_inject. I've tried the same response as other voicemail
> >> > function,
> >> > which is similar to the authorization response, but that didn't seem
> to
> >> > do
> >> > it. Anybody care to share an example response when voicemail_inject is
> >> > requested?
> >> > Thanks,
> >> > Frank
> >> >
> >> > --
> >> >
> >> > ----=======================----
> >> > Frank Park
> >> > Telonium Communications, LLC
> >> > frank at telonium.com
> >> > http://www.telonium.com
> >> > Follow Us on Twitter: @GetTelonium
> >> > 404-566-8888 x1001 Office
> >> > 404-939-4242 Cell
> >> > ----=======================----
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> 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
> >> 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
> >
> >
> >
> > --
> >
> > ----=======================----
> > Frank Park
> > Telonium Communications, LLC
> > frank at telonium.com
> > http://www.telonium.com
> > Follow Us on Twitter: @GetTelonium
> > 404-566-8888 x1001 Office
> > 404-939-4242 Cell
> > ----=======================----
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> 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
>



-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From frank at telonium.com  Tue Apr  5 18:08:58 2011
From: frank at telonium.com (Frank Park)
Date: Tue, 5 Apr 2011 10:08:58 -0400
Subject: [Freeswitch-users] xml_curl response for voicemail_inject
In-Reply-To: <BANLkTi=CHYXEE=hQmM9qE16U96u=7mMPdw@mail.gmail.com>
References: <BANLkTi=AwsfhJw7j6yLBBmy7hAUwVegkbA@mail.gmail.com>
	<BANLkTikvfBZEYeA=UsoKRhXOJdrnMVgJGQ@mail.gmail.com>
	<BANLkTimCsmMSagdMQBPhv=DQo0_ChVNMbA@mail.gmail.com>
	<BANLkTik+OyUkwX6i3k+9i9HYpRuPg7AVqA@mail.gmail.com>
	<BANLkTi=CHYXEE=hQmM9qE16U96u=7mMPdw@mail.gmail.com>
Message-ID: <BANLkTinwdLPAYawgi4Z0a4nTL42_dkYVxA@mail.gmail.com>

Is there a way to disable this option in the VM prompt until I can fix this
issue? I didn't see the option in the voicemail.conf.xml


Frank


On Tue, Apr 5, 2011 at 10:01 AM, Frank Park <frank at telonium.com> wrote:

> Yes, this is enclosed in a domain and works very well for authorization of
> extensions and other voicemail functions, but when forward option is pressed
> in the IVR and when a user enters the extensions, it responds by "invalid
> extension".
>
> Frank
>
>
>
> On Mon, Apr 4, 2011 at 8:42 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> yes should be.  Is that use enclosed in the domain?  maybe there is
>> some hard to find typo in the xml, try saving it to disk and parsing
>> it perhaps?
>>
>>
>> On Mon, Apr 4, 2011 at 12:52 PM, Frank Park <frank at telonium.com> wrote:
>> > Yeah.. the current response to voicemail_inject is identical to any
>> > directory lookup, which looks something like this:
>> > ...
>> >       <user id="1001">
>> >         <params>
>> >           <param name="a1-hash" value="xxxx"/>
>> >           <param name="vm-a1-hash" value="xxxx"/>
>> >           <param name="inbound-bypass-media" value="true"/>
>> >           <param name="vm-email-all-messages" value="true"/>
>> >           <param name="vm-attach-file" value="true"/>
>> >           <param name="vm-mailto" value="xxxxxx"/>
>> >           <param name="vm-keep-local-after-email" value="true"/>
>> >         </params>
>> >         <variables>
>> >           <variable name="toll_allow" value="domestic"/>
>> >           <variable name="accountcode" value="xxxx"/>
>> >           <variable name="effective_caller_id_name" value="xxxxx"/>
>> >           <variable name="effective_caller_id_number" value="xxxx"/>
>> >           <variable name="user_context" value="xxxx"/>
>> >           <variable name="domain_name" value="xxxxx"/>
>> >         </variables>
>> > ...
>> > Shouldn't this be enough?
>> > Frank
>> >
>> >
>> >
>> >
>> >
>> > On Mon, Apr 4, 2011 at 1:36 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> Its just looking for the user record so I can get the params and
>> >> variables from it.
>> >>
>> >>
>> >> On Mon, Apr 4, 2011 at 10:53 AM, Frank Park <frank at telonium.com>
>> wrote:
>> >> > I had a quick question.
>> >> > I am trying to figure out what directory response is expected when a
>> >> > user
>> >> > wants to forward a voicemail to another extension.
>> >> > Looking at the post request of xml_curl, I see that it's invoking
>> >> > voicemail_inject. I've tried the same response as other voicemail
>> >> > function,
>> >> > which is similar to the authorization response, but that didn't seem
>> to
>> >> > do
>> >> > it. Anybody care to share an example response when voicemail_inject
>> is
>> >> > requested?
>> >> > Thanks,
>> >> > Frank
>> >> >
>> >> > --
>> >> >
>> >> > ----=======================----
>> >> > Frank Park
>> >> > Telonium Communications, LLC
>> >> > frank at telonium.com
>> >> > http://www.telonium.com
>> >> > Follow Us on Twitter: @GetTelonium
>> >> > 404-566-8888 x1001 Office
>> >> > 404-939-4242 Cell
>> >> > ----=======================----
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> 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
>> >> 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
>> >
>> >
>> >
>> > --
>> >
>> > ----=======================----
>> > Frank Park
>> > Telonium Communications, LLC
>> > frank at telonium.com
>> > http://www.telonium.com
>> > Follow Us on Twitter: @GetTelonium
>> > 404-566-8888 x1001 Office
>> > 404-939-4242 Cell
>> > ----=======================----
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> 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
>>
>
>
>
> --
>
> ----=======================----
> Frank Park
> Telonium Communications, LLC
> frank at telonium.com
> http://www.telonium.com
> Follow Us on Twitter: @GetTelonium
> 404-566-8888 x1001 Office
> 404-939-4242 Cell
> ----=======================----
>
>
>


-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From Lars.Bobka at web.de  Tue Apr  5 12:00:38 2011
From: Lars.Bobka at web.de (Lars Bobka)
Date: Tue, 5 Apr 2011 10:00:38 +0200 (CEST)
Subject: [Freeswitch-users] Gateway ReInvite Problem with a=inactive in SDP
Message-ID: <1418601206.2098287.1301990438986.JavaMail.fmail@mwmweb033>

Hi,

I have a problem with ReInvites and bypass-media option.
A call over a gateway comes in. The gateway is a shared line from a broadsoft application server.
The Invite over the gateway shows an a=inactive in the SDP.

v=0
o=BroadWorks 66603944 1 IN IP4 xxx.xxx.xxx.xxx
s=-
c=IN IP4 0.0.0.0
t=0 0
m=audio 21568 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=ptime:30
a=fmtp:101 0-15
a=bsoft: 1 image udptl t38
a=inactive

After sending the 200OK over the gateway ang getting an ACK, the broadworks server sends a ReInvite, because of the shared line.
Now the FS should answer the ReInvite with a 200Ok and sendrecv, but the FS sends inactive to the broadworks server and I have no audio
in the call. Only when I hold hold the call and unhold the call, the inactive messages is not given and I have 2way audio.

I try <param name="NDLB-sendrecv-in-session" value="true"/> everywhere (external.xml, internal.xml, user.xml), but I had no luck.
So could you please help me.

Regards Lars
___________________________________________________________
Schon geh?rt? WEB.DE hat einen genialen Phishing-Filter in die
Toolbar eingebaut! http://produkte.web.de/go/toolbar


From rajkumar.kmry at gmail.com  Tue Apr  5 14:52:26 2011
From: rajkumar.kmry at gmail.com (Rajkumar K)
Date: Tue, 5 Apr 2011 16:22:26 +0530
Subject: [Freeswitch-users] how to find available agents in mod callcenter
	queue
Message-ID: <BANLkTinDOjzMs3wwNpYSiMbXcqxHxoOdQg@mail.gmail.com>

Hi,

Is there any way to find the available agents count for the particular queue
in mod callcenter.
Note: available agents refers to agents who are all not in call and waiting
for it.

regards
rajkumar k
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From rajkumar.kmry at gmail.com  Tue Apr  5 16:07:45 2011
From: rajkumar.kmry at gmail.com (Rajkumar K)
Date: Tue, 5 Apr 2011 17:37:45 +0530
Subject: [Freeswitch-users] callcenter_config commands
Message-ID: <BANLkTinxAck_VS+9tK6ekoJeRy=6zPoSqg@mail.gmail.com>

Hi,

I found the following useful commands in mod_callcenter module.

callcenter_config queue list agents [queue_name] [status]
callcenter_config queue list members [queue_name]
callcenter_config queue list tiers [queue_name]
callcenter_config queue count agents [queue_name] [status]
callcenter_config queue count members [queue_name]

But when I run the following command, it says simply "+OK" and no members
list even though there are calls in queue1.

  => callcenter_config queue list members queue1

I am using FreeSWITCH Version 1.0.head (git-8f2ee97 2010-12-05 17-19-28
-0600). Should I have to use any latest version?

regards
rajkumar k
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From msc at freeswitch.org  Tue Apr  5 19:21:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Apr 2011 08:21:52 -0700
Subject: [Freeswitch-users] xml_curl response for voicemail_inject
In-Reply-To: <BANLkTinwdLPAYawgi4Z0a4nTL42_dkYVxA@mail.gmail.com>
References: <BANLkTi=AwsfhJw7j6yLBBmy7hAUwVegkbA@mail.gmail.com>
	<BANLkTikvfBZEYeA=UsoKRhXOJdrnMVgJGQ@mail.gmail.com>
	<BANLkTimCsmMSagdMQBPhv=DQo0_ChVNMbA@mail.gmail.com>
	<BANLkTik+OyUkwX6i3k+9i9HYpRuPg7AVqA@mail.gmail.com>
	<BANLkTi=CHYXEE=hQmM9qE16U96u=7mMPdw@mail.gmail.com>
	<BANLkTinwdLPAYawgi4Z0a4nTL42_dkYVxA@mail.gmail.com>
Message-ID: <BANLkTinzMqjawEEduXjV4bWNJE=n14p_gA@mail.gmail.com>

On Tue, Apr 5, 2011 at 7:08 AM, Frank Park <frank at telonium.com> wrote:

> Is there a way to disable this option in the VM prompt until I can fix this
> issue? I didn't see the option in the voicemail.conf.xml
>
> If you are just wanting to disable to voicing of the option then you need
to look in conf/lang/en/vm/sounds.xml. Find the phrase macro that voices the
caller options and comment out the one that says to press x to forward the
message.

-MC
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From max.clark at gmail.com  Tue Apr  5 19:26:48 2011
From: max.clark at gmail.com (Max Clark)
Date: Tue, 5 Apr 2011 08:26:48 -0700
Subject: [Freeswitch-users] Sane Dialplan Regex
Message-ID: <BANLkTimsBAa-vBwpdhwyCejnttiDaKTxWQ@mail.gmail.com>

Hello,

I am trying to clean up the number of invalid calls being sent
upstream to PSTN gateways. I've been researching numbering formats and
could use some guidance/clarification.

Currently I'm matching NADP calls looking by looking for a destination
number matching 1 + 10 digits:

<condition field="destination_number" expression="^(1\d{10})$">

>From what I've found NPA should be 201-999 and NXX should be 200-999
(ugly regex):

<condition field="destination_number"
expression="^(1[2-9][0-9]{2}[2-9]\d{2}\d{4})$">

Are there any cases where the NPA & NXX would not conform to this list?

Thanks,
Max


From monemran at gmail.com  Tue Apr  5 19:31:33 2011
From: monemran at gmail.com (M.Emran)
Date: Tue, 5 Apr 2011 21:31:33 +0600
Subject: [Freeswitch-users] How to get remote gateway IP
Message-ID: <BANLkTinRcJjnkfm8X=HOjtsiaqQ5k8wQfA@mail.gmail.com>

Hi,

What is the way to read variable for termination gateway ip while it was
ringing/answer and stop in lua script?

-- 
Regards
----------
M Emran
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From rhuddleston at gmail.com  Tue Apr  5 19:35:40 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Tue, 5 Apr 2011 11:35:40 -0400
Subject: [Freeswitch-users] How to get remote gateway IP
In-Reply-To: <BANLkTinRcJjnkfm8X=HOjtsiaqQ5k8wQfA@mail.gmail.com>
References: <BANLkTinRcJjnkfm8X=HOjtsiaqQ5k8wQfA@mail.gmail.com>
Message-ID: <235f01cbf3a7$1eeda590$5cc8f0b0$@com>

I get it from the XML CDR parsing facility I wrote. I have a LUA script that
churns through multiple gateways - but I post parse the XML for the answer

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of M.Emran
Sent: Tuesday, April 05, 2011 11:32 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] How to get remote gateway IP

 

Hi,

What is the way to read variable for termination gateway ip while it was
ringing/answer and stop in lua script?

-- 
Regards 
---------- 
M Emran

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From monemran at gmail.com  Tue Apr  5 19:42:39 2011
From: monemran at gmail.com (M.Emran)
Date: Tue, 5 Apr 2011 21:42:39 +0600
Subject: [Freeswitch-users] How to get remote gateway IP
In-Reply-To: <235f01cbf3a7$1eeda590$5cc8f0b0$@com>
References: <BANLkTinRcJjnkfm8X=HOjtsiaqQ5k8wQfA@mail.gmail.com>
	<235f01cbf3a7$1eeda590$5cc8f0b0$@com>
Message-ID: <BANLkTi=DJMANN4MspaKcBGu2V1qM_BaBMQ@mail.gmail.com>

But i am using execute_on_ring, execute_on_answer and api_hangup_hook for
cdr.
All variable can read except the termination ip. any one knows how to do
that ?


On Tue, Apr 5, 2011 at 9:35 PM, Robert Huddleston <rhuddleston at gmail.com>wrote:

> I get it from the XML CDR parsing facility I wrote? I have a LUA script
> that churns through multiple gateways ? but I post parse the XML for the
> answer
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *M.Emran
> *Sent:* Tuesday, April 05, 2011 11:32 AM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] How to get remote gateway IP
>
>
>
> Hi,
>
> What is the way to read variable for termination gateway ip while it was
> ringing/answer and stop in lua script?
>
> --
> Regards
> ----------
> M Emran
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
----------
M Emran
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From msc at freeswitch.org  Tue Apr  5 19:56:13 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Apr 2011 08:56:13 -0700
Subject: [Freeswitch-users] Sane Dialplan Regex
In-Reply-To: <BANLkTimsBAa-vBwpdhwyCejnttiDaKTxWQ@mail.gmail.com>
References: <BANLkTimsBAa-vBwpdhwyCejnttiDaKTxWQ@mail.gmail.com>
Message-ID: <BANLkTik2kMuuD6PZkh9RegiA4a5Qc0nOBg@mail.gmail.com>

On Tue, Apr 5, 2011 at 8:26 AM, Max Clark <max.clark at gmail.com> wrote:

> Hello,
>
> I am trying to clean up the number of invalid calls being sent
> upstream to PSTN gateways. I've been researching numbering formats and
> could use some guidance/clarification.
>
> Currently I'm matching NADP calls looking by looking for a destination
> number matching 1 + 10 digits:
>
> <condition field="destination_number" expression="^(1\d{10})$">
>
> >From what I've found NPA should be 201-999 and NXX should be 200-999
> (ugly regex):
>
> <condition field="destination_number"
> expression="^(1[2-9][0-9]{2}[2-9]\d{2}\d{4})$">
>

I don't believe that regex is correct. You might want to base your pattern
off the one listed here:
http://wiki.freeswitch.org/wiki/Regex#NANPA_.2B1NxxNxxXXXX_E.164_Dialstring

>
> Are there any cases where the NPA & NXX would not conform to this list?
>
Not unless NANPA decides to make some changes...
-MC
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From msc at freeswitch.org  Tue Apr  5 19:57:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Apr 2011 08:57:54 -0700
Subject: [Freeswitch-users] How to get remote gateway IP
In-Reply-To: <BANLkTi=DJMANN4MspaKcBGu2V1qM_BaBMQ@mail.gmail.com>
References: <BANLkTinRcJjnkfm8X=HOjtsiaqQ5k8wQfA@mail.gmail.com>
	<235f01cbf3a7$1eeda590$5cc8f0b0$@com>
	<BANLkTi=DJMANN4MspaKcBGu2V1qM_BaBMQ@mail.gmail.com>
Message-ID: <BANLkTin-h=+MbAO+21LukF-ZeDjEStgupA@mail.gmail.com>

On Tue, Apr 5, 2011 at 8:42 AM, M.Emran <monemran at gmail.com> wrote:

> But i am using execute_on_ring, execute_on_answer and api_hangup_hook for
> cdr.
> All variable can read except the termination ip. any one knows how to do
> that ?
>
> What are you using for your hangup hook? Did you do this:

http://wiki.freeswitch.org/wiki/Channel_Variables#session_in_hangup_hook

-MC
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From msc at freeswitch.org  Tue Apr  5 20:21:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Apr 2011 09:21:29 -0700
Subject: [Freeswitch-users] simultaneous voice conference question
In-Reply-To: <963D2073207646E2BAFF12EC8F8D0656@e1705>
References: <963D2073207646E2BAFF12EC8F8D0656@e1705>
Message-ID: <BANLkTimOqNN8BMoOSBfq1P0u_XRK69k0Ng@mail.gmail.com>

I'd get a tcpdump w/ media and try to analyze in wireshark.
-MC

On Sun, Apr 3, 2011 at 10:18 AM, Madovsky <infos at madovsky.org> wrote:

>  When at least 3 persons are in conference I noticed
> that we can't hear the 3 voices in same time, only one voice at a time.
> is it normal ?
>
> 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
>
>
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From steveu at coppice.org  Tue Apr  5 20:27:03 2011
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 06 Apr 2011 00:27:03 +0800
Subject: [Freeswitch-users] Hylafax server emulation
Message-ID: <4D9B42D7.4020008@coppice.org>

Hi,

It has always been clear that a HylaFAX compatible FAX job submission 
server would add considerably to the value of the FAX facilities in 
Asterisk and Freeswitch, but somehow it hasn't happened until now. I 
recently found that in 2005 someone produced something fairly basic for 
Asterisk in Perl, but it doesn't seem to have been well publicised, and 
it looks like development stalled long ago.

I now have the skeleton of HylaFAX compatible FAX job submission server, 
in C, working. It accepts FAX submissions from sendfax and a couple of 
the windows HylaFAX clients, though it needs a lot more polishing. Now I 
need to look at the best thing to do on the Freeswitch side. I aim to 
make the server maintain its own database of FAX jobs. It will attach to 
Freeswitch, by ESL; push the jobs through FS; deal with scheduling, 
retries, etc; and report the final result to the user, just as HylaFAX 
does. The thing I am rather unsure about is the best way to handle the 
accounts used to accept FAX jobs? Should I maintain a separate database 
of FAX accounts, or hook into an existing database? I would welcome 
suggestions for what would be the most useful approach.

Steve



From anthony.minessale at gmail.com  Tue Apr  5 21:03:19 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Apr 2011 12:03:19 -0500
Subject: [Freeswitch-users] Hylafax server emulation
In-Reply-To: <4D9B42D7.4020008@coppice.org>
References: <4D9B42D7.4020008@coppice.org>
Message-ID: <BANLkTi=VU6kUNXkwJ67_gxP+R9pcsm5t5w@mail.gmail.com>

This is a common cross-roads.  I think your choices are:

1) Try and be agnostic like we have done in FreeSWITCH or to just
choose a particular database and make it a dependency.  There is a
fair argument to either side.  I personally chose the agnostic
approach because I did not want to limit the possibilities of how
developers chose to integrate FreeSWITCH into their existing
infrastructure.  It allows you to use it standalone or connect to any
existing db with ODBC and create its own tables.

2) Use a specific DB.  You have a fairly specific application with a
fairly specific task so it would not be a bad decision to just choose
a db with a well-developed client API like postgres, sqlite etc.
There are some disadvantages to ODBC if the implementation of the
connector of choice is poorly done or has memory leaks or artificial
limitations.  Also you must then conform to ANSI sql with ODBC.
Native db API would give you any specific extension etc to that db.


I may have just re-iterated your question with more specific details
but maybe that can help you decide.  I guess it would be based on your
intentions and how flexible you want to make it at the cost of extra
abstraction.




On Tue, Apr 5, 2011 at 11:27 AM, Steve Underwood <steveu at coppice.org> wrote:
> Hi,
>
> It has always been clear that a HylaFAX compatible FAX job submission
> server would add considerably to the value of the FAX facilities in
> Asterisk and Freeswitch, but somehow it hasn't happened until now. I
> recently found that in 2005 someone produced something fairly basic for
> Asterisk in Perl, but it doesn't seem to have been well publicised, and
> it looks like development stalled long ago.
>
> I now have the skeleton of HylaFAX compatible FAX job submission server,
> in C, working. It accepts FAX submissions from sendfax and a couple of
> the windows HylaFAX clients, though it needs a lot more polishing. Now I
> need to look at the best thing to do on the Freeswitch side. I aim to
> make the server maintain its own database of FAX jobs. It will attach to
> Freeswitch, by ESL; push the jobs through FS; deal with scheduling,
> retries, etc; and report the final result to the user, just as HylaFAX
> does. The thing I am rather unsure about is the best way to handle the
> accounts used to accept FAX jobs? Should I maintain a separate database
> of FAX accounts, or hook into an existing database? I would welcome
> suggestions for what would be the most useful approach.
>
> 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
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Tue Apr  5 21:17:43 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 5 Apr 2011 12:17:43 -0500
Subject: [Freeswitch-users] Gateway ReInvite Problem with a=inactive in
	SDP
In-Reply-To: <1418601206.2098287.1301990438986.JavaMail.fmail@mwmweb033>
References: <1418601206.2098287.1301990438986.JavaMail.fmail@mwmweb033>
Message-ID: <BANLkTikur2CNRr70G-4jQwvg-oJn0zK4Fw@mail.gmail.com>

bypass media means FS does not get involved, all it will do is forward
the sdp as-is to the other party.



On Tue, Apr 5, 2011 at 3:00 AM, Lars Bobka <Lars.Bobka at web.de> wrote:
> Hi,
>
> I have a problem with ReInvites and bypass-media option.
> A call over a gateway comes in. The gateway is a shared line from a broadsoft application server.
> The Invite over the gateway shows an a=inactive in the SDP.
>
> v=0
> o=BroadWorks 66603944 1 IN IP4 xxx.xxx.xxx.xxx
> s=-
> c=IN IP4 0.0.0.0
> t=0 0
> m=audio 21568 RTP/AVP 8 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=ptime:30
> a=fmtp:101 0-15
> a=bsoft: 1 image udptl t38
> a=inactive
>
> After sending the 200OK over the gateway ang getting an ACK, the broadworks server sends a ReInvite, because of the shared line.
> Now the FS should answer the ReInvite with a 200Ok and sendrecv, but the FS sends inactive to the broadworks server and I have no audio
> in the call. Only when I hold hold the call and unhold the call, the inactive messages is not given and I have 2way audio.
>
> I try <param name="NDLB-sendrecv-in-session" value="true"/> everywhere (external.xml, internal.xml, user.xml), but I had no luck.
> So could you please help me.
>
> Regards Lars
> ___________________________________________________________
> Schon geh?rt? WEB.DE hat einen genialen Phishing-Filter in die
> Toolbar eingebaut! http://produkte.web.de/go/toolbar
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From nicevoip at googlemail.com  Tue Apr  5 21:39:53 2011
From: nicevoip at googlemail.com (Nice Voip)
Date: Tue, 5 Apr 2011 19:39:53 +0200
Subject: [Freeswitch-users] Connecting HP 8500 All in One FAX Machine to FS
Message-ID: <BANLkTikawF1e2UX4-AparihLMYi=JQ7ZcA@mail.gmail.com>

Hi All,

I have got HP 8500 All in One machine, including FAX, now i want to use it
to send faxes to my VoIP provider through FS, how do i connect it to FS? it
has wireless card, but i don't see any VoIP settings.

Many thanks.
- NV
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From william.suffill at gmail.com  Tue Apr  5 22:12:52 2011
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 5 Apr 2011 14:12:52 -0400
Subject: [Freeswitch-users] Connecting HP 8500 All in One FAX Machine to
	FS
In-Reply-To: <BANLkTikawF1e2UX4-AparihLMYi=JQ7ZcA@mail.gmail.com>
References: <BANLkTikawF1e2UX4-AparihLMYi=JQ7ZcA@mail.gmail.com>
Message-ID: <BANLkTikDGdgqmPb_kYxebpf0K-U838oFUw@mail.gmail.com>

The HP doesn't know anything about voip. You would have to connect via an
phone cable to an ATA which would handle the analog to voip for you. Another
option would be to just scan with the HP and use FS to Fax the images.

-- W
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From curriegrad2004 at gmail.com  Tue Apr  5 22:16:15 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Tue, 5 Apr 2011 11:16:15 -0700
Subject: [Freeswitch-users] Connecting HP 8500 All in One FAX Machine to
	FS
In-Reply-To: <BANLkTikDGdgqmPb_kYxebpf0K-U838oFUw@mail.gmail.com>
References: <BANLkTikawF1e2UX4-AparihLMYi=JQ7ZcA@mail.gmail.com>
	<BANLkTikDGdgqmPb_kYxebpf0K-U838oFUw@mail.gmail.com>
Message-ID: <BANLkTik_UHtC-1a+FL_hrKuLLVjxC_gdow@mail.gmail.com>

And remember to use PCMU (G711) or T.38 on the server to send faxes...

On Tue, Apr 5, 2011 at 11:12 AM, William Suffill
<william.suffill at gmail.com> wrote:
> The HP doesn't know anything about voip. You would have to connect via an
> phone cable to an ATA which would handle the analog to voip for you. Another
> option would be to just scan with the HP and use FS to Fax the images.
> -- W
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From nicevoip at googlemail.com  Tue Apr  5 22:24:12 2011
From: nicevoip at googlemail.com (Nice Voip)
Date: Tue, 5 Apr 2011 20:24:12 +0200
Subject: [Freeswitch-users] Connecting HP 8500 All in One FAX Machine to
	FS
In-Reply-To: <BANLkTik_UHtC-1a+FL_hrKuLLVjxC_gdow@mail.gmail.com>
References: <BANLkTikawF1e2UX4-AparihLMYi=JQ7ZcA@mail.gmail.com>
	<BANLkTikDGdgqmPb_kYxebpf0K-U838oFUw@mail.gmail.com>
	<BANLkTik_UHtC-1a+FL_hrKuLLVjxC_gdow@mail.gmail.com>
Message-ID: <BANLkTi=k0kRyQzZy0sDP06L_-ReeaonrRw@mail.gmail.com>

I would prefer ATA, which (less expensive) ATA would you recommend? which
works good with FS?

On Tue, Apr 5, 2011 at 8:16 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> And remember to use PCMU (G711) or T.38 on the server to send faxes...
>
> On Tue, Apr 5, 2011 at 11:12 AM, William Suffill
> <william.suffill at gmail.com> wrote:
> > The HP doesn't know anything about voip. You would have to connect via an
> > phone cable to an ATA which would handle the analog to voip for you.
> Another
> > option would be to just scan with the HP and use FS to Fax the images.
> > -- W
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From randy.andrade at gmail.com  Tue Apr  5 22:31:18 2011
From: randy.andrade at gmail.com (Randy Andrade)
Date: Tue, 5 Apr 2011 14:31:18 -0400
Subject: [Freeswitch-users] Connecting HP 8500 All in One FAX Machine to
	FS
In-Reply-To: <BANLkTi=k0kRyQzZy0sDP06L_-ReeaonrRw@mail.gmail.com>
References: <BANLkTikawF1e2UX4-AparihLMYi=JQ7ZcA@mail.gmail.com>
	<BANLkTikDGdgqmPb_kYxebpf0K-U838oFUw@mail.gmail.com>
	<BANLkTik_UHtC-1a+FL_hrKuLLVjxC_gdow@mail.gmail.com>
	<BANLkTi=k0kRyQzZy0sDP06L_-ReeaonrRw@mail.gmail.com>
Message-ID: <BANLkTi=RUBFSkNm7dphLNpT-SZzWMVdpdg@mail.gmail.com>

I personally use a Linksys SPA2102 with a Brother MFC-490CW all-in-one for
faxing with great success.

On Tue, Apr 5, 2011 at 2:24 PM, Nice Voip <nicevoip at googlemail.com> wrote:

> I would prefer ATA, which (less expensive) ATA would you recommend? which
> works good with FS?
>
>
> On Tue, Apr 5, 2011 at 8:16 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:
>
>> And remember to use PCMU (G711) or T.38 on the server to send faxes...
>>
>> On Tue, Apr 5, 2011 at 11:12 AM, William Suffill
>> <william.suffill at gmail.com> wrote:
>> > The HP doesn't know anything about voip. You would have to connect via
>> an
>> > phone cable to an ATA which would handle the analog to voip for you.
>> Another
>> > option would be to just scan with the HP and use FS to Fax the images.
>> > -- W
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Apr  5 23:09:46 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 5 Apr 2011 12:09:46 -0700
Subject: [Freeswitch-users] Transfer attempt for a previously a replaced
 call fails
In-Reply-To: <4D9AADEB.7040803@opensipstack.org>
References: <4D9A9A42.2070804@opensipstack.org>
	<BANLkTimsX4oJStWYDvz2yhPvf1e9F7kezw@mail.gmail.com>
	<4D9AADEB.7040803@opensipstack.org>
Message-ID: <BANLkTi=XdyGKZcLmHmqe1_kupp5n7ib6KQ@mail.gmail.com>

I'll have to defer to those more experienced than I in such matters.
However, I can offer two tips:

#1 - turn off the crazy sofia debugging - it's just noise. All you need to
do to enable SIP trace is "sofia global siptrace on"
#2 - when you pastebin the console output use the FreeSWITCH log syntax
highlighting - it makes it *much* easier to see what's going on.

-MC

On Mon, Apr 4, 2011 at 10:51 PM, Joegen E. Baclor
<joegen at opensipstack.org>wrote:

>  Hi Michael,
>
> I have pasted both working and none working logs on pastebin.
>
> FreeSWITCH Version 1.0.7 (hacked-20110326T123355Z)
> working:  http://pastebin.freeswitch.org/16008
> not working:  http://pastebin.freeswitch.org/16009
>
> The call flow for the working call is
> UA1 ->  (FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp)
> FSIVRApp knows the uuid of the bridge call.  Pressing # on the IVR results
> to a uuid_deflect on the bridged channel.  This works and call successfully
> transfers to the new destination.
>
> The call flow for the none working call is
>
> 1.  UA1 -> UA2  is in conversation
> 2.  UA1 puts UA2 on hold
>
> -- start of FS interaction here --
>
> 3.  UA1 ->  (FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp)  (on line 2)
> 4.  UA1 sends REFER (replacing its call with UA2) to FSBridgeDialPlan.
> 5.  Flow is now UA2 ->  ([REPLACED]FSBridgeDialPlan) -> (SIP-Loopback) ->
> (FSIVRApp)
> 6.  UA2 presses #.
> 7.  IVRApp performs uuid_deflect on FSBridgeDialPlan.
> 8. FSBridgeDialPlan drops call (no REFER is done)
>
> Thanks for your help.
>
> Joegen
>
>
> On 04/05/2011 12:35 PM, Michael Collins wrote:
>
> What do you see on the console when you try this? A console debug log with
> siptrace would go a long way toward figuring out what is happening.
>
>  -MC
>
> On Mon, Apr 4, 2011 at 9:27 PM, Joegen E. Baclor <joegen at opensipstack.org>wrote:
>
>> Hi List,
>>
>> I have a scenario where a bridged call has been replaced due to a
>> consultative transfer.  This works pretty well and audio is
>> bidirectional.  I have the original uuid of the call in a var
>> somewhere.  The trouble begins when I uuid_deflect the bridged call once
>> again to attempt another transfer.  Sofia disconnects the channel.  I am
>> using the original uuid of the call (uuid prior to replaces).  Is this
>> the right way of doing it?
>>
>> Joegen
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> 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
>
>
>
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From kris at kriskinc.com  Tue Apr  5 23:20:39 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 5 Apr 2011 15:20:39 -0400
Subject: [Freeswitch-users] Hylafax server emulation
In-Reply-To: <4D9B42D7.4020008@coppice.org>
References: <4D9B42D7.4020008@coppice.org>
Message-ID: <BANLkTinaftq2FapGp4tM3Ysjr1_Wd8gGbw@mail.gmail.com>

Steve,

  Very cool (and I'm very interested).

  What about using the existing user directory (perhaps with
additional params) for the accounts and the FreeSWITCH core DB
(whether SQLite or ODBC) for the jobs, etc?

On Tue, Apr 5, 2011 at 12:27 PM, Steve Underwood <steveu at coppice.org> wrote:
> Hi,
>
> It has always been clear that a HylaFAX compatible FAX job submission
> server would add considerably to the value of the FAX facilities in
> Asterisk and Freeswitch, but somehow it hasn't happened until now. I
> recently found that in 2005 someone produced something fairly basic for
> Asterisk in Perl, but it doesn't seem to have been well publicised, and
> it looks like development stalled long ago.
>
> I now have the skeleton of HylaFAX compatible FAX job submission server,
> in C, working. It accepts FAX submissions from sendfax and a couple of
> the windows HylaFAX clients, though it needs a lot more polishing. Now I
> need to look at the best thing to do on the Freeswitch side. I aim to
> make the server maintain its own database of FAX jobs. It will attach to
> Freeswitch, by ESL; push the jobs through FS; deal with scheduling,
> retries, etc; and report the final result to the user, just as HylaFAX
> does. The thing I am rather unsure about is the best way to handle the
> accounts used to accept FAX jobs? Should I maintain a separate database
> of FAX accounts, or hook into an existing database? I would welcome
> suggestions for what would be the most useful approach.
>
> 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
>



-- 
Kristian Kielhofner


From nick.rosier at gmail.com  Wed Apr  6 00:06:09 2011
From: nick.rosier at gmail.com (Nick Rosier)
Date: Tue, 5 Apr 2011 22:06:09 +0200
Subject: [Freeswitch-users] Gateway with dynamic IP address
In-Reply-To: <AANLkTinZWWijJBBWr0ArbtSGjHuv1cpHDDxKiZqj4Agg@mail.gmail.com>
References: <C9C8E1CC-4621-44FF-8D19-D8DB35D41530@gmail.com>
	<AANLkTinQNJpc0UP4hSjnSaa5zqcqgVoLhV0ScrPDvwNN@mail.gmail.com>
	<828493E7-A5E7-4896-844F-271AB72AD38B@gmail.com>
	<AANLkTinZWWijJBBWr0ArbtSGjHuv1cpHDDxKiZqj4Agg@mail.gmail.com>
Message-ID: <BANLkTik1B7XhyCnDe2zs3Pqob4MD4evaPQ@mail.gmail.com>

Has anyone been able to get this working? I'm still stuck; everything
is working except outbound dialing through the "gateways".

N.

On 1 April 2011 08:41, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> You can create usual user in directory, it will register with FS, and then
> you can dial it with arbitrary number, getting its host/port using
> sofia_contact and constructing request URI you need.
>
> 2011/4/1 Juan Wajnerman <juan.wajnerman at gmail.com>
>>
>> That's exactly what I don't want to set: a static IP address for the
>> gateway. In other words I'd like to use a "user" as if it were a gateway. Is
>> that even possible in FreeSwitch?
>>
>> On Apr 1, 2011, at 1:41 AM, Michael Collins <msc at freeswitch.org> wrote:
>>
>>
>>
>> On Thu, Mar 31, 2011 at 6:25 AM, Juan Wajnerman <juan.wajnerman at gmail.com>
>> wrote:
>>>
>>> I asked this question yesterday in the IRC but I couldn't get a solution.
>>> I'd like to have a gateway configured in FreeSwitch without specifying
>>> the static IP address.
>>> I have this configuration:
>>>
>>> <user id="gw">
>>> ? <gateways>
>>> ? ? <gateway name="gw">
>>> ? ? ? <param name="username" value="gw"/>
>>> ? ? ? <param name="password" value="password"/>
>>> ? ? ? <param name="register" value="false"/>
>>> ? ? ? <param name="extension" value="2"/>
>>> ? ? </gateway>
>>> ? </gateways>
>>> ? <params>
>>> ? ? <param name="password" value="password"/>
>>> ? </params>
>>> ? <variables>
>>> ? ? <variable name="user_context" value="default" />
>>> ? </variables>
>>> ?</user>
>>>
>>> and the SIP device is registering properly, but I cannot dial with
>>> addresses like: "sofia/gateway/gw/123456789".
>>> Note that this works if the gateway name is the IP address or host name,
>>> or if I add a "proxy" setting with the IP address.
>>
>> You haven't set the realm parameter. Look at the example.com.xml file in
>> conf/sip_profiles/external/ and you'll see in the comments that if you don't
>> set the realm param then it goes to the name of the gateway. Set the realm
>> to the target IP or host name and try again.
>> -MC
>>
>>>
>>> I have a similar configuration in asterisk, where the sip.conf contains:
>>>
>>> [gw]
>>> type=friend
>>> secret=password
>>> context=default
>>> host=dynamic
>>>
>>> And once the gateway is registered in asterisk, I can dial with
>>> "SIP/gw/123456789".
>>> Is there any way to make a similar configuration in FreeSwitch?
>>>
>>> Thanks!
>>> - Juan
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Wed Apr  6 00:29:49 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Tue, 5 Apr 2011 16:29:49 -0400
Subject: [Freeswitch-users] Gateway with dynamic IP address
In-Reply-To: <BANLkTik1B7XhyCnDe2zs3Pqob4MD4evaPQ@mail.gmail.com>
References: <C9C8E1CC-4621-44FF-8D19-D8DB35D41530@gmail.com>
	<AANLkTinZWWijJBBWr0ArbtSGjHuv1cpHDDxKiZqj4Agg@mail.gmail.com>
	<BANLkTik1B7XhyCnDe2zs3Pqob4MD4evaPQ@mail.gmail.com>
Message-ID: <201104051629.49864.sos@sokhapkin.dyndns.org>

"Gateway" with dynamic IP address makes no sense to me... Maybe I'm mistaken?

On Tuesday 05 April 2011, Nick Rosier wrote:
> Has anyone been able to get this working? I'm still stuck; everything
> is working except outbound dialing through the "gateways".
> 
> N.
> 
> On 1 April 2011 08:41, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> > You can create usual user in directory, it will register with FS, and
> > then you can dial it with arbitrary number, getting its host/port using
> > sofia_contact and constructing request URI you need.
> > 
> > 2011/4/1 Juan Wajnerman <juan.wajnerman at gmail.com>
> > 
> >> That's exactly what I don't want to set: a static IP address for the
> >> gateway. In other words I'd like to use a "user" as if it were a
> >> gateway. Is that even possible in FreeSwitch?
> >> 
> >> On Apr 1, 2011, at 1:41 AM, Michael Collins <msc at freeswitch.org> wrote:
> >> 
> >> 
> >> 
> >> On Thu, Mar 31, 2011 at 6:25 AM, Juan Wajnerman
> >> <juan.wajnerman at gmail.com>
> >> 
> >> wrote:
> >>> I asked this question yesterday in the IRC but I couldn't get a
> >>> solution. I'd like to have a gateway configured in FreeSwitch without
> >>> specifying the static IP address.
> >>> I have this configuration:
> >>> 
> >>> <user id="gw">
> >>>   <gateways>
> >>>     <gateway name="gw">
> >>>       <param name="username" value="gw"/>
> >>>       <param name="password" value="password"/>
> >>>       <param name="register" value="false"/>
> >>>       <param name="extension" value="2"/>
> >>>     </gateway>
> >>>   </gateways>
> >>>   <params>
> >>>     <param name="password" value="password"/>
> >>>   </params>
> >>>   <variables>
> >>>     <variable name="user_context" value="default" />
> >>>   </variables>
> >>>  </user>
> >>> 
> >>> and the SIP device is registering properly, but I cannot dial with
> >>> addresses like: "sofia/gateway/gw/123456789".
> >>> Note that this works if the gateway name is the IP address or host
> >>> name, or if I add a "proxy" setting with the IP address.
> >> 
> >> You haven't set the realm parameter. Look at the example.com.xml file in
> >> conf/sip_profiles/external/ and you'll see in the comments that if you
> >> don't set the realm param then it goes to the name of the gateway. Set
> >> the realm to the target IP or host name and try again.
> >> -MC
> >> 
> >>> I have a similar configuration in asterisk, where the sip.conf
> >>> contains:
> >>> 
> >>> [gw]
> >>> type=friend
> >>> secret=password
> >>> context=default
> >>> host=dynamic
> >>> 
> >>> And once the gateway is registered in asterisk, I can dial with
> >>> "SIP/gw/123456789".
> >>> Is there any way to make a similar configuration in FreeSwitch?
> >>> 
> >>> Thanks!
> >>> - Juan
> >>> _______________________________________________
> >>> FreeSWITCH-users 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
> >> 
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> > 
> > --
> > Best regards,
> > 
> > Dmitry Sytchev,
> > IT Engineer
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Wed Apr  6 00:45:26 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 5 Apr 2011 22:45:26 +0200
Subject: [Freeswitch-users] Gateway with dynamic IP address
In-Reply-To: <BANLkTik1B7XhyCnDe2zs3Pqob4MD4evaPQ@mail.gmail.com>
References: <C9C8E1CC-4621-44FF-8D19-D8DB35D41530@gmail.com>
	<AANLkTinQNJpc0UP4hSjnSaa5zqcqgVoLhV0ScrPDvwNN@mail.gmail.com>
	<828493E7-A5E7-4896-844F-271AB72AD38B@gmail.com>
	<AANLkTinZWWijJBBWr0ArbtSGjHuv1cpHDDxKiZqj4Agg@mail.gmail.com>,
	<BANLkTik1B7XhyCnDe2zs3Pqob4MD4evaPQ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C43A41E5@cooper>

What you wan't to do is to add a user. Then you dial this user, which by then is registered in FreeSWITCH, and it will find the path.

So no gateway in this case, it's when you want to register to an external server, a user is when someone registers to you, and you wan't to be able to dial outside through this.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Nick Rosier [nick.rosier at gmail.com]
Skickat: den 5 april 2011 22:06
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Gateway with dynamic IP address

Has anyone been able to get this working? I'm still stuck; everything
is working except outbound dialing through the "gateways".

N.

On 1 April 2011 08:41, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> You can create usual user in directory, it will register with FS, and then
> you can dial it with arbitrary number, getting its host/port using
> sofia_contact and constructing request URI you need.
>
> 2011/4/1 Juan Wajnerman <juan.wajnerman at gmail.com>
>>
>> That's exactly what I don't want to set: a static IP address for the
>> gateway. In other words I'd like to use a "user" as if it were a gateway. Is
>> that even possible in FreeSwitch?
>>
>> On Apr 1, 2011, at 1:41 AM, Michael Collins <msc at freeswitch.org> wrote:
>>
>>
>>
>> On Thu, Mar 31, 2011 at 6:25 AM, Juan Wajnerman <juan.wajnerman at gmail.com>
>> wrote:
>>>
>>> I asked this question yesterday in the IRC but I couldn't get a solution.
>>> I'd like to have a gateway configured in FreeSwitch without specifying
>>> the static IP address.
>>> I have this configuration:
>>>
>>> <user id="gw">
>>>   <gateways>
>>>     <gateway name="gw">
>>>       <param name="username" value="gw"/>
>>>       <param name="password" value="password"/>
>>>       <param name="register" value="false"/>
>>>       <param name="extension" value="2"/>
>>>     </gateway>
>>>   </gateways>
>>>   <params>
>>>     <param name="password" value="password"/>
>>>   </params>
>>>   <variables>
>>>     <variable name="user_context" value="default" />
>>>   </variables>
>>>  </user>
>>>
>>> and the SIP device is registering properly, but I cannot dial with
>>> addresses like: "sofia/gateway/gw/123456789".
>>> Note that this works if the gateway name is the IP address or host name,
>>> or if I add a "proxy" setting with the IP address.
>>
>> You haven't set the realm parameter. Look at the example.com.xml file in
>> conf/sip_profiles/external/ and you'll see in the comments that if you don't
>> set the realm param then it goes to the name of the gateway. Set the realm
>> to the target IP or host name and try again.
>> -MC
>>
>>>
>>> I have a similar configuration in asterisk, where the sip.conf contains:
>>>
>>> [gw]
>>> type=friend
>>> secret=password
>>> context=default
>>> host=dynamic
>>>
>>> And once the gateway is registered in asterisk, I can dial with
>>> "SIP/gw/123456789".
>>> Is there any way to make a similar configuration in FreeSwitch?
>>>
>>> Thanks!
>>> - Juan
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4d9b7ad932761210261109!



From sc_zhangming at sina.com  Wed Apr  6 06:13:50 2011
From: sc_zhangming at sina.com (=?gb2312?B?1cXD9w==?=)
Date: Wed, 6 Apr 2011 10:13:50 +0800
Subject: [Freeswitch-users] uuid_hold is not send hold message
Message-ID: <8slt5a$arb4s0@irxd5-187.sinamail.sina.com.cn>

Michael Collins???? 

??use step, freeswitch version 1.07
        1.   fs_cli 
        2.  /event plain CHANNEL_HOLD CHANNEL_UNHOLD
        3. show  calls 
        freeswitch at 10.108.226.220@internal> show calls
call_uuid,call_created,call_created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid,hostname
b0a7bdfd-430d-4746-91a4-8c318f9e1695,2011-04-08 04:22:37,1302207757,switch_ivr_multi_threaded_bridge,700000,700000,700001,sofia/internal/700000 at 10.108.226.220,b0a7bdfd-430d-4746-91a4-8c318f9e1695,Outbound Call,700001,700001,sofia/internal/sip:700001 at 10.108.226.44:18692,e2faae72-ef43-4019-b7e6-1120d7e42e1e,localhost.localdomain

1 total.
        4.   uuid_hold  b0a7bdfd-430d-4746-91a4-8c318f9e1695
            2011-04-08 04:27:55.539115 [DEBUG] switch_channel.c:1373 (sofia/internal/700000 at 10.108.226.220) Callstate Change ACTIVE -> HELD
2011-04-08 04:27:55.539115 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/700000 at 10.108.226.220 [BREAK]
2011-04-08 04:27:55.539115 [DEBUG] switch_ivr.c:1272 ??????
2011-04-08 04:27:55.539115 [DEBUG] switch_core_session.c:954 Send signal sofia/internal/sip:700001 at 10.108.226.44:18692 [BREAK]
2011-04-08 04:27:55.539115 [DEBUG] sofia.c:4646 Channel sofia/internal/700000 at 10.108.226.220 entering state [calling][0]
2011-04-08 04:27:55.544824 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/sip:700001 at 10.108.226.44:18692 [BREAK]
2011-04-08 04:27:55.544824 [INFO] sofia.c:729 sofia/internal/700000 at 10.108.226.220 Update Callee ID to "Outbound Call" <700000>
2011-04-08 04:27:55.547789 [DEBUG] sofia.c:4646 Channel sofia/internal/700000 at 10.108.226.220 entering state [ready][200]
2011-04-08 04:27:55.547789 [DEBUG] sofia.c:4654 Duplicate SDP
v=0
o=Idefisk_user 47685 6056184806875839522 IN IP4 10.108.226.44
s=Idefisk_user
c=IN IP4 10.108.226.44
t=0 0
m=audio 8000 RTP/AVP 8 97 0 110 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000]
2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000]
2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000]
2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:2690 Already using PCMA
2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4571 Set 2833 dtmf send/recv payload to 101
2011-04-08 04:27:55.725181 [DEBUG] switch_ivr.c:563 sofia/internal/sip:700001 at 10.108.226.44:18692 Command Execute playback(local_stream://moh)
EXECUTE sofia/internal/sip:700001 at 10.108.226.44:18692 playback(local_stream://moh)
2011-04-08 04:27:55.725181 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz
2011-04-08 04:27:55.725181 [DEBUG] switch_ivr_play_say.c:1236 Codec Activated L16 at 8000hz 1 channels 20ms

        5.  it 's not return  hold message, if  user  Zoiper  hold button  freeswitch  return hold message 

            Zoiper hold  freeswich return message:
            2011-04-08 04:34:51.330157 [DEBUG] sofia.c:4646 Channel sofia/internal/700000 at 10.108.226.220 entering state [received][100]
RECV EVENT
Event-Name: CHANNEL_HOLD
Core-UUID: 5e94c1c9-305d-4c4d-b00c-ea24f6346559
FreeSWITCH-Hostname: localhost.localdomain
FreeSWITCH-IPv4: 10.108.226.220
FreeSWITCH-IPv6: ::1
Event-Date-Local: 2011-04-08 04:34:51
Event-Date-GMT: Thu, 07 Apr 2011 20:34:51 GMT
Event-Date-Timestamp: 1302208491330157
Event-Calling-File: switch_channel.c
Event-Calling-Function: switch_channel_mark_hold
Event-Calling-Line-Number: 642
Channel-State: CS_EXECUTE
Channel-Call-State: HELD
Channel-State-Number: 4
Channel-Name: sofia/internal/700000 at 10.108.226.220
Unique-ID: eb1222c9-4175-469a-8549-667c0dd921a6
Call-Direction: inbound
Presence-Call-Direction: inbound
Channel-Presence-ID: 700000 at 10.108.226.220
Channel-Call-UUID: eb1222c9-4175-469a-8549-667c0dd921a6
Answer-State: answered
Channel-Read-Codec-Name: PCMA
Channel-Read-Codec-Rate: 8000
Channel-Read-Codec-Bit-Rate: 64000
Channel-Write-Codec-Name: PCMA
Channel-Write-Codec-Rate: 8000
Channel-Write-Codec-Bit-Rate: 64000
Caller-Direction: inbound
Caller-Username: 700000
Caller-Dialplan: XML
Caller-Caller-ID-Name: 700000
Caller-Caller-ID-Number: 700000
Caller-Callee-ID-Name: Outbound Call
Caller-Callee-ID-Number: 700001
Caller-Network-Addr: 10.108.226.44
Caller-ANI: 700000
Caller-Destination-Number: 700001
Caller-Unique-ID: eb1222c9-4175-469a-8549-667c0dd921a6
Caller-Source: mod_sofia
Caller-Context: default
Caller-Channel-Name: sofia/internal/700000 at 10.108.226.220
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1302208456152936
Caller-Channel-Created-Time: 1302208456152936
Caller-Channel-Answered-Time: 1302208463040930
Caller-Channel-Progress-Time: 1302208456340619
Caller-Channel-Progress-Media-Time: 0
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
Other-Type: originatee
Other-Leg-Direction: outbound
Other-Leg-Username: 700000
Other-Leg-Dialplan: XML
Other-Leg-Caller-ID-Name: 700000
Other-Leg-Caller-ID-Number: 700000
Other-Leg-Callee-ID-Name: Outbound Call
Other-Leg-Callee-ID-Number: 700001
Other-Leg-Network-Addr: 10.108.226.44
Other-Leg-ANI: 700000
Other-Leg-Destination-Number: 700001
Other-Leg-Unique-ID: 1595a55f-d97c-47c1-aa6c-42c8efcb5b27
Other-Leg-Source: mod_sofia
Other-Leg-Context: default
Other-Leg-Channel-Name: sofia/internal/sip:700001 at 10.108.226.44:18692
Other-Leg-Screen-Bit: true
Other-Leg-Privacy-Hide-Name: false
Other-Leg-Privacy-Hide-Number: false


2011-04-08 04:34:51.330157 [DEBUG] sofia.c:4657 Remote SDP:
v=0
o=Idefisk_user 6056184806875821816 6056184806875816806 IN IP4 10.108.226.44
s=Idefisk_user
c=IN IP4 0.0.0.0
t=0 0
m=audio 8000 RTP/AVP 8 97 0 110 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendonly

2011-04-08 04:34:51.330157 [DEBUG] switch_channel.c:1373 (sofia/internal/700000 at 10.108.226.220) Callstate Change ACTIVE -> HELD
2011-04-08 04:34:51.330157 [DEBUG] switch_core_session.c:954 Send signal sofia/internal/sip:700001 at 10.108.226.44:18692 [BREAK]
2011-04-08 04:34:51.346542 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/sip:700001 at 10.108.226.44:18692 [BREAK]
2011-04-08 04:34:51.524112 [DEBUG] switch_ivr.c:563 sofia/internal/sip:700001 at 10.108.226.44:18692 Command Execute playback(local_stream://moh)
EXECUTE sofia/internal/sip:700001 at 10.108.226.44:18692 playback(local_stream://moh)
2011-04-08 04:34:51.524112 [DEBUG] mod_local_stream.c:421 Opening Stream [moh/8000] 8000hz
2011-04-08 04:34:51.524112 [DEBUG] switch_ivr_play_say.c:1236 Codec Activated L16 at 8000hz 1 channels 20ms
2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000]
2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000]
2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000]
2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:2690 Already using PCMA
2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4571 Set 2833 dtmf send/recv payload to 101
2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:2976 Audio params changed for sofia/internal/700000 at 10.108.226.220 from 10.108.226.44:8000 to 0.0.0.0:8000
2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:2987 AUDIO RTP [sofia/internal/700000 at 10.108.226.220] 10.108.226.220 port 19022 -> 0.0.0.0 port 8000 codec: 8 ms: 20
2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:3017 AUDIO RTP CHANGING DEST TO: [0.0.0.0:8000]
2011-04-08 04:34:51.580870 [DEBUG] sofia.c:5057 Processing updated SDP
2011-04-08 04:34:51.583974 [DEBUG] sofia.c:4646 Channel sofia/internal/700000 at 10.108.226.220 entering state [completed][200]
2011-04-08 04:34:51.583974 [DEBUG] sofia.c:4646 Channel sofia/internal/700000 at 10.108.226.220 entering state [ready][200]


uuid_hold  command user problem:
        1.  uuid_hold   uuid  is error still return OK
        2.  uuid_hold   <uuid> OR  uuid_hold off  <uuid>   freeswitch is not return  hold or unhold message 


======== 2011-04-05 03:23:09 ???????? ========

I just tried this on latest git and it worked fine for me. Can you pastebin the console debug output when you use it?
-MC


2011/4/2 ?? <sc_zhangming at sina.com>

freeswitch-users????

       uuid_hold command , freeswitch is not send HOLD  message. who know it.

?????????
??


??????????
????????sc_zhangming at sina.com
??????????2011-04-02
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




= = = = = = = = = = = = = = = = = = = = = = 
?????????
??

????????????????
??????????????sc_zhangming at sina.com
???????????????2011-04-06
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From frank at telonium.com  Wed Apr  6 06:41:42 2011
From: frank at telonium.com (Frank Park)
Date: Tue, 5 Apr 2011 22:41:42 -0400
Subject: [Freeswitch-users] xml_curl response for voicemail_inject
In-Reply-To: <BANLkTinzMqjawEEduXjV4bWNJE=n14p_gA@mail.gmail.com>
References: <BANLkTi=AwsfhJw7j6yLBBmy7hAUwVegkbA@mail.gmail.com>
	<BANLkTikvfBZEYeA=UsoKRhXOJdrnMVgJGQ@mail.gmail.com>
	<BANLkTimCsmMSagdMQBPhv=DQo0_ChVNMbA@mail.gmail.com>
	<BANLkTik+OyUkwX6i3k+9i9HYpRuPg7AVqA@mail.gmail.com>
	<BANLkTi=CHYXEE=hQmM9qE16U96u=7mMPdw@mail.gmail.com>
	<BANLkTinwdLPAYawgi4Z0a4nTL42_dkYVxA@mail.gmail.com>
	<BANLkTinzMqjawEEduXjV4bWNJE=n14p_gA@mail.gmail.com>
Message-ID: <BANLkTini6Q3AdNNPhw72XNCz5N+wKU6iGA@mail.gmail.com>

Yeah.. I would like to resolve this issue by actually fixing it, but this
would be the temporary fix for me... Thanks!

Frank

On Tue, Apr 5, 2011 at 11:21 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Apr 5, 2011 at 7:08 AM, Frank Park <frank at telonium.com> wrote:
>
>> Is there a way to disable this option in the VM prompt until I can fix
>> this issue? I didn't see the option in the voicemail.conf.xml
>>
>> If you are just wanting to disable to voicing of the option then you need
> to look in conf/lang/en/vm/sounds.xml. Find the phrase macro that voices the
> caller options and comment out the one that says to press x to forward the
> message.
>
> -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
>
>


-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From dunchan at freemail.hu  Wed Apr  6 09:27:46 2011
From: dunchan at freemail.hu (dunchan)
Date: Wed, 06 Apr 2011 07:27:46 +0200
Subject: [Freeswitch-users] FS and Phone are in the same IP
Message-ID: <4D9BF9D2.8010401@freemail.hu>

Hi!

My voip phone doesn't get the SIP bye request from the called party (via 
PSTN gateway)

Phone is in the same machine with FS, IP addr is same too, Contact 
header is filled corretly.

If phone is in other machine everything works fine.

Should I change the NAT handling somewhere, or what?

thanks,
Viktor


From pablosaro at gmail.com  Wed Apr  6 09:58:23 2011
From: pablosaro at gmail.com (Pablo Hernan Saro)
Date: Wed, 6 Apr 2011 02:58:23 -0300
Subject: [Freeswitch-users] Hylafax server emulation
In-Reply-To: <BANLkTinaftq2FapGp4tM3Ysjr1_Wd8gGbw@mail.gmail.com>
References: <4D9B42D7.4020008@coppice.org>
	<BANLkTinaftq2FapGp4tM3Ysjr1_Wd8gGbw@mail.gmail.com>
Message-ID: <BANLkTimXgW1FNMFtFwTHGQ_sgtXNEiDwqw@mail.gmail.com>

Hi Steve,

I'm not sure if the following will work. Let's say that you have a SIP trunk
service configured in a FS box for interconnection with the PSTN. To avoid
individual analog fax machines (reduce hardware costs and attached
maintenance costs) and take advantage of fax server capabilities, a solution
would be install HylaFAX with t38modem built with OPAL support (this enables
routing modems to SIP URIs) sending/receiving calls to/from FS via SIP
(configure your dialplan for enabling t38 at FS side). That way you get the
best from HylaFAX (jub submission, scheduling, retrying, reporting and other
facilities) and FS (dealing with SIP trunk service providers, cdr, call
routing) at the same time.
End users can interact with HylaFAX using compatible windows clients, email
interface or web interface. A very good idea would be link HylaFAX accounts
to FS directory (connect HylaFAX to FS db may be?).
My two cents...


On Tue, Apr 5, 2011 at 4:20 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Steve,
>
>  Very cool (and I'm very interested).
>
>  What about using the existing user directory (perhaps with
> additional params) for the accounts and the FreeSWITCH core DB
> (whether SQLite or ODBC) for the jobs, etc?
>
> On Tue, Apr 5, 2011 at 12:27 PM, Steve Underwood <steveu at coppice.org>
> wrote:
> > Hi,
> >
> > It has always been clear that a HylaFAX compatible FAX job submission
> > server would add considerably to the value of the FAX facilities in
> > Asterisk and Freeswitch, but somehow it hasn't happened until now. I
> > recently found that in 2005 someone produced something fairly basic for
> > Asterisk in Perl, but it doesn't seem to have been well publicised, and
> > it looks like development stalled long ago.
> >
> > I now have the skeleton of HylaFAX compatible FAX job submission server,
> > in C, working. It accepts FAX submissions from sendfax and a couple of
> > the windows HylaFAX clients, though it needs a lot more polishing. Now I
> > need to look at the best thing to do on the Freeswitch side. I aim to
> > make the server maintain its own database of FAX jobs. It will attach to
> > Freeswitch, by ESL; push the jobs through FS; deal with scheduling,
> > retries, etc; and report the final result to the user, just as HylaFAX
> > does. The thing I am rather unsure about is the best way to handle the
> > accounts used to accept FAX jobs? Should I maintain a separate database
> > of FAX accounts, or hook into an existing database? I would welcome
> > suggestions for what would be the most useful approach.
> >
> > 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
> >
>
>
>
> --
> Kristian Kielhofner
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fs-list at communicatefreely.net  Wed Apr  6 05:17:55 2011
From: fs-list at communicatefreely.net (Tim St. Pierre)
Date: Tue, 05 Apr 2011 21:17:55 -0400
Subject: [Freeswitch-users] xml_curl response for voicemail_inject
In-Reply-To: <BANLkTikvfBZEYeA=UsoKRhXOJdrnMVgJGQ@mail.gmail.com>
References: <BANLkTi=AwsfhJw7j6yLBBmy7hAUwVegkbA@mail.gmail.com>
	<BANLkTikvfBZEYeA=UsoKRhXOJdrnMVgJGQ@mail.gmail.com>
Message-ID: <4D9BBF43.7050407@communicatefreely.net>

I'm having the same problem.

I'm returning a complete directory any time it's asked for, but I don't 
see FS requesting anything here.

There is a request when it starts playing the message, but when I choose 
the forwarding option and enter an extension, I don't see any other 
directory requests.

On the console, I get

2011-04-05 21:11:20.271219 [ERR] mod_voicemail.c:2767 Can't find profile
2011-04-05 21:11:20.271219 [ERR] mod_voicemail.c:1550 Failed to Carbon 
Copy to 
5109                                                                       

Extension 5109 is the extension I was trying to forward to, and it's in 
the same domain as the extension I'm checking voice mail on.

Why is it looking for the profile?  I would expect FS to do a directory 
lookup on the extension number that I entered, but that doesn't seem to 
be happening.  Any ideas?




From joegen at opensipstack.org  Wed Apr  6 04:02:10 2011
From: joegen at opensipstack.org (Joegen E. Baclor)
Date: Wed, 06 Apr 2011 08:02:10 +0800
Subject: [Freeswitch-users] Transfer attempt for a previously a replaced
 call fails
In-Reply-To: <BANLkTi=XdyGKZcLmHmqe1_kupp5n7ib6KQ@mail.gmail.com>
References: <4D9A9A42.2070804@opensipstack.org>	<BANLkTimsX4oJStWYDvz2yhPvf1e9F7kezw@mail.gmail.com>	<4D9AADEB.7040803@opensipstack.org>
	<BANLkTi=XdyGKZcLmHmqe1_kupp5n7ib6KQ@mail.gmail.com>
Message-ID: <4D9BAD82.3000205@opensipstack.org>

I'll keep that in mind.  If more information is needed to get into the 
bottom of this, I will happily oblige.  Thanks for helping.

On 04/06/2011 03:09 AM, Michael Collins wrote:
> I'll have to defer to those more experienced than I in such matters. 
> However, I can offer two tips:
>
> #1 - turn off the crazy sofia debugging - it's just noise. All you 
> need to do to enable SIP trace is "sofia global siptrace on"
> #2 - when you pastebin the console output use the FreeSWITCH log 
> syntax highlighting - it makes it *much* easier to see what's going on.
>
> -MC
>
> On Mon, Apr 4, 2011 at 10:51 PM, Joegen E. Baclor 
> <joegen at opensipstack.org <mailto:joegen at opensipstack.org>> wrote:
>
>     Hi Michael,
>
>     I have pasted both working and none working logs on pastebin.
>
>     FreeSWITCH Version 1.0.7 (hacked-20110326T123355Z)
>     working: http://pastebin.freeswitch.org/16008
>     not working: http://pastebin.freeswitch.org/16009
>
>     The call flow for the working call is
>     UA1 ->  (FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp)
>     FSIVRApp knows the uuid of the bridge call.  Pressing # on the IVR
>     results to a uuid_deflect on the bridged channel.  This works and
>     call successfully transfers to the new destination.
>
>     The call flow for the none working call is
>
>     1.  UA1 -> UA2  is in conversation
>     2.  UA1 puts UA2 on hold
>
>     -- start of FS interaction here --
>
>     3.  UA1 ->  (FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp) 
>     (on line 2)
>     4.  UA1 sends REFER (replacing its call with UA2) to FSBridgeDialPlan.
>     5.  Flow is now UA2 ->  ([REPLACED]FSBridgeDialPlan) ->
>     (SIP-Loopback) -> (FSIVRApp)
>     6.  UA2 presses #.
>     7.  IVRApp performs uuid_deflect on FSBridgeDialPlan.
>     8. FSBridgeDialPlan drops call (no REFER is done)
>
>     Thanks for your help.
>
>     Joegen
>
>
>     On 04/05/2011 12:35 PM, Michael Collins wrote:
>>     What do you see on the console when you try this? A console debug
>>     log with siptrace would go a long way toward figuring out what is
>>     happening.
>>
>>     -MC
>>
>>     On Mon, Apr 4, 2011 at 9:27 PM, Joegen E. Baclor
>>     <joegen at opensipstack.org <mailto:joegen at opensipstack.org>> wrote:
>>
>>         Hi List,
>>
>>         I have a scenario where a bridged call has been replaced due to a
>>         consultative transfer.  This works pretty well and audio is
>>         bidirectional.  I have the original uuid of the call in a var
>>         somewhere.  The trouble begins when I uuid_deflect the
>>         bridged call once
>>         again to attempt another transfer.  Sofia disconnects the
>>         channel.  I am
>>         using the original uuid of the call (uuid prior to replaces).
>>          Is this
>>         the right way of doing it?
>>
>>         Joegen
>>
>>         _______________________________________________
>>         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
>
>

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From tayeb.meftah at gmail.com  Wed Apr  6 15:49:55 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Wed, 06 Apr 2011 13:49:55 +0200
Subject: [Freeswitch-users] FS and Phone are in the same IP
In-Reply-To: <4D9BF9D2.8010401@freemail.hu>
References: <4D9BF9D2.8010401@freemail.hu>
Message-ID: <4D9C5363.3060106@gmail.com>

where's nat in this path?
you say phone and fs have same ip
is this a hardphone or a softphone?
if this is a hardphone, how come a pc and a phone can fill in the  same 
network with same ip?
thanks
On 06/04/2011 07:27, dunchan wrote:
> Hi!
>
> My voip phone doesn't get the SIP bye request from the called party (via
> PSTN gateway)
>
> Phone is in the same machine with FS, IP addr is same too, Contact
> header is filled corretly.
>
> If phone is in other machine everything works fine.
>
> Should I change the NAT handling somewhere, or what?
>
> thanks,
> Viktor
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>    


-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883



From steveu at coppice.org  Wed Apr  6 16:27:49 2011
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 06 Apr 2011 20:27:49 +0800
Subject: [Freeswitch-users] Hylafax server emulation
In-Reply-To: <BANLkTimXgW1FNMFtFwTHGQ_sgtXNEiDwqw@mail.gmail.com>
References: <4D9B42D7.4020008@coppice.org>	<BANLkTinaftq2FapGp4tM3Ysjr1_Wd8gGbw@mail.gmail.com>
	<BANLkTimXgW1FNMFtFwTHGQ_sgtXNEiDwqw@mail.gmail.com>
Message-ID: <4D9C5C45.7030702@coppice.org>

Wow. That's one of the most bizarre solutions I've seen in a long time. :-\

Steve

On 04/06/2011 01:58 PM, Pablo Hernan Saro wrote:
> Hi Steve,
>
> I'm not sure if the following will work. Let's say that you have a SIP 
> trunk service configured in a FS box for interconnection with the 
> PSTN. To avoid individual analog fax machines (reduce hardware costs 
> and attached maintenance costs) and take advantage of fax server 
> capabilities, a solution would be install HylaFAX with t38modem built 
> with OPAL support (this enables routing modems to SIP URIs) 
> sending/receiving calls to/from FS via SIP (configure your dialplan 
> for enabling t38 at FS side). That way you get the best from HylaFAX 
> (jub submission, scheduling, retrying, reporting and other facilities) 
> and FS (dealing with SIP trunk service providers, cdr, call routing) 
> at the same time.
> End users can interact with HylaFAX using compatible windows clients, 
> email interface or web interface. A very good idea would be link 
> HylaFAX accounts to FS directory (connect HylaFAX to FS db may be?).
> My two cents...
>
>
> On Tue, Apr 5, 2011 at 4:20 PM, Kristian Kielhofner <kris at kriskinc.com 
> <mailto:kris at kriskinc.com>> wrote:
>
>     Steve,
>
>      Very cool (and I'm very interested).
>
>      What about using the existing user directory (perhaps with
>     additional params) for the accounts and the FreeSWITCH core DB
>     (whether SQLite or ODBC) for the jobs, etc?
>
>     On Tue, Apr 5, 2011 at 12:27 PM, Steve Underwood
>     <steveu at coppice.org <mailto:steveu at coppice.org>> wrote:
>     > Hi,
>     >
>     > It has always been clear that a HylaFAX compatible FAX job
>     submission
>     > server would add considerably to the value of the FAX facilities in
>     > Asterisk and Freeswitch, but somehow it hasn't happened until now. I
>     > recently found that in 2005 someone produced something fairly
>     basic for
>     > Asterisk in Perl, but it doesn't seem to have been well
>     publicised, and
>     > it looks like development stalled long ago.
>     >
>     > I now have the skeleton of HylaFAX compatible FAX job submission
>     server,
>     > in C, working. It accepts FAX submissions from sendfax and a
>     couple of
>     > the windows HylaFAX clients, though it needs a lot more
>     polishing. Now I
>     > need to look at the best thing to do on the Freeswitch side. I
>     aim to
>     > make the server maintain its own database of FAX jobs. It will
>     attach to
>     > Freeswitch, by ESL; push the jobs through FS; deal with scheduling,
>     > retries, etc; and report the final result to the user, just as
>     HylaFAX
>     > does. The thing I am rather unsure about is the best way to
>     handle the
>     > accounts used to accept FAX jobs? Should I maintain a separate
>     database
>     > of FAX accounts, or hook into an existing database? I would welcome
>     > suggestions for what would be the most useful approach.
>     >
>     > Steve
>     >
>



From Nabble at slickdeals.endjunk.com  Wed Apr  6 16:36:13 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 6 Apr 2011 05:36:13 -0700 (PDT)
Subject: [Freeswitch-users] SIP Proxy and DNS SRV options
Message-ID: <1302093373200-6245865.post@n2.nabble.com>

Under the SIP Settings and Proxy and Registration sections in Line 1 and Line
2 TABs of a Linksys WRTP54G VoIP router with firmware v3.1.27, I have
noticed the following options (see the attached image file below):

1. SIP Proxy-Require option in the SIP Settings section and Proxy option in
the Proxy and Registration section.

2. Use DNS SRV, DNS SRV Auto Prefix, and Proxy Redundancy Method options in
the Proxy and Registration section.

The question I have is what are the equivalent settings in FS for the above
mentioned options. Any examples will be a plus.


          
http://freeswitch-users.2379917.n2.nabble.com/file/n6245865/SIP_DNS_SRV.png 

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SIP-Proxy-and-DNS-SRV-options-tp6245865p6245865.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From dunchan at freemail.hu  Wed Apr  6 16:41:07 2011
From: dunchan at freemail.hu (Viktor)
Date: Wed, 06 Apr 2011 14:41:07 +0200
Subject: [Freeswitch-users] FS and Phone are in the same IP
Message-ID: <jwjgdk7ku98e2qgmivhj8bbn.1302093375307@email.android.com>

There is no nat, preveously i got a tipp to check this section.
Phone is a softphone.
I think the problem is the FS doesn't forward the bye message to softphone which is listen on same ip but different port. FS suppuse the bye message sent to him.
How can i avoid that?
Thanks,
Viktor

Sent from Samsung Mobile

Meftah Tayeb <tayeb.meftah at gmail.com> wrote:

>where's nat in this path?
>you say phone and fs have same ip
>is this a hardphone or a softphone?
>if this is a hardphone, how come a pc and a phone can fill in the  same 
>network with same ip?
>thanks
>On 06/04/2011 07:27, dunchan wrote:
>> Hi!
>>
>> My voip phone doesn't get the SIP bye request from the called party (via
>> PSTN gateway)
>>
>> Phone is in the same machine with FS, IP addr is same too, Contact
>> header is filled corretly.
>>
>> If phone is in other machine everything works fine.
>>
>> Should I change the NAT handling somewhere, or what?
>>
>> thanks,
>> Viktor
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>    
>
>
>-- 
>Meftah Tayeb
>inum: +883510001288000
>phone: +13477595883
>
>

From vetali100 at gmail.com  Wed Apr  6 16:49:53 2011
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Wed, 6 Apr 2011 15:49:53 +0300
Subject: [Freeswitch-users] Need to enable NDLB_force_port to work properly
 with Windows PC Internet Gateway
Message-ID: <BANLkTi==YA4=kiAvwL7ZOXwnSDJBQNSQGQ@mail.gmail.com>

Hi list,

I am testing the following configuration:

Sip ATA adapter (Linksys PAP2) -> Windows PC (Internet Gateway) -> ...
Internet ... -> FreeSWITCH

Sip adapter sends REGISTER to the FS, from port 5060, however Windows
Gateway transfers this packet to FS from a different port (for example
62000).
FS replies to the port 5060 (it looks like it takes it from the SIP text
information).
So Windows Gateway receives the answer to the port 5060, but since it
expects  it to be received in 62000, it looks like it drops the packets.

The only way to fix this, is to set NDLB_force_port= true in the internal
profile configuration file.
In this case, FreeSWITCH replies exactly to the port 62000, and windows
forwards it to the Sip adapter on port 5060.


Do you know whether it is possible to configure Windows to route properly?
Or maybe there is a way to configure the Sip adapter? (Like it will say to
windows PC - hey, don't change the port, use the same as I am using... :))
Or it is a fantastic wish)

Otherwise I will have to create 2 internal sip profiles, one
with NDLB_force_port = true (for those who uses Windows PC as the gateway),
and another with NDLB_force_port = false (for those who use normal routers).
And I need 2 default dialplans, etc... so it complicates the solution.


Thank you,
Vitalie
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From randy.andrade at gmail.com  Wed Apr  6 16:54:52 2011
From: randy.andrade at gmail.com (Randy Andrade)
Date: Wed, 6 Apr 2011 08:54:52 -0400
Subject: [Freeswitch-users] Need to enable NDLB_force_port to work
 properly with Windows PC Internet Gateway
In-Reply-To: <BANLkTi==YA4=kiAvwL7ZOXwnSDJBQNSQGQ@mail.gmail.com>
References: <BANLkTi==YA4=kiAvwL7ZOXwnSDJBQNSQGQ@mail.gmail.com>
Message-ID: <BANLkTinnJtnvfFVZ1s791DW07n2FqQ5a4w@mail.gmail.com>

Is there no option of putting an inexpensive switch or router in front of
the Windows PC to perform Internet Gateway function? It would probably be
the preferred methodology.

On Wed, Apr 6, 2011 at 8:49 AM, Vitalii Colosov <vetali100 at gmail.com> wrote:

> Hi list,
>
> I am testing the following configuration:
>
> Sip ATA adapter (Linksys PAP2) -> Windows PC (Internet Gateway) -> ...
> Internet ... -> FreeSWITCH
>
> Sip adapter sends REGISTER to the FS, from port 5060, however Windows
> Gateway transfers this packet to FS from a different port (for example
> 62000).
> FS replies to the port 5060 (it looks like it takes it from the SIP text
> information).
> So Windows Gateway receives the answer to the port 5060, but since it
> expects  it to be received in 62000, it looks like it drops the packets.
>
> The only way to fix this, is to set NDLB_force_port= true in the internal
> profile configuration file.
> In this case, FreeSWITCH replies exactly to the port 62000, and windows
> forwards it to the Sip adapter on port 5060.
>
>
> Do you know whether it is possible to configure Windows to route properly?
> Or maybe there is a way to configure the Sip adapter? (Like it will say to
> windows PC - hey, don't change the port, use the same as I am using... :))
> Or it is a fantastic wish)
>
> Otherwise I will have to create 2 internal sip profiles, one
> with NDLB_force_port = true (for those who uses Windows PC as the gateway),
> and another with NDLB_force_port = false (for those who use normal routers).
> And I need 2 default dialplans, etc... so it complicates the solution.
>
>
> Thank you,
> Vitalie
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rhuddleston at gmail.com  Wed Apr  6 16:55:42 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Wed, 6 Apr 2011 08:55:42 -0400
Subject: [Freeswitch-users] FS and Phone are in the same IP
In-Reply-To: <jwjgdk7ku98e2qgmivhj8bbn.1302093375307@email.android.com>
References: <jwjgdk7ku98e2qgmivhj8bbn.1302093375307@email.android.com>
Message-ID: <26f401cbf459$f0e72d20$d2b58760$@com>

At VON conference 2 years ago we saw a firm who made an ATA device that
plugged into that fax machine. It handled the audio / digital translation of
sounds to a binary file - then pushed the file via http / ftp to their
servers which in turn completed the fax process to remote station.

Pretty impressive I thought.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Viktor
Sent: Wednesday, April 06, 2011 8:41 AM
To: Meftah Tayeb; FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FS and Phone are in the same IP

There is no nat, preveously i got a tipp to check this section.
Phone is a softphone.
I think the problem is the FS doesn't forward the bye message to softphone
which is listen on same ip but different port. FS suppuse the bye message
sent to him.
How can i avoid that?
Thanks,
Viktor

Sent from Samsung Mobile

Meftah Tayeb <tayeb.meftah at gmail.com> wrote:

>where's nat in this path?
>you say phone and fs have same ip
>is this a hardphone or a softphone?
>if this is a hardphone, how come a pc and a phone can fill in the  same 
>network with same ip?
>thanks
>On 06/04/2011 07:27, dunchan wrote:
>> Hi!
>>
>> My voip phone doesn't get the SIP bye request from the called party (via
>> PSTN gateway)
>>
>> Phone is in the same machine with FS, IP addr is same too, Contact
>> header is filled corretly.
>>
>> If phone is in other machine everything works fine.
>>
>> Should I change the NAT handling somewhere, or what?
>>
>> thanks,
>> Viktor
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>    
>
>
>-- 
>Meftah Tayeb
>inum: +883510001288000
>phone: +13477595883
>
>
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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  Wed Apr  6 17:05:57 2011
From: vetali100 at gmail.com (Vitalii Colosov)
Date: Wed, 6 Apr 2011 16:05:57 +0300
Subject: [Freeswitch-users] Need to enable NDLB_force_port to work
 properly with Windows PC Internet Gateway
In-Reply-To: <BANLkTinnJtnvfFVZ1s791DW07n2FqQ5a4w@mail.gmail.com>
References: <BANLkTi==YA4=kiAvwL7ZOXwnSDJBQNSQGQ@mail.gmail.com>
	<BANLkTinnJtnvfFVZ1s791DW07n2FqQ5a4w@mail.gmail.com>
Message-ID: <BANLkTi=xdt1CWpTyb5UwudMW+sCQRNA=xA@mail.gmail.com>

Hi Randy,
Thanks for the advice.

In general yes, there are always some options to be considered.

But mostly I am curious now - is Windows Internet gateway really not
following the standards, or it is just a matter of proper configuration?

Vitalie


2011/4/6 Randy Andrade <randy.andrade at gmail.com>

> Is there no option of putting an inexpensive switch or router in front of
> the Windows PC to perform Internet Gateway function? It would probably be
> the preferred methodology.
>
> On Wed, Apr 6, 2011 at 8:49 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>
>> Hi list,
>>
>> I am testing the following configuration:
>>
>> Sip ATA adapter (Linksys PAP2) -> Windows PC (Internet Gateway) -> ...
>> Internet ... -> FreeSWITCH
>>
>> Sip adapter sends REGISTER to the FS, from port 5060, however Windows
>> Gateway transfers this packet to FS from a different port (for example
>> 62000).
>> FS replies to the port 5060 (it looks like it takes it from the SIP text
>> information).
>> So Windows Gateway receives the answer to the port 5060, but since it
>> expects  it to be received in 62000, it looks like it drops the packets.
>>
>> The only way to fix this, is to set NDLB_force_port= true in the internal
>> profile configuration file.
>> In this case, FreeSWITCH replies exactly to the port 62000, and windows
>> forwards it to the Sip adapter on port 5060.
>>
>>
>> Do you know whether it is possible to configure Windows to route properly?
>> Or maybe there is a way to configure the Sip adapter? (Like it will say to
>> windows PC - hey, don't change the port, use the same as I am using... :))
>> Or it is a fantastic wish)
>>
>> Otherwise I will have to create 2 internal sip profiles, one
>> with NDLB_force_port = true (for those who uses Windows PC as the gateway),
>> and another with NDLB_force_port = false (for those who use normal routers).
>> And I need 2 default dialplans, etc... so it complicates the solution.
>>
>>
>> Thank you,
>> Vitalie
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From randy.andrade at gmail.com  Wed Apr  6 17:21:13 2011
From: randy.andrade at gmail.com (Randy Andrade)
Date: Wed, 6 Apr 2011 09:21:13 -0400
Subject: [Freeswitch-users] Need to enable NDLB_force_port to work
 properly with Windows PC Internet Gateway
In-Reply-To: <BANLkTi=xdt1CWpTyb5UwudMW+sCQRNA=xA@mail.gmail.com>
References: <BANLkTi==YA4=kiAvwL7ZOXwnSDJBQNSQGQ@mail.gmail.com>
	<BANLkTinnJtnvfFVZ1s791DW07n2FqQ5a4w@mail.gmail.com>
	<BANLkTi=xdt1CWpTyb5UwudMW+sCQRNA=xA@mail.gmail.com>
Message-ID: <BANLkTi=M82X4Acje=D7tJgjsJRhkY3peEA@mail.gmail.com>

It sounds like the Windows Internet Gateway is performing a standard PAT
(port address translation) version of NAT, but since it is not SIP aware, it
is not smart enough to realize that the SIP header contains directions to
reply on a specific port (5060) it just figures "if I pass this data out
port 62000, I expect it back on port 62000" it doesn't know anything about
what the data contains..

Most routers have (at least a limited) an awareness of SIP traffic, so they
"read" the SIP messages and manipulate port information when it re-sends the
packet. That's to say when it passes the data out port 62000 on the public /
WAN interface, it's changed the SIP header to tell FS to reply back on port
62000.. when it gets the reply, it then changes the SIP header again and
passes the message back to the ATA on port 5060 of it's private / LAN
interface.

On Wed, Apr 6, 2011 at 9:05 AM, Vitalii Colosov <vetali100 at gmail.com> wrote:

> Hi Randy,
> Thanks for the advice.
>
> In general yes, there are always some options to be considered.
>
> But mostly I am curious now - is Windows Internet gateway really not
> following the standards, or it is just a matter of proper configuration?
>
> Vitalie
>
>
> 2011/4/6 Randy Andrade <randy.andrade at gmail.com>
>
>> Is there no option of putting an inexpensive switch or router in front of
>> the Windows PC to perform Internet Gateway function? It would probably be
>> the preferred methodology.
>>
>> On Wed, Apr 6, 2011 at 8:49 AM, Vitalii Colosov <vetali100 at gmail.com>wrote:
>>
>>> Hi list,
>>>
>>> I am testing the following configuration:
>>>
>>> Sip ATA adapter (Linksys PAP2) -> Windows PC (Internet Gateway) -> ...
>>> Internet ... -> FreeSWITCH
>>>
>>> Sip adapter sends REGISTER to the FS, from port 5060, however Windows
>>> Gateway transfers this packet to FS from a different port (for example
>>> 62000).
>>> FS replies to the port 5060 (it looks like it takes it from the SIP text
>>> information).
>>> So Windows Gateway receives the answer to the port 5060, but since it
>>> expects  it to be received in 62000, it looks like it drops the packets.
>>>
>>> The only way to fix this, is to set NDLB_force_port= true in the internal
>>> profile configuration file.
>>> In this case, FreeSWITCH replies exactly to the port 62000, and windows
>>> forwards it to the Sip adapter on port 5060.
>>>
>>>
>>> Do you know whether it is possible to configure Windows to route
>>> properly?
>>> Or maybe there is a way to configure the Sip adapter? (Like it will say
>>> to windows PC - hey, don't change the port, use the same as I am using...
>>> :)) Or it is a fantastic wish)
>>>
>>> Otherwise I will have to create 2 internal sip profiles, one
>>> with NDLB_force_port = true (for those who uses Windows PC as the gateway),
>>> and another with NDLB_force_port = false (for those who use normal routers).
>>> And I need 2 default dialplans, etc... so it complicates the solution.
>>>
>>>
>>> Thank you,
>>> Vitalie
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Nabble at slickdeals.endjunk.com  Wed Apr  6 17:41:12 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 6 Apr 2011 06:41:12 -0700 (PDT)
Subject: [Freeswitch-users] FS and Phone are in the same IP
In-Reply-To: <4D9C5363.3060106@gmail.com>
References: <4D9BF9D2.8010401@freemail.hu> <4D9C5363.3060106@gmail.com>
Message-ID: <1302097272699-6246064.post@n2.nabble.com>


Meftah Tayeb wrote:
> if this is a hardphone, how come a pc and a phone can fill in the  same 
> network with same ip?
With an FS system hosted on Windows platform, perhaps it is possible to add
a USB2FX dongle with supported software/drivers to make it an extension to
FS. I don't know if one can use the USB2FX dongle provided by MagicJack.

You just gave me an idea. I have an old 
http://fobbit.net/voipblaster/index.html VoipBlaster  and IIRC it is
supported by OpenGK(?). Perhaps, I can start to incorporate this old 
http://fobbit.net/voipblaster/index.html VoipBlaster  into my Seagate 
http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
DockStar . This way, I will have both a hardware phone and FS hosted on a
Seagate 
http://www.seagate.com/www/en-us/products/network_storage/freeagent_dockstar
DockStar . 

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-and-Phone-are-in-the-same-IP-tp6244790p6246064.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From Lars.Bobka at web.de  Wed Apr  6 17:52:31 2011
From: Lars.Bobka at web.de (Lars Bobka)
Date: Wed, 6 Apr 2011 15:52:31 +0200 (CEST)
Subject: [Freeswitch-users]  Gateway ReInvite Problem with a=inactive in
Message-ID: <509194844.131234.1302097951202.JavaMail.fmail@mwmweb040>

ok, I understand, that the sdp is media and so untouched by the freeswitch.

But isn?t there a possibility to change this behaviour, for example in the dialplan, or somewhere else.
It could also be an automatic hold/unhold, after answering the call, that the call becomes active.
Is something like this possible?

regards
Lars


___________________________________________________________
Schon geh?rt? WEB.DE hat einen genialen Phishing-Filter in die
Toolbar eingebaut! http://produkte.web.de/go/toolbar


From kbdfck at gmail.com  Wed Apr  6 18:35:58 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 6 Apr 2011 18:35:58 +0400
Subject: [Freeswitch-users] Correct way to determine bridge result?
Message-ID: <BANLkTinVtDgMmaatHtoEVixXxz+exji8jw@mail.gmail.com>

Hi all!

How to correctly determine bridge result (answer/no answer and reason) after
executing Bridge on already answered channel? endpoint_disposition shows
'ANSWER' as call is answered.

Call goes to FS, FS answers, reads digits and attempts bridge to some
destination. But I can't reliable determine whether this call was answered
or rejected with some status, as originate_disposition is set to SUCCESS
even if we just get early media, and endpoint_disposition contains 'ANSWER'
related to current channel.


-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From Nabble at slickdeals.endjunk.com  Wed Apr  6 18:57:57 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 6 Apr 2011 07:57:57 -0700 (PDT)
Subject: [Freeswitch-users] Connecting HP 8500 All in One FAX Machine to
	FS
In-Reply-To: <BANLkTikawF1e2UX4-AparihLMYi=JQ7ZcA@mail.gmail.com>
References: <BANLkTikawF1e2UX4-AparihLMYi=JQ7ZcA@mail.gmail.com>
Message-ID: <1302101877860-6246332.post@n2.nabble.com>


Nice Voip wrote:
> I have got HP 8500 All in One machine, including FAX, now i want to use it
> to send faxes to my VoIP provider through FS, how do i connect it to FS?
How does HP8500 support net-printing FAX? On my Canon ImageClass D480,
Windows users can send fax by printing it to the FAX printer. If your HP8500
does the same thing, perhaps you can write a LUA scripting file do interface
with your HP8500 to do the net-printing FAX.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Connecting-HP-8500-All-in-One-FAX-Machine-to-FS-tp6243233p6246332.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From anthony.minessale at gmail.com  Wed Apr  6 18:59:30 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Apr 2011 09:59:30 -0500
Subject: [Freeswitch-users] Gateway ReInvite Problem with a=inactive in
In-Reply-To: <509194844.131234.1302097951202.JavaMail.fmail@mwmweb040>
References: <509194844.131234.1302097951202.JavaMail.fmail@mwmweb040>
Message-ID: <BANLkTi=BXowFADcPRyFC8Z5_q3Ah27RDYw@mail.gmail.com>

Yes,
set the param media-option to resume-media-on-hold in your sofia profile.


On Wed, Apr 6, 2011 at 8:52 AM, Lars Bobka <Lars.Bobka at web.de> wrote:
> ok, I understand, that the sdp is media and so untouched by the freeswitch.
>
> But isn?t there a possibility to change this behaviour, for example in the dialplan, or somewhere else.
> It could also be an automatic hold/unhold, after answering the call, that the call becomes active.
> Is something like this possible?
>
> regards
> Lars
>
>
> ___________________________________________________________
> Schon geh?rt? WEB.DE hat einen genialen Phishing-Filter in die
> Toolbar eingebaut! http://produkte.web.de/go/toolbar
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From kaushalshriyan at gmail.com  Wed Apr  6 10:28:29 2011
From: kaushalshriyan at gmail.com (Kaushal Shriyan)
Date: Wed, 6 Apr 2011 11:58:29 +0530
Subject: [Freeswitch-users] Freeswitch
Message-ID: <BANLkTikCO-30PnPTgkam7+t4fo9VEAH=pQ@mail.gmail.com>

Hi,

I have couple of questions regarding Asterisk.

a) Does it has Automated Dialing Feature like dialing 1000 and 1000 of phone
numbers?
b) Does it Support VoiceXML ?
c) What PRI Card is recommended for using Asterisk ?

Thanks

Kaushal
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From kaushalshriyan at gmail.com  Wed Apr  6 10:44:08 2011
From: kaushalshriyan at gmail.com (Kaushal Shriyan)
Date: Wed, 6 Apr 2011 12:14:08 +0530
Subject: [Freeswitch-users] Freeswitch
In-Reply-To: <BANLkTikCO-30PnPTgkam7+t4fo9VEAH=pQ@mail.gmail.com>
References: <BANLkTikCO-30PnPTgkam7+t4fo9VEAH=pQ@mail.gmail.com>
Message-ID: <BANLkTimQvgj7bU1_bxSi2VWwYz8WzcZJ4g@mail.gmail.com>

typo it was meant for FreeSwitch

On Wed, Apr 6, 2011 at 11:58 AM, Kaushal Shriyan
<kaushalshriyan at gmail.com>wrote:

> Hi,
>
> I have couple of questions regarding Asterisk.
>
> a) Does it has Automated Dialing Feature like dialing 1000 and 1000 of
> phone numbers?
> b) Does it Support VoiceXML ?
> c) What PRI Card is recommended for using Asterisk ?
>
> Thanks
>
> Kaushal
>
>
>
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From gmaruzz at gmail.com  Wed Apr  6 19:33:54 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 6 Apr 2011 17:33:54 +0200
Subject: [Freeswitch-users] Freeswitch
In-Reply-To: <BANLkTimQvgj7bU1_bxSi2VWwYz8WzcZJ4g@mail.gmail.com>
References: <BANLkTikCO-30PnPTgkam7+t4fo9VEAH=pQ@mail.gmail.com>
	<BANLkTimQvgj7bU1_bxSi2VWwYz8WzcZJ4g@mail.gmail.com>
Message-ID: <BANLkTimHtysU_z2rbns+asa5X2nXEMHrew@mail.gmail.com>

On Wed, Apr 6, 2011 at 8:44 AM, Kaushal Shriyan
<kaushalshriyan at gmail.com> wrote:
> typo it was meant for FreeSwitch
>
> On Wed, Apr 6, 2011 at 11:58 AM, Kaushal Shriyan <kaushalshriyan at gmail.com>
> wrote:
>>
>> Hi,
>> I have couple of questions regarding Asterisk.
>> a) Does it has Automated Dialing Feature like dialing 1000 and 1000 of
>> phone numbers?


With FreeSWITCH you can voice spam the entire globe.
I've heard of Jim Strlbinsky in Pallaawooka that's using FreeSWITCH
running on an EEEpc reaching 10.000.000 cps.
Yes, that's ten million calls per seconds on an EEEpc!
(OK, that's using g711. If you use a cpu hungry compressed format to
save on bandwidth, let's say speex or celt, then you can't expect much
more than one million calls per seconds on an EEEpc).


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From anthony.minessale at gmail.com  Wed Apr  6 20:05:42 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Apr 2011 11:05:42 -0500
Subject: [Freeswitch-users] BUG FIX: "Buffer size sanity check failed!"
 drops FAX receiving unneeded
In-Reply-To: <4D998A5A.6080901@priv.de>
References: <4D998A5A.6080901@priv.de>
Message-ID: <BANLkTimSg1hmwtCUwfJkA_DRAgULW32QBQ@mail.gmail.com>

Can you please get a pcap of a single call (without your patch) as
well as a full capture of the freeswitch console logs and post it to
JIRA.

Are you using the described TDM inside FreeSWITCH or is this a SIP
call from an Asterisk machine?



On Mon, Apr 4, 2011 at 4:07 AM, Markus Mueller <freeswitch at priv.de> wrote:
> Hello FreeSwitch users and programmers,
>
> I found a problem on receiving faxes and want to share a working patch
> for this. The problem is that on receiving a fax, it is unneeded aborted
> by a sanity check. Sanity checks are fine, but a unneeded abort instead
> of a warning is in productive versions not the best solution.
>
> The message apearing is:
>
> 2011-04-04 10:44:52.060860 [CRIT] switch_core_codec.c:660 Buffer size
> sanity check failed!
>
> which is normaly aborting in receiving the fax. Simply decreasing this
> fault to a warning let the server receive the fax without any problems.
> After the patch the message apears up to five times per fax before the
> fax is beeing accepted. I am using libpri with the three HFC ISDN Cards
> and the DAHDI from Debian Squeeze 6.0. For more informations about my
> hardware just write me an email.
>
> Regards,
> Markus Mueller
> http://projekte.priv.de/
>
> root at sip:/usr/local/src/freeswitch/src# diff -U 4 switch_core_codec.c*
> --- switch_core_codec.c 2011-03-14 10:49:17.000000000 +0100
> +++ switch_core_codec.c.org ? ? 2011-03-14 10:47:02.000000000 +0100
> @@ -657,9 +657,9 @@
> ? ? ? ? ? ? ? ? uint32_t frames = encoded_data_len /
> codec->implementation->encoded_bytes_per_packet;
>
> ? ? ? ? ? ? ? ? if (frames &&
> codec->implementation->decoded_bytes_per_packet * frames >
> *decoded_data_len) {
> ? ? ? ? ? ? ? ? ? ? ? ? switch_log_printf(SWITCH_CHANNEL_LOG,
> SWITCH_LOG_CRIT, "Buffer size sanity check failed!\n");
> - ? ? ? ? ? ? ? ? ? ? ? // return SWITCH_STATUS_GENERR;
> + ? ? ? ? ? ? ? ? ? ? ? return SWITCH_STATUS_GENERR;
> ? ? ? ? ? ? ? ? }
> ? ? ? ? }
>
> ? ? ? ? if (codec->mutex) switch_mutex_lock(codec->mutex);
> root at sip:/usr/local/src/freeswitch/src#
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From anthony.minessale at gmail.com  Wed Apr  6 20:33:04 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Apr 2011 11:33:04 -0500
Subject: [Freeswitch-users] uuid_hold is not send hold message
In-Reply-To: <8slt5a$arb4s0@irxd5-187.sinamail.sina.com.cn>
References: <8slt5a$arb4s0@irxd5-187.sinamail.sina.com.cn>
Message-ID: <BANLkTinX0Ac7QFt1=e679pP0QWL8SoeB4Q@mail.gmail.com>

you can not use uuid_hold to hold a call then the phone does the unhold.
you need a phone that supports the notify with talk/hold event and use

uuid_phone_event  <uuid> hold
uuid_phone_event  <uuid> talk


2011/4/5 ?? <sc_zhangming at sina.com>

> Michael Collins????
>
>  use step, freeswitch version 1.07
>         1.   fs_cli
>         2.  /event plain CHANNEL_HOLD CHANNEL_UNHOLD
>         3. show  calls
>         freeswitch at 10.108.226.220@internal> show calls
>
> call_uuid,call_created,call_created_epoch,function,caller_cid_name,caller_cid_num,caller_dest_num,caller_chan_name,caller_uuid,callee_cid_name,callee_cid_num,callee_dest_num,callee_chan_name,callee_uuid,hostname
> b0a7bdfd-430d-4746-91a4-8c318f9e1695,2011-04-08
> 04:22:37,1302207757,switch_ivr_multi_threaded_bridge,700000,700000,700001,sofia/internal/
> 700000 at 10.108.226.220,b0a7bdfd-430d-4746-91a4-8c318f9e1695,Outbound
> Call,700001,700001,sofia/internal/sip:700001 at 10.108.226.44:18692
> ,e2faae72-ef43-4019-b7e6-1120d7e42e1e,localhost.localdomain
>
> 1 total.
>         4.   uuid_hold  b0a7bdfd-430d-4746-91a4-8c318f9e1695
>             2011-04-08 04:27:55.539115 [DEBUG] switch_channel.c:1373 (
> sofia/internal/700000 at 10.108.226.220) Callstate Change ACTIVE -> HELD
> 2011-04-08 04:27:55.539115 [DEBUG] switch_core_session.c:709 Send signal
> sofia/internal/700000 at 10.108.226.220 [BREAK]
> 2011-04-08 04:27:55.539115 [DEBUG] switch_ivr.c:1272 ??????
> 2011-04-08 04:27:55.539115 [DEBUG] switch_core_session.c:954 Send signal
> sofia/internal/sip:700001 at 10.108.226.44:18692 [BREAK]
> 2011-04-08 04:27:55.539115 [DEBUG] sofia.c:4646 Channel
> sofia/internal/700000 at 10.108.226.220 entering state [calling][0]
> 2011-04-08 04:27:55.544824 [DEBUG] switch_core_session.c:709 Send signal
> sofia/internal/sip:700001 at 10.108.226.44:18692 [BREAK]
> 2011-04-08 04:27:55.544824 [INFO] sofia.c:729
> sofia/internal/700000 at 10.108.226.220 Update Callee ID to "Outbound Call"
> <700000>
> 2011-04-08 04:27:55.547789 [DEBUG] sofia.c:4646 Channel
> sofia/internal/700000 at 10.108.226.220 entering state [ready][200]
> 2011-04-08 04:27:55.547789 [DEBUG] sofia.c:4654 Duplicate SDP
> v=0
> o=Idefisk_user 47685 6056184806875839522 IN IP4 10.108.226.44
> s=Idefisk_user
> c=IN IP4 10.108.226.44
> t=0 0
> m=audio 8000 RTP/AVP 8 97 0 110 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:97 iLBC/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:110 speex/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
>
> 2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000]
> 2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000]
> 2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000]
> 2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
> 2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:2690 Already using PCMA
> 2011-04-08 04:27:55.547789 [DEBUG] sofia_glue.c:4571 Set 2833 dtmf
> send/recv payload to 101
> 2011-04-08 04:27:55.725181 [DEBUG] switch_ivr.c:563 sofia/internal/
> sip:700001 at 10.108.226.44:18692 Command Execute
> playback(local_stream://moh)
> EXECUTE sofia/internal/sip:700001 at 10.108.226.44:18692playback(local_stream://moh)
> 2011-04-08 04:27:55.725181 [DEBUG] mod_local_stream.c:421 Opening Stream
> [moh/8000] 8000hz
> 2011-04-08 04:27:55.725181 [DEBUG] switch_ivr_play_say.c:1236 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
>         5.  it 's not return  hold message, if  user  Zoiper  hold button
> freeswitch  return hold message
>
>             Zoiper hold  freeswich return message:
>             2011-04-08 04:34:51.330157 [DEBUG] sofia.c:4646 Channel
> sofia/internal/700000 at 10.108.226.220 entering state [received][100]
> RECV EVENT
> Event-Name: CHANNEL_HOLD
> Core-UUID: 5e94c1c9-305d-4c4d-b00c-ea24f6346559
> FreeSWITCH-Hostname: localhost.localdomain
> FreeSWITCH-IPv4: 10.108.226.220
> FreeSWITCH-IPv6: ::1
> Event-Date-Local: 2011-04-08 04:34:51
> Event-Date-GMT: Thu, 07 Apr 2011 20:34:51 GMT
> Event-Date-Timestamp: 1302208491330157
> Event-Calling-File: switch_channel.c
> Event-Calling-Function: switch_channel_mark_hold
> Event-Calling-Line-Number: 642
> Channel-State: CS_EXECUTE
> Channel-Call-State: HELD
> Channel-State-Number: 4
> Channel-Name: sofia/internal/700000 at 10.108.226.220
> Unique-ID: eb1222c9-4175-469a-8549-667c0dd921a6
> Call-Direction: inbound
> Presence-Call-Direction: inbound
> Channel-Presence-ID: 700000 at 10.108.226.220
> Channel-Call-UUID: eb1222c9-4175-469a-8549-667c0dd921a6
> Answer-State: answered
> Channel-Read-Codec-Name: PCMA
> Channel-Read-Codec-Rate: 8000
> Channel-Read-Codec-Bit-Rate: 64000
> Channel-Write-Codec-Name: PCMA
> Channel-Write-Codec-Rate: 8000
> Channel-Write-Codec-Bit-Rate: 64000
> Caller-Direction: inbound
> Caller-Username: 700000
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: 700000
> Caller-Caller-ID-Number: 700000
> Caller-Callee-ID-Name: Outbound Call
> Caller-Callee-ID-Number: 700001
> Caller-Network-Addr: 10.108.226.44
> Caller-ANI: 700000
> Caller-Destination-Number: 700001
> Caller-Unique-ID: eb1222c9-4175-469a-8549-667c0dd921a6
> Caller-Source: mod_sofia
> Caller-Context: default
> Caller-Channel-Name: sofia/internal/700000 at 10.108.226.220
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1302208456152936
> Caller-Channel-Created-Time: 1302208456152936
> Caller-Channel-Answered-Time: 1302208463040930
> Caller-Channel-Progress-Time: 1302208456340619
> Caller-Channel-Progress-Media-Time: 0
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: true
> Caller-Privacy-Hide-Name: false
> Caller-Privacy-Hide-Number: false
> Other-Type: originatee
> Other-Leg-Direction: outbound
> Other-Leg-Username: 700000
> Other-Leg-Dialplan: XML
> Other-Leg-Caller-ID-Name: 700000
> Other-Leg-Caller-ID-Number: 700000
> Other-Leg-Callee-ID-Name: Outbound Call
> Other-Leg-Callee-ID-Number: 700001
> Other-Leg-Network-Addr: 10.108.226.44
> Other-Leg-ANI: 700000
> Other-Leg-Destination-Number: 700001
> Other-Leg-Unique-ID: 1595a55f-d97c-47c1-aa6c-42c8efcb5b27
> Other-Leg-Source: mod_sofia
> Other-Leg-Context: default
> Other-Leg-Channel-Name: sofia/internal/sip:700001 at 10.108.226.44:18692
> Other-Leg-Screen-Bit: true
> Other-Leg-Privacy-Hide-Name: false
> Other-Leg-Privacy-Hide-Number: false
>
>
> 2011-04-08 04:34:51.330157 [DEBUG] sofia.c:4657 Remote SDP:
> v=0
> o=Idefisk_user 6056184806875821816 6056184806875816806 IN IP4 10.108.226.44
> s=Idefisk_user
> c=IN IP4 0.0.0.0
> t=0 0
> m=audio 8000 RTP/AVP 8 97 0 110 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:97 iLBC/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:110 speex/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendonly
>
> 2011-04-08 04:34:51.330157 [DEBUG] switch_channel.c:1373 (
> sofia/internal/700000 at 10.108.226.220) Callstate Change ACTIVE -> HELD
> 2011-04-08 04:34:51.330157 [DEBUG] switch_core_session.c:954 Send signal
> sofia/internal/sip:700001 at 10.108.226.44:18692 [BREAK]
> 2011-04-08 04:34:51.346542 [DEBUG] switch_core_session.c:709 Send signal
> sofia/internal/sip:700001 at 10.108.226.44:18692 [BREAK]
> 2011-04-08 04:34:51.524112 [DEBUG] switch_ivr.c:563 sofia/internal/
> sip:700001 at 10.108.226.44:18692 Command Execute
> playback(local_stream://moh)
> EXECUTE sofia/internal/sip:700001 at 10.108.226.44:18692playback(local_stream://moh)
> 2011-04-08 04:34:51.524112 [DEBUG] mod_local_stream.c:421 Opening Stream
> [moh/8000] 8000hz
> 2011-04-08 04:34:51.524112 [DEBUG] switch_ivr_play_say.c:1236 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[G7221:115:32000:20:48000]
> 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[G7221:107:16000:20:32000]
> 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[G722:9:8000:20:64000]
> 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4467 Audio Codec Compare
> [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000]
> 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:2690 Already using PCMA
> 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:4571 Set 2833 dtmf
> send/recv payload to 101
> 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:2976 Audio params changed
> for sofia/internal/700000 at 10.108.226.220 from 10.108.226.44:8000 to
> 0.0.0.0:8000
> 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:2987 AUDIO RTP
> [sofia/internal/700000 at 10.108.226.220] 10.108.226.220 port 19022 ->
> 0.0.0.0 port 8000 codec: 8 ms: 20
> 2011-04-08 04:34:51.580870 [DEBUG] sofia_glue.c:3017 AUDIO RTP CHANGING
> DEST TO: [0.0.0.0:8000]
> 2011-04-08 04:34:51.580870 [DEBUG] sofia.c:5057 Processing updated SDP
> 2011-04-08 04:34:51.583974 [DEBUG] sofia.c:4646 Channel
> sofia/internal/700000 at 10.108.226.220 entering state [completed][200]
> 2011-04-08 04:34:51.583974 [DEBUG] sofia.c:4646 Channel
> sofia/internal/700000 at 10.108.226.220 entering state [ready][200]
>
>
> uuid_hold  command user problem:
>         1.  uuid_hold   uuid  is error still return OK
>         2.  uuid_hold   <uuid> OR  uuid_hold off  <uuid>   freeswitch is
> not return  hold or unhold message
>
> ======== 2011-04-05 03:23:09 ???????? ========
>
>
> I just tried this on latest git and it worked fine for me. Can you pastebin
> the console debug output when you use it?
> -MC
>
> 2011/4/2 ?? <sc_zhangming at sina.com>
>
>> freeswitch-users????
>>
>>        uuid_hold command , freeswitch is not send HOLD  message. who know
>> it.
>>
>> ?
>> ??
>>
>>
>> ??
>> sc_zhangming at sina.com
>> 2011-04-02
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  = = = = = = = = = = = = = = = = = = = = = =
>
> ?
> ??
>
>  ??
>  sc_zhangming at sina.com
>  2011-04-06
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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From pablosaro at gmail.com  Wed Apr  6 20:44:56 2011
From: pablosaro at gmail.com (Pablo Hernan Saro)
Date: Wed, 6 Apr 2011 13:44:56 -0300
Subject: [Freeswitch-users] Hylafax server emulation
In-Reply-To: <4D9C5C45.7030702@coppice.org>
References: <4D9B42D7.4020008@coppice.org>
	<BANLkTinaftq2FapGp4tM3Ysjr1_Wd8gGbw@mail.gmail.com>
	<BANLkTimXgW1FNMFtFwTHGQ_sgtXNEiDwqw@mail.gmail.com>
	<4D9C5C45.7030702@coppice.org>
Message-ID: <BANLkTik=eKe6ZjQO+uF-UVxrb_N6FQCMkg@mail.gmail.com>

I don't proclaim myself the best telephony engineer in the world, so I'm
free to go wrong and open to learn. You can consider my contribution the
most bizarre solution you've seen in a long time, we're all free to think
and speak whatever we want (this is beyond discussion). IMHO, it is a simple
way to integrate HylaFAX and FS without programming a line of code. SIP and
t38 is out there to be used, in fact AFAIK all SIP trunk service providers
offer fax support this way. Why you believe it's the most bizarre solution?

PS: I do not want to challenge you, just want to feed my knowledge.

Pablo

On Wed, Apr 6, 2011 at 9:27 AM, Steve Underwood <steveu at coppice.org> wrote:

> Wow. That's one of the most bizarre solutions I've seen in a long time. :-\
>
> Steve
>
> On 04/06/2011 01:58 PM, Pablo Hernan Saro wrote:
> > Hi Steve,
> >
> > I'm not sure if the following will work. Let's say that you have a SIP
> > trunk service configured in a FS box for interconnection with the
> > PSTN. To avoid individual analog fax machines (reduce hardware costs
> > and attached maintenance costs) and take advantage of fax server
> > capabilities, a solution would be install HylaFAX with t38modem built
> > with OPAL support (this enables routing modems to SIP URIs)
> > sending/receiving calls to/from FS via SIP (configure your dialplan
> > for enabling t38 at FS side). That way you get the best from HylaFAX
> > (jub submission, scheduling, retrying, reporting and other facilities)
> > and FS (dealing with SIP trunk service providers, cdr, call routing)
> > at the same time.
> > End users can interact with HylaFAX using compatible windows clients,
> > email interface or web interface. A very good idea would be link
> > HylaFAX accounts to FS directory (connect HylaFAX to FS db may be?).
> > My two cents...
> >
> >
> > On Tue, Apr 5, 2011 at 4:20 PM, Kristian Kielhofner <kris at kriskinc.com
> > <mailto:kris at kriskinc.com>> wrote:
> >
> >     Steve,
> >
> >      Very cool (and I'm very interested).
> >
> >      What about using the existing user directory (perhaps with
> >     additional params) for the accounts and the FreeSWITCH core DB
> >     (whether SQLite or ODBC) for the jobs, etc?
> >
> >     On Tue, Apr 5, 2011 at 12:27 PM, Steve Underwood
> >     <steveu at coppice.org <mailto:steveu at coppice.org>> wrote:
> >     > Hi,
> >     >
> >     > It has always been clear that a HylaFAX compatible FAX job
> >     submission
> >     > server would add considerably to the value of the FAX facilities in
> >     > Asterisk and Freeswitch, but somehow it hasn't happened until now.
> I
> >     > recently found that in 2005 someone produced something fairly
> >     basic for
> >     > Asterisk in Perl, but it doesn't seem to have been well
> >     publicised, and
> >     > it looks like development stalled long ago.
> >     >
> >     > I now have the skeleton of HylaFAX compatible FAX job submission
> >     server,
> >     > in C, working. It accepts FAX submissions from sendfax and a
> >     couple of
> >     > the windows HylaFAX clients, though it needs a lot more
> >     polishing. Now I
> >     > need to look at the best thing to do on the Freeswitch side. I
> >     aim to
> >     > make the server maintain its own database of FAX jobs. It will
> >     attach to
> >     > Freeswitch, by ESL; push the jobs through FS; deal with scheduling,
> >     > retries, etc; and report the final result to the user, just as
> >     HylaFAX
> >     > does. The thing I am rather unsure about is the best way to
> >     handle the
> >     > accounts used to accept FAX jobs? Should I maintain a separate
> >     database
> >     > of FAX accounts, or hook into an existing database? I would welcome
> >     > suggestions for what would be the most useful approach.
> >     >
> >     > 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
>
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From msc at freeswitch.org  Wed Apr  6 23:06:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 6 Apr 2011 12:06:51 -0700
Subject: [Freeswitch-users] Correct way to determine bridge result?
In-Reply-To: <BANLkTinVtDgMmaatHtoEVixXxz+exji8jw@mail.gmail.com>
References: <BANLkTinVtDgMmaatHtoEVixXxz+exji8jw@mail.gmail.com>
Message-ID: <BANLkTim5Ek_rx_qWO3AaLVz4si7N2B+U5w@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Variable_bridge_hangup_cause

On Wed, Apr 6, 2011 at 7:35 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:

> Hi all!
>
> How to correctly determine bridge result (answer/no answer and reason)
> after executing Bridge on already answered channel? endpoint_disposition
> shows 'ANSWER' as call is answered.
>
> Call goes to FS, FS answers, reads digits and attempts bridge to some
> destination. But I can't reliable determine whether this call was answered
> or rejected with some status, as originate_disposition is set to SUCCESS
> even if we just get early media, and endpoint_disposition contains 'ANSWER'
> related to current channel.
>
>
> --
> Best regards,
>
> Dmitry Sytchev,
> IT Engineer
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From richocet2 at hotmail.com  Thu Apr  7 00:16:53 2011
From: richocet2 at hotmail.com (Dave Bracken)
Date: Wed, 6 Apr 2011 15:16:53 -0500
Subject: [Freeswitch-users] softphone to outbound help
Message-ID: <SNT144-w36A0F769DFF434C37D2A98F0A50@phx.gbl>


Can anyone tell me what i need to do to be able to dial a 7 digit number on my softphone and have it dial out to a phone i have sitting on my desk? I guess what i really need to know is what all files do i need to set up and where, etc. I have never done anything like this, so i need your help. Thanks in advance,
Dave 		 	   		  
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From gmaruzz at gmail.com  Thu Apr  7 01:24:30 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 6 Apr 2011 23:24:30 +0200
Subject: [Freeswitch-users] softphone to outbound help
In-Reply-To: <SNT144-w36A0F769DFF434C37D2A98F0A50@phx.gbl>
References: <SNT144-w36A0F769DFF434C37D2A98F0A50@phx.gbl>
Message-ID: <BANLkTi=rxc76ZNx1b5iYc=Mmp5K1mNKPaw@mail.gmail.com>

Look in the wiki for "home pbx", you'll find a good and commented example.
Http://Wiki.freeswitch.org

On 4/6/11, Dave Bracken <richocet2 at hotmail.com> wrote:
>
> Can anyone tell me what i need to do to be able to dial a 7 digit number on
> my softphone and have it dial out to a phone i have sitting on my desk? I
> guess what i really need to know is what all files do i need to set up and
> where, etc. I have never done anything like this, so i need your help.
> Thanks in advance,
> Dave 		 	   		

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From mattzerah+freeswitch at gmail.com  Thu Apr  7 02:15:55 2011
From: mattzerah+freeswitch at gmail.com (Matt Paine)
Date: Thu, 7 Apr 2011 08:15:55 +1000
Subject: [Freeswitch-users] BUG FIX: "Buffer size sanity check failed!"
 drops FAX receiving unneeded
In-Reply-To: <BANLkTimSg1hmwtCUwfJkA_DRAgULW32QBQ@mail.gmail.com>
References: <4D998A5A.6080901@priv.de>
	<BANLkTimSg1hmwtCUwfJkA_DRAgULW32QBQ@mail.gmail.com>
Message-ID: <BANLkTi=sZwvZw+TJV0W1tMJOc38ytv-ddw@mail.gmail.com>

I can second this behaviour... It wont be the weekend until I can actually
get any packet captures for the unpatched code, I can certianly find some
full freeswitch console logs if that will help.

Has a JIRA bug been filed for this yet? What is the number so I can
contribute?

Matt.


On 7 April 2011 02:05, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> Can you please get a pcap of a single call (without your patch) as
> well as a full capture of the freeswitch console logs and post it to
> JIRA.
>
> Are you using the described TDM inside FreeSWITCH or is this a SIP
> call from an Asterisk machine?
>
>
>
> On Mon, Apr 4, 2011 at 4:07 AM, Markus Mueller <freeswitch at priv.de> wrote:
> > Hello FreeSwitch users and programmers,
> >
> > I found a problem on receiving faxes and want to share a working patch
> > for this. The problem is that on receiving a fax, it is unneeded aborted
> > by a sanity check. Sanity checks are fine, but a unneeded abort instead
> > of a warning is in productive versions not the best solution.
> >
> > The message apearing is:
> >
> > 2011-04-04 10:44:52.060860 [CRIT] switch_core_codec.c:660 Buffer size
> > sanity check failed!
> >
> > which is normaly aborting in receiving the fax. Simply decreasing this
> > fault to a warning let the server receive the fax without any problems.
> > After the patch the message apears up to five times per fax before the
> > fax is beeing accepted. I am using libpri with the three HFC ISDN Cards
> > and the DAHDI from Debian Squeeze 6.0. For more informations about my
> > hardware just write me an email.
> >
> > Regards,
> > Markus Mueller
> > http://projekte.priv.de/
> >
> > root at sip:/usr/local/src/freeswitch/src# diff -U 4 switch_core_codec.c*
> > --- switch_core_codec.c 2011-03-14 10:49:17.000000000 +0100
> > +++ switch_core_codec.c.org     2011-03-14 10:47:02.000000000 +0100
> > @@ -657,9 +657,9 @@
> >                 uint32_t frames = encoded_data_len /
> > codec->implementation->encoded_bytes_per_packet;
> >
> >                 if (frames &&
> > codec->implementation->decoded_bytes_per_packet * frames >
> > *decoded_data_len) {
> >                         switch_log_printf(SWITCH_CHANNEL_LOG,
> > SWITCH_LOG_CRIT, "Buffer size sanity check failed!\n");
> > -                       // return SWITCH_STATUS_GENERR;
> > +                       return SWITCH_STATUS_GENERR;
> >                 }
> >         }
> >
> >         if (codec->mutex) switch_mutex_lock(codec->mutex);
> > root at sip:/usr/local/src/freeswitch/src#
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> 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
>
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From bwibowo at gmail.com  Thu Apr  7 03:42:18 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Thu, 7 Apr 2011 06:42:18 +0700
Subject: [Freeswitch-users] webphone app
Message-ID: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>

looking for webphone sip based on flash.
any info, please share


thx

budi wibowo
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From infos at madovsky.org  Thu Apr  7 03:54:48 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 6 Apr 2011 19:54:48 -0400
Subject: [Freeswitch-users] webphone app
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
Message-ID: <32EF1A658EFC4E5393D8D1A3A486DA31@e1705>

boophone.com
  ----- Original Message ----- 
  From: budi wibowo 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, April 06, 2011 7:42 PM
  Subject: [Freeswitch-users] webphone app


  looking for webphone sip based on flash. 
  any info, please share




  thx


  budi wibowo


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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From bwibowo at gmail.com  Thu Apr  7 04:01:28 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Thu, 7 Apr 2011 07:01:28 +0700
Subject: [Freeswitch-users] webphone app
In-Reply-To: <32EF1A658EFC4E5393D8D1A3A486DA31@e1705>
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
	<32EF1A658EFC4E5393D8D1A3A486DA31@e1705>
Message-ID: <BANLkTikdW-5-hDz9A+0mDid_KNEQNhzf6Q@mail.gmail.com>

thx, but i want to link the webphone to Freeswitch.
not use any body's sip server


thx

budi


On Thu, Apr 7, 2011 at 6:54 AM, Madovsky <infos at madovsky.org> wrote:

>  boophone.com
>
> ----- Original Message -----
> *From:* budi wibowo <bwibowo at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, April 06, 2011 7:42 PM
> *Subject:* [Freeswitch-users] webphone app
>
> looking for webphone sip based on flash.
> any info, please share
>
>
> thx
>
> budi wibowo
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lists at telefaks.de  Thu Apr  7 04:22:38 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 07 Apr 2011 02:22:38 +0200
Subject: [Freeswitch-users] Hylafax server emulation
In-Reply-To: <BANLkTinaftq2FapGp4tM3Ysjr1_Wd8gGbw@mail.gmail.com>
References: <4D9B42D7.4020008@coppice.org>
	<BANLkTinaftq2FapGp4tM3Ysjr1_Wd8gGbw@mail.gmail.com>
Message-ID: <4D9D03CE.1070607@telefaks.de>

I would also love this approach. That way it may be easy to link an
outgoing fax number and local station/header info to an extension.

Best regards
Peter

Kristian Kielhofner schrieb:
> Steve,
>
>   Very cool (and I'm very interested).
>
>   What about using the existing user directory (perhaps with
> additional params) for the accounts and the FreeSWITCH core DB
> (whether SQLite or ODBC) for the jobs, etc?
>
> On Tue, Apr 5, 2011 at 12:27 PM, Steve Underwood <steveu at coppice.org> wrote:
>   
>> Hi,
>>
>> It has always been clear that a HylaFAX compatible FAX job submission
>> server would add considerably to the value of the FAX facilities in
>> Asterisk and Freeswitch, but somehow it hasn't happened until now. I
>> recently found that in 2005 someone produced something fairly basic for
>> Asterisk in Perl, but it doesn't seem to have been well publicised, and
>> it looks like development stalled long ago.
>>
>> I now have the skeleton of HylaFAX compatible FAX job submission server,
>> in C, working. It accepts FAX submissions from sendfax and a couple of
>> the windows HylaFAX clients, though it needs a lot more polishing. Now I
>> need to look at the best thing to do on the Freeswitch side. I aim to
>> make the server maintain its own database of FAX jobs. It will attach to
>> Freeswitch, by ESL; push the jobs through FS; deal with scheduling,
>> retries, etc; and report the final result to the user, just as HylaFAX
>> does. The thing I am rather unsure about is the best way to handle the
>> accounts used to accept FAX jobs? Should I maintain a separate database
>> of FAX accounts, or hook into an existing database? I would welcome
>> suggestions for what would be the most useful approach.
>>
>> 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
>>
>>     
>
>
>
>   


-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
mailto:lists (att) telefaks.de
Internet: www.telefaks.de




From manavid at gmail.com  Thu Apr  7 05:17:34 2011
From: manavid at gmail.com (Moe Navid)
Date: Wed, 6 Apr 2011 18:17:34 -0700
Subject: [Freeswitch-users] webphone app
In-Reply-To: <BANLkTikdW-5-hDz9A+0mDid_KNEQNhzf6Q@mail.gmail.com>
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
	<32EF1A658EFC4E5393D8D1A3A486DA31@e1705>
	<BANLkTikdW-5-hDz9A+0mDid_KNEQNhzf6Q@mail.gmail.com>
Message-ID: <BANLkTi=73yyZ4Chh8OwfiJhNXAVXnrZ8MA@mail.gmail.com>

I tried this about a year ago, it was ok

http://code.google.com/p/red5phone

On Wed, Apr 6, 2011 at 5:01 PM, budi wibowo <bwibowo at gmail.com> wrote:

> thx, but i want to link the webphone to Freeswitch.
> not use any body's sip server
>
>
> thx
>
> budi
>
>
> On Thu, Apr 7, 2011 at 6:54 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  boophone.com
>>
>> ----- Original Message -----
>> *From:* budi wibowo <bwibowo at gmail.com>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Wednesday, April 06, 2011 7:42 PM
>> *Subject:* [Freeswitch-users] webphone app
>>
>> looking for webphone sip based on flash.
>> any info, please share
>>
>>
>> thx
>>
>> budi wibowo
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Apr  7 05:30:38 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 6 Apr 2011 20:30:38 -0500
Subject: [Freeswitch-users] BUG FIX: "Buffer size sanity check failed!"
 drops FAX receiving unneeded
In-Reply-To: <BANLkTi=sZwvZw+TJV0W1tMJOc38ytv-ddw@mail.gmail.com>
References: <4D998A5A.6080901@priv.de>
	<BANLkTimSg1hmwtCUwfJkA_DRAgULW32QBQ@mail.gmail.com>
	<BANLkTi=sZwvZw+TJV0W1tMJOc38ytv-ddw@mail.gmail.com>
Message-ID: <BANLkTinAe5vrZDDHEueFVpaHJruzjUL20w@mail.gmail.com>

I think I have a fix for it.
The patch provided would lead to memory corruption by feeding data to
the codec decoder that probably is not even really audio data with a
length that would cause a buffer overflow.

Instead when this happens I think I can write out all zeros to the
buffer at the typical packet len and return that.
Since these frames really should be ignored anyway.

I would like to see the pcaps nevertheless so I can identify why it happens.


On Wed, Apr 6, 2011 at 5:15 PM, Matt Paine
<mattzerah+freeswitch at gmail.com> wrote:
> I can second this behaviour... It wont be the weekend until I can actually
> get any packet captures for the unpatched code, I can certianly find some
> full freeswitch console logs if that will help.
> Has a JIRA bug been filed for this yet? What is the number so I can
> contribute?
> Matt.
>
> On 7 April 2011 02:05, Anthony Minessale <anthony.minessale at gmail.com>
> wrote:
>>
>> Can you please get a pcap of a single call (without your patch) as
>> well as a full capture of the freeswitch console logs and post it to
>> JIRA.
>>
>> Are you using the described TDM inside FreeSWITCH or is this a SIP
>> call from an Asterisk machine?
>>
>>
>>
>> On Mon, Apr 4, 2011 at 4:07 AM, Markus Mueller <freeswitch at priv.de> wrote:
>> > Hello FreeSwitch users and programmers,
>> >
>> > I found a problem on receiving faxes and want to share a working patch
>> > for this. The problem is that on receiving a fax, it is unneeded aborted
>> > by a sanity check. Sanity checks are fine, but a unneeded abort instead
>> > of a warning is in productive versions not the best solution.
>> >
>> > The message apearing is:
>> >
>> > 2011-04-04 10:44:52.060860 [CRIT] switch_core_codec.c:660 Buffer size
>> > sanity check failed!
>> >
>> > which is normaly aborting in receiving the fax. Simply decreasing this
>> > fault to a warning let the server receive the fax without any problems.
>> > After the patch the message apears up to five times per fax before the
>> > fax is beeing accepted. I am using libpri with the three HFC ISDN Cards
>> > and the DAHDI from Debian Squeeze 6.0. For more informations about my
>> > hardware just write me an email.
>> >
>> > Regards,
>> > Markus Mueller
>> > http://projekte.priv.de/
>> >
>> > root at sip:/usr/local/src/freeswitch/src# diff -U 4 switch_core_codec.c*
>> > --- switch_core_codec.c 2011-03-14 10:49:17.000000000 +0100
>> > +++ switch_core_codec.c.org ? ? 2011-03-14 10:47:02.000000000 +0100
>> > @@ -657,9 +657,9 @@
>> > ? ? ? ? ? ? ? ? uint32_t frames = encoded_data_len /
>> > codec->implementation->encoded_bytes_per_packet;
>> >
>> > ? ? ? ? ? ? ? ? if (frames &&
>> > codec->implementation->decoded_bytes_per_packet * frames >
>> > *decoded_data_len) {
>> > ? ? ? ? ? ? ? ? ? ? ? ? switch_log_printf(SWITCH_CHANNEL_LOG,
>> > SWITCH_LOG_CRIT, "Buffer size sanity check failed!\n");
>> > - ? ? ? ? ? ? ? ? ? ? ? // return SWITCH_STATUS_GENERR;
>> > + ? ? ? ? ? ? ? ? ? ? ? return SWITCH_STATUS_GENERR;
>> > ? ? ? ? ? ? ? ? }
>> > ? ? ? ? }
>> >
>> > ? ? ? ? if (codec->mutex) switch_mutex_lock(codec->mutex);
>> > root at sip:/usr/local/src/freeswitch/src#
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> 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
>> 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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From infos at madovsky.org  Thu Apr  7 07:35:26 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 6 Apr 2011 23:35:26 -0400
Subject: [Freeswitch-users] webphone app
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com><32EF1A658EFC4E5393D8D1A3A486DA31@e1705>
	<BANLkTikdW-5-hDz9A+0mDid_KNEQNhzf6Q@mail.gmail.com>
Message-ID: <93D7DF166EB94D98AEA5C03368283113@e1705>

this work with freeswitch
  ----- Original Message ----- 
  From: budi wibowo 
  To: FreeSWITCH Users Help 
  Sent: Wednesday, April 06, 2011 8:01 PM
  Subject: Re: [Freeswitch-users] webphone app


  thx, but i want to link the webphone to Freeswitch.
  not use any body's sip server




  thx


  budi 



  On Thu, Apr 7, 2011 at 6:54 AM, Madovsky <infos at madovsky.org> wrote:

    boophone.com
      ----- Original Message ----- 
      From: budi wibowo 
      To: FreeSWITCH Users Help 
      Sent: Wednesday, April 06, 2011 7:42 PM
      Subject: [Freeswitch-users] webphone app


      looking for webphone sip based on flash.  
      any info, please share




      thx


      budi wibowo


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


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



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






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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From acrow at integrafin.co.uk  Thu Apr  7 11:51:58 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Thu, 07 Apr 2011 08:51:58 +0100
Subject: [Freeswitch-users] Freeswitchsolutions.com?
Message-ID: <4D9D6D1E.4000503@integrafin.co.uk>

Hi list, Anthony,

We are investigating a possible move from a proprietary phone system to 
Freeswitch. Advertised on the FS site is the above address which appears 
to say it provides commercial support for FS.

In the last two weeks both me and a colleague filled in the contact form 
asking about prices, SLAs etc but have had no response. Is the company 
real or is there a problem with the contact form?

Thanks

Alex

-- 
This message is intended only for the addressee and may contain
confidential information.  Unless you are that person, you may not
disclose its contents or use it in any way and are requested to delete
the message along with any attachments and notify us immediately.

"Transact" is operated by Integrated Financial Arrangements plc
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Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)



From freeswitch at peely.com  Thu Apr  7 12:33:56 2011
From: freeswitch at peely.com (peely)
Date: Thu, 7 Apr 2011 01:33:56 -0700 (PDT)
Subject: [Freeswitch-users] webphone app
In-Reply-To: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
Message-ID: <1302165236411-6249102.post@n2.nabble.com>

http://www.flashphoner.com/

It's commercial, but well worth it and very stable, unlike red5. It's built
on the wowza media server.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/webphone-app-tp6248061p6249102.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From avi at avimarcus.net  Thu Apr  7 12:53:11 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 7 Apr 2011 11:53:11 +0300
Subject: [Freeswitch-users] webphone app
In-Reply-To: <1302165236411-6249102.post@n2.nabble.com>
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
	<1302165236411-6249102.post@n2.nabble.com>
Message-ID: <BANLkTi=myJN6oX1EcMVZpCa71+dvz-g5tg@mail.gmail.com>

There's also phono.com - I don't think it quite worked for me in chrome,
ymmv.
-Avi

On Thu, Apr 7, 2011 at 11:33 AM, peely <freeswitch at peely.com> wrote:

> http://www.flashphoner.com/
>
> It's commercial, but well worth it and very stable, unlike red5. It's built
> on the wowza media server.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/webphone-app-tp6248061p6249102.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
>
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From fdelawarde at wirelessmundi.com  Thu Apr  7 13:11:09 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Thu, 07 Apr 2011 11:11:09 +0200
Subject: [Freeswitch-users] Freeswitch
In-Reply-To: <BANLkTimHtysU_z2rbns+asa5X2nXEMHrew@mail.gmail.com>
References: <BANLkTikCO-30PnPTgkam7+t4fo9VEAH=pQ@mail.gmail.com>
	<BANLkTimQvgj7bU1_bxSi2VWwYz8WzcZJ4g@mail.gmail.com>
	<BANLkTimHtysU_z2rbns+asa5X2nXEMHrew@mail.gmail.com>
Message-ID: <1302167469.2186.545.camel@luna.tc.commsmundi.com>

On Wed, 2011-04-06 at 17:33 +0200, Giovanni Maruzzelli wrote:
> I've heard of Jim Strlbinsky in Pallaawooka that's using FreeSWITCH
> running on an EEEpc reaching 10.000.000 cps.

Is that real or a late april fools joke? :-)

Fran?ois.



From freeswitch at peely.com  Thu Apr  7 14:01:39 2011
From: freeswitch at peely.com (peely)
Date: Thu, 7 Apr 2011 03:01:39 -0700 (PDT)
Subject: [Freeswitch-users] webphone app
In-Reply-To: <BANLkTi=myJN6oX1EcMVZpCa71+dvz-g5tg@mail.gmail.com>
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
	<1302165236411-6249102.post@n2.nabble.com>
	<BANLkTi=myJN6oX1EcMVZpCa71+dvz-g5tg@mail.gmail.com>
Message-ID: <1302170499206-6249353.post@n2.nabble.com>

I believe Phono needs the Voxeo cloud in order to run, although for now this
appears to be free if you are terminating to SIP.


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/webphone-app-tp6248061p6249353.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From shamun.toha at gmail.com  Thu Apr  7 14:31:25 2011
From: shamun.toha at gmail.com (Shamun toha md)
Date: Thu, 7 Apr 2011 12:31:25 +0200
Subject: [Freeswitch-users] What FreeSwitch version i am using after
	updating git pull?
Message-ID: <BANLkTin+LQ1nqZkf89bt9-0X7=JyE4CvEA@mail.gmail.com>

How do we know if i am using the latest Git version (nightly builids)? How
to know if i am really using 1.0.7 or 1.0.6 oldest?

So far tried as >

git pull && make current >
make current (showed some error) >
make clean modwipe (completed successfully) >
./configure && make & make install >
/usr/local/freeswitch/bin/freeswitch >
> preesed F12 shows: > FreeSWITCH Version 1.0.head (git-828960a 2010-09-25
12-51-42 -0500)

Let me know plz, is this the 1.0.7 ?
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From peter.olsson at visionutveckling.se  Thu Apr  7 14:43:31 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 7 Apr 2011 12:43:31 +0200
Subject: [Freeswitch-users] What FreeSwitch version i am using
	after	updating git pull?
In-Reply-To: <BANLkTin+LQ1nqZkf89bt9-0X7=JyE4CvEA@mail.gmail.com>
References: <BANLkTin+LQ1nqZkf89bt9-0X7=JyE4CvEA@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C4D307D0@cooper>

This does not seem to be pulled from git. Please follow the instructions on the wiki http://wiki.freeswitch.org/wiki/Installation_Guide.

More or less like this;

git clone git://git.freeswitch.org/freeswitch.git freeswitch (only first time you checkout)
cd freeswitch
./bootstrap.sh (only first time)
./configure (only first time)
make current

Next time you just do git pull && make current


Mvh
Peter Olsson

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Shamun toha md
Skickat: den 7 april 2011 12:31
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] What FreeSwitch version i am using after updating git pull?

How do we know if i am using the latest Git version (nightly builids)? How to know if i am really using 1.0.7 or 1.0.6 oldest?

So far tried as >

git pull && make current >
make current (showed some error) >
make clean modwipe (completed successfully) >
./configure && make & make install >
/usr/local/freeswitch/bin/freeswitch >
> preesed F12 shows: > FreeSWITCH Version 1.0.head (git-828960a 2010-09-25 12-51-42 -0500)

Let me know plz, is this the 1.0.7 ?


!DSPAM:4d9d937a32761735852961!
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From gmaruzz at gmail.com  Thu Apr  7 14:44:59 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 7 Apr 2011 12:44:59 +0200
Subject: [Freeswitch-users] What FreeSwitch version i am using after
 updating git pull?
In-Reply-To: <BANLkTin+LQ1nqZkf89bt9-0X7=JyE4CvEA@mail.gmail.com>
References: <BANLkTin+LQ1nqZkf89bt9-0X7=JyE4CvEA@mail.gmail.com>
Message-ID: <BANLkTik_r+2AuwM=VQbyLSxQAJSERo3NiQ@mail.gmail.com>

better if you do:

git pull ; ./bootstrap.sh && ./configure && make install

On Thu, Apr 7, 2011 at 12:31 PM, Shamun toha md <shamun.toha at gmail.com> wrote:
> How do we know if i am using the latest Git version (nightly builids)? How
> to know if i am really using 1.0.7 or 1.0.6 oldest?
>
> So far tried as >
>
> git pull && make current >
> make current (showed some error) >
> make clean modwipe (completed successfully) >
> ./configure && make & make install >
> /usr/local/freeswitch/bin/freeswitch >
>> preesed F12 shows: > FreeSWITCH Version 1.0.head (git-828960a 2010-09-25
>> 12-51-42 -0500)
>
> Let me know plz, is this the 1.0.7 ?
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sameer2k3t at gmail.com  Thu Apr  7 15:04:40 2011
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Thu, 7 Apr 2011 16:04:40 +0500
Subject: [Freeswitch-users] codec negotiation
Message-ID: <BANLkTi=mD2mv726VZrmMO+ud2_tMgjf4EA@mail.gmail.com>

hello every 1
i need help regarding codec negotiation
I set abs codec string in my dialplan $xml_output .='<action
application="export" data="nolocal:absolute_codec_string=PCMA,PCMU"/>';

but still leg B is carrying the same codecs as leg A

disable_transcoding is false in my internal sip profile
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From steveayre at gmail.com  Thu Apr  7 16:55:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 7 Apr 2011 13:55:33 +0100
Subject: [Freeswitch-users] codec negotiation
In-Reply-To: <BANLkTi=mD2mv726VZrmMO+ud2_tMgjf4EA@mail.gmail.com>
References: <BANLkTi=mD2mv726VZrmMO+ud2_tMgjf4EA@mail.gmail.com>
Message-ID: <BANLkTini2w_5uc-LZRJxy4VwdcUKcthk7w@mail.gmail.com>

Can you show the debug-level log output including siptrace?

-Steve


On 7 April 2011 12:04, Sameer Khan <sameer2k3t at gmail.com> wrote:

> hello every 1
> i need help regarding codec negotiation
> I set abs codec string in my dialplan $xml_output .='<action
> application="export" data="nolocal:absolute_codec_string=PCMA,PCMU"/>';
>
> but still leg B is carrying the same codecs as leg A
>
> disable_transcoding is false in my internal sip profile
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Thu Apr  7 16:57:27 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 7 Apr 2011 13:57:27 +0100
Subject: [Freeswitch-users] What FreeSwitch version i am using after
 updating git pull?
In-Reply-To: <BANLkTin+LQ1nqZkf89bt9-0X7=JyE4CvEA@mail.gmail.com>
References: <BANLkTin+LQ1nqZkf89bt9-0X7=JyE4CvEA@mail.gmail.com>
Message-ID: <BANLkTikcgxsYN5nVZ1FH2q3CfMj2db119g@mail.gmail.com>

There's not really a 1.0.7 - it's the nightly snapshot of the latest git
version.

FreeSWITCH Version 1.0.head (git-828960a 2010-09-25 12-51-42 -0500)
>

That indicates you're on a rather old git version still. You can check the
latest at http://fisheye.freeswitch.org/changelog/freeswitch.git

-Steve


On 7 April 2011 11:31, Shamun toha md <shamun.toha at gmail.com> wrote:

> How do we know if i am using the latest Git version (nightly builids)? How
> to know if i am really using 1.0.7 or 1.0.6 oldest?
>
> So far tried as >
>
> git pull && make current >
> make current (showed some error) >
> make clean modwipe (completed successfully) >
> ./configure && make & make install >
> /usr/local/freeswitch/bin/freeswitch >
> > preesed F12 shows: > FreeSWITCH Version 1.0.head (git-828960a 2010-09-25
> 12-51-42 -0500)
>
> Let me know plz, is this the 1.0.7 ?
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Apr  7 19:09:12 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Apr 2011 10:09:12 -0500
Subject: [Freeswitch-users] Freeswitchsolutions.com?
In-Reply-To: <4D9D6D1E.4000503@integrafin.co.uk>
References: <4D9D6D1E.4000503@integrafin.co.uk>
Message-ID: <BANLkTi=sbn=cxXLqcj2M286+zii5NdvuyA@mail.gmail.com>

Indeed. Also consulting at freeswitch.org is another point of contact.



On Thu, Apr 7, 2011 at 2:51 AM, Alex Crow <acrow at integrafin.co.uk> wrote:
> Hi list, Anthony,
>
> We are investigating a possible move from a proprietary phone system to
> Freeswitch. Advertised on the FS site is the above address which appears
> to say it provides commercial support for FS.
>
> In the last two weeks both me and a colleague filled in the contact form
> asking about prices, SLAs etc but have had no response. Is the company
> real or is there a problem with the contact form?
>
> Thanks
>
> Alex
>
> --
> This message is intended only for the addressee and may contain
> confidential information. ?Unless you are that person, you may not
> disclose its contents or use it in any way and are requested to delete
> the message along with any attachments and notify us immediately.
>
> "Transact" is operated by Integrated Financial Arrangements plc
> Domain House, 5-7 Singer Street, London ?EC2A 4BQ
> Tel: (020) 7608 4900 Fax: (020) 7608 5300
> (Registered office: as above; Registered in England and Wales under number: 3727592)
> Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From valery.kalinin at gmail.com  Thu Apr  7 05:43:46 2011
From: valery.kalinin at gmail.com (Valery Kalinin)
Date: Thu, 7 Apr 2011 07:43:46 +0600
Subject: [Freeswitch-users] Cannot compile freetdm!
Message-ID: <BANLkTik3PXUY_=4hg1DQ23iGtGipxpG50Q@mail.gmail.com>

# cd /usr/local/freeswitch/libs/freetdm
# ./configure --with-libisdn
# make

bla-bla-bla

cc1: warnings being treated as errors
src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'ftdm_isdn_931_34':
src/ftmod/ftmod_isdn/ftmod_isdn.c:982: warning: unused variable 'cplen'
src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'isdn_configure_span':
src/ftmod/ftmod_isdn/ftmod_isdn.c:2794: warning: passing argument 2 of
'Q931SetLogCB' from incompatible pointer type
make: *** [ftmod_isdn_la-ftmod_isdn.lo] Error 1

Why?
libisdn-0.0.1 installed


From kevygreen at gmail.com  Thu Apr  7 06:10:14 2011
From: kevygreen at gmail.com (Kevin Green)
Date: Wed, 6 Apr 2011 22:10:14 -0400
Subject: [Freeswitch-users] webphone app
In-Reply-To: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
Message-ID: <BANLkTinr=i8tfyadpJsQwekkVb-yRnaQtg@mail.gmail.com>

Take a look at http://phono.com/ it's under the wings of Voxeo.

-Kevin

On Wed, Apr 6, 2011 at 7:42 PM, budi wibowo <bwibowo at gmail.com> wrote:

> looking for webphone sip based on flash.
> any info, please share
>
>
> thx
>
> budi wibowo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kaushalshriyan at gmail.com  Thu Apr  7 06:16:12 2011
From: kaushalshriyan at gmail.com (Kaushal Shriyan)
Date: Thu, 7 Apr 2011 07:46:12 +0530
Subject: [Freeswitch-users] Freeswitch
In-Reply-To: <BANLkTimHtysU_z2rbns+asa5X2nXEMHrew@mail.gmail.com>
References: <BANLkTikCO-30PnPTgkam7+t4fo9VEAH=pQ@mail.gmail.com>
	<BANLkTimQvgj7bU1_bxSi2VWwYz8WzcZJ4g@mail.gmail.com>
	<BANLkTimHtysU_z2rbns+asa5X2nXEMHrew@mail.gmail.com>
Message-ID: <BANLkTi=V=bLF1xU73zrcsu91xpZNCyrvjg@mail.gmail.com>

On Wed, Apr 6, 2011 at 9:03 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> On Wed, Apr 6, 2011 at 8:44 AM, Kaushal Shriyan
> <kaushalshriyan at gmail.com> wrote:
> > typo it was meant for FreeSwitch
> >
> > On Wed, Apr 6, 2011 at 11:58 AM, Kaushal Shriyan <
> kaushalshriyan at gmail.com>
> > wrote:
> >>
> >> Hi,
> >> I have couple of questions regarding Asterisk.
> >> a) Does it has Automated Dialing Feature like dialing 1000 and 1000 of
> >> phone numbers?
>
>
> With FreeSWITCH you can voice spam the entire globe.
> I've heard of Jim Strlbinsky in Pallaawooka that's using FreeSWITCH
> running on an EEEpc reaching 10.000.000 cps.
> Yes, that's ten million calls per seconds on an EEEpc!
> (OK, that's using g711. If you use a cpu hungry compressed format to
> save on bandwidth, let's say speex or celt, then you can't expect much
> more than one million calls per seconds on an EEEpc).
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
Hi Giovanni

Please suggest/guide me understand about the below questions.

b) Does it Support VoiceXML ?
c) What PRI Card is recommended for using FreeSwitch ?

Thanks

Kaushal
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From valery.kalinin at gmail.com  Thu Apr  7 13:10:16 2011
From: valery.kalinin at gmail.com (Valery K)
Date: Thu, 7 Apr 2011 02:10:16 -0700 (PDT)
Subject: [Freeswitch-users]  Cannot compile freetdm
Message-ID: <31340769.post@talk.nabble.com>


Enter:
# cd /usr/local/freeswitch/libs/freetdm
# ./configure --with-libisdn
# make

... compile ...

cc1: warnings being treated as errors
src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'ftdm_isdn_931_34':
src/ftmod/ftmod_isdn/ftmod_isdn.c:982: warning: unused variable 'cplen'
src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'isdn_configure_span':
src/ftmod/ftmod_isdn/ftmod_isdn.c:2794: warning: passing argument 2 of
'Q931SetLogCB' from incompatible pointer type
make: *** [ftmod_isdn_la-ftmod_isdn.lo] Error 1

Why?
libisdn-0.0.1 installed
-- 
View this message in context: http://old.nabble.com/Cannot-compile-freetdm-tp31340769p31340769.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From eric at loopfx.com  Thu Apr  7 19:11:06 2011
From: eric at loopfx.com (Eric Beard)
Date: Thu, 7 Apr 2011 11:11:06 -0400
Subject: [Freeswitch-users] FS does not relay BYE
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DC971ED2@doctorbob.loop.local>

Hello,

I have just started using freeSwitch as a way to terminate calls from Microsoft Speech Server to voip gateways.  I have almost everything working with a few exceptions.  One of the problems I am having is that the final BYE issued by the terminator does not get relayed back to MSS, so MSS keeps the call open for an additional minute, then issues its own BYE, which freeSwitch can't match up to a call because it tore the call down already.

The sequence:


-          MSS running on my machine originates a call, sends INVITE to freeSwitch running on a separate machine, with an internal and external NIC.

-          freeSwitch relays the INVITE to the gateway (in this case Affinity, but I get the same behavior with other gateways).

-          My cell phone rings, I pick it up, then hang up the call.

-          The gateway issues a BYE to freeSwitch, freeSwitch says OK and tears down the call without passing on the BYE.

If I originate a call from my machine with a soft phone, it works fine.  The only difference I can see is that the soft phone uses UDP, while MSS only talks SIP over TCP.

I have pasted logs for the session at http://pastebin.freeswitch.org/16037.

Thanks!

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

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From eric at loopfx.com  Thu Apr  7 19:22:25 2011
From: eric at loopfx.com (Eric Beard)
Date: Thu, 7 Apr 2011 11:22:25 -0400
Subject: [Freeswitch-users] Recording transfer audio
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DC971EE9@doctorbob.loop.local>

Hello,

I asked this question yesterday over irc.  My apologies if someone answered there already, my irc client kept crashing so I might have missed it.

I am using freeSwitch to terminate calls that originate on the local network from an IVR system. I have it working with several different voip terminators, and I can record sessions successfully, except when the call gets transferred. When I do a transfer, I see two WAV files, one of which is the original session in which I can hear the call up until the transfer. The other file is all silence, which I assume is the session up to the point where the INVITE is answered. But I get no audio of the conversation after the transfer. My suspicion is that freeSwitch is not hairpinning the audio, so it does not have access to the RTP.

I pasted everything from the console at http://pastebin.freeswitch.org/16027.
Thanks!

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

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From fieldpeak at gmail.com  Thu Apr  7 19:55:31 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 7 Apr 2011 23:55:31 +0800
Subject: [Freeswitch-users] How to limit the max number of registration users
Message-ID: <BANLkTikrqjFR66mgzHsXUUEDGz82ed0Baw@mail.gmail.com>

Could anyone help advise how to limit the max number of registration users?
thanks if any advise.

Regards,
Charles
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From joegen at opensipstack.org  Thu Apr  7 20:05:31 2011
From: joegen at opensipstack.org (Joegen E. Baclor)
Date: Fri, 08 Apr 2011 00:05:31 +0800
Subject: [Freeswitch-users] Transfer attempt for a previously a replaced
 call fails
In-Reply-To: <4D9BAD82.3000205@opensipstack.org>
References: <4D9A9A42.2070804@opensipstack.org>	<BANLkTimsX4oJStWYDvz2yhPvf1e9F7kezw@mail.gmail.com>	<4D9AADEB.7040803@opensipstack.org>	<BANLkTi=XdyGKZcLmHmqe1_kupp5n7ib6KQ@mail.gmail.com>
	<4D9BAD82.3000205@opensipstack.org>
Message-ID: <4D9DE0CB.7000402@opensipstack.org>

Just bumping this thread.   If I need to provide more info, just let me 
know.  Or if this is a known bug and a fix is due for a future version 
that is also acceptable.

On 04/06/2011 08:02 AM, Joegen E. Baclor wrote:
> I'll keep that in mind.  If more information is needed to get into the 
> bottom of this, I will happily oblige.  Thanks for helping.
>
> On 04/06/2011 03:09 AM, Michael Collins wrote:
>> I'll have to defer to those more experienced than I in such matters. 
>> However, I can offer two tips:
>>
>> #1 - turn off the crazy sofia debugging - it's just noise. All you 
>> need to do to enable SIP trace is "sofia global siptrace on"
>> #2 - when you pastebin the console output use the FreeSWITCH log 
>> syntax highlighting - it makes it *much* easier to see what's going on.
>>
>> -MC
>>
>> On Mon, Apr 4, 2011 at 10:51 PM, Joegen E. Baclor 
>> <joegen at opensipstack.org <mailto:joegen at opensipstack.org>> wrote:
>>
>>     Hi Michael,
>>
>>     I have pasted both working and none working logs on pastebin.
>>
>>     FreeSWITCH Version 1.0.7 (hacked-20110326T123355Z)
>>     working: http://pastebin.freeswitch.org/16008
>>     not working: http://pastebin.freeswitch.org/16009
>>
>>     The call flow for the working call is
>>     UA1 ->  (FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp)
>>     FSIVRApp knows the uuid of the bridge call.  Pressing # on the
>>     IVR results to a uuid_deflect on the bridged channel.  This works
>>     and call successfully transfers to the new destination.
>>
>>     The call flow for the none working call is
>>
>>     1.  UA1 -> UA2  is in conversation
>>     2.  UA1 puts UA2 on hold
>>
>>     -- start of FS interaction here --
>>
>>     3.  UA1 ->  (FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp) 
>>     (on line 2)
>>     4.  UA1 sends REFER (replacing its call with UA2) to
>>     FSBridgeDialPlan.
>>     5.  Flow is now UA2 ->  ([REPLACED]FSBridgeDialPlan) ->
>>     (SIP-Loopback) -> (FSIVRApp)
>>     6.  UA2 presses #.
>>     7.  IVRApp performs uuid_deflect on FSBridgeDialPlan.
>>     8. FSBridgeDialPlan drops call (no REFER is done)
>>
>>     Thanks for your help.
>>
>>     Joegen
>>
>>
>>     On 04/05/2011 12:35 PM, Michael Collins wrote:
>>>     What do you see on the console when you try this? A console
>>>     debug log with siptrace would go a long way toward figuring out
>>>     what is happening.
>>>
>>>     -MC
>>>
>>>     On Mon, Apr 4, 2011 at 9:27 PM, Joegen E. Baclor
>>>     <joegen at opensipstack.org <mailto:joegen at opensipstack.org>> wrote:
>>>
>>>         Hi List,
>>>
>>>         I have a scenario where a bridged call has been replaced due
>>>         to a
>>>         consultative transfer.  This works pretty well and audio is
>>>         bidirectional.  I have the original uuid of the call in a var
>>>         somewhere.  The trouble begins when I uuid_deflect the
>>>         bridged call once
>>>         again to attempt another transfer.  Sofia disconnects the
>>>         channel.  I am
>>>         using the original uuid of the call (uuid prior to
>>>         replaces).  Is this
>>>         the right way of doing it?
>>>
>>>         Joegen
>>>
>>>         _______________________________________________
>>>         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

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From fieldpeak at gmail.com  Thu Apr  7 20:11:07 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Fri, 8 Apr 2011 00:11:07 +0800
Subject: [Freeswitch-users] FS -How to limit the max number of registration
	users
In-Reply-To: <BANLkTi=+4wCkiTCHgoLD+avkjJhVkjbZYw@mail.gmail.com>
References: <BANLkTi=+4wCkiTCHgoLD+avkjJhVkjbZYw@mail.gmail.com>
Message-ID: <BANLkTiniKfdR7e7AYLSs4Ghj0pq1V33emA@mail.gmail.com>

Could anyone help advise how to limit the max number of registration users
on FS?
Thanks if any advice.

Regards,
Charles
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From brian at freeswitch.org  Thu Apr  7 20:23:33 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 7 Apr 2011 11:23:33 -0500
Subject: [Freeswitch-users] FS does not relay BYE
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DC971ED2@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DC971ED2@doctorbob.loop.local>
Message-ID: <75934029-74CE-421B-AEDF-DFC994C7D0C4@freeswitch.org>

sofia loglevel all 9

you'll see why its not.

/b

On Apr 7, 2011, at 10:11 AM, Eric Beard wrote:

> Hello,
> 
> I have just started using freeSwitch as a way to terminate calls from Microsoft Speech Server to voip gateways.  I have almost everything working with a few exceptions.  One of the problems I am having is that the final BYE issued by the terminator does not get relayed back to MSS, so MSS keeps the call open for an additional minute, then issues its own BYE, which freeSwitch can't match up to a call because it tore the call down already.
> 
> The sequence:
> 
> 
> -          MSS running on my machine originates a call, sends INVITE to freeSwitch running on a separate machine, with an internal and external NIC.
> 
> -          freeSwitch relays the INVITE to the gateway (in this case Affinity, but I get the same behavior with other gateways).
> 
> -          My cell phone rings, I pick it up, then hang up the call.
> 
> -          The gateway issues a BYE to freeSwitch, freeSwitch says OK and tears down the call without passing on the BYE.
> 
> If I originate a call from my machine with a soft phone, it works fine.  The only difference I can see is that the soft phone uses UDP, while MSS only talks SIP over TCP.
> 
> I have pasted logs for the session at http://pastebin.freeswitch.org/16037.
> 
> Thanks!
> 
> -----------------------
> Eric Z. Beard, CTO
> Loop LLC
> w (877) 850-2010 x9249
> m (727) 776-2768
> eric at loopfx.com<mailto:eric at loopfx.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 eric at loopfx.com  Thu Apr  7 20:46:16 2011
From: eric at loopfx.com (Eric Beard)
Date: Thu, 7 Apr 2011 12:46:16 -0400
Subject: [Freeswitch-users] FS does not relay BYE
In-Reply-To: <75934029-74CE-421B-AEDF-DFC994C7D0C4@freeswitch.org>
References: <D4A666272A214C4DBBABFE569CA34CA401DC971ED2@doctorbob.loop.local>
	<75934029-74CE-421B-AEDF-DFC994C7D0C4@freeswitch.org>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DC971F8D@doctorbob.loop.local>

Here are the logs from the OK that FS sends to the terminator.  I don't see anything obvious.

send 557 bytes to udp/[69.30.55.34]:5060 at 12:36:43.431454:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 69.30.55.34;branch=z9hG4bKb10d.db340dd6.0
   Via: SIP/2.0/UDP 69.30.55.46:5060;branch=z9hG4bK19032810
   From: <sip:17277762768 at 69.30.55.34>;tag=as1bf91603
   To: "18778502010" <sip:18778502010 at 97.76.110.215>;tag=DtZyD89QQvjUj
   Call-ID: 7fc47854-dbb6-122e-ad9a-0014220d7aff
   CSeq: 103 BYE
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-8c5586b 2011-04-01 14-22-43 -0500
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Content-Length: 0
   
   ------------------------------------------------------------------------
nta: sent 200 OK for BYE (103)
nua(0x7fe6a80625f0): removing session usage
nua(0x7fe6a80625f0): call state changed: ready -> terminated
nua(0x7fe6a80625f0): event i_state 200 Session Terminated
nua(0x7fe6a80625f0): event i_terminated 200 Session Terminated
soa_destroy(static::0x7fe6b005b390) called
nta_leg_destroy(0x7fe6b005ab50)
nua(0x7fe6a80625f0): recv signal r_destroy
nta_leg_destroy((nil))
2011-04-07 12:36:43.440470 [DEBUG] switch_ivr_bridge.c:501 sofia/external/17277762768 ending bridge by request from read function
2011-04-07 12:36:43.440470 [DEBUG] switch_ivr_bridge.c:495 sofia/external/17277762768 ending bridge by request from write function
2011-04-07 12:36:43.440470 [DEBUG] switch_ivr_bridge.c:582 BRIDGE THREAD DONE [sofia/external/17277762768]
2011-04-07 12:36:43.440470 [DEBUG] switch_ivr_bridge.c:602 Send signal sofia/internal/18778502010 at bert [BREAK]
2011-04-07 12:36:43.440470 [DEBUG] switch_ivr_bridge.c:582 BRIDGE THREAD DONE [sofia/internal/18778502010 at bert]
2011-04-07 12:36:43.440470 [DEBUG] switch_ivr_bridge.c:602 Send signal sofia/external/17277762768 [BREAK]
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:374 (sofia/external/17277762768) State EXCHANGE_MEDIA going to sleep
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:325 (sofia/external/17277762768) Running State Change CS_HANGUP
2011-04-07 12:36:43.440470 [DEBUG] switch_ivr_bridge.c:1306 sofia/external/17277762768 skip receive message [UNBRIDGE] (channel is hungup already)
2011-04-07 12:36:43.440470 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/18778502010 at bert [BREAK]
2011-04-07 12:36:43.440470 [NOTICE] switch_core_state_machine.c:189 sofia/internal/18778502010 at bert has executed the last dialplan instruction, hanging up.
2011-04-07 12:36:43.440470 [DEBUG] switch_channel.c:2563 (sofia/internal/18778502010 at bert) Callstate Change ACTIVE -> HANGUP
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:565 (sofia/external/17277762768) State HANGUP
2011-04-07 12:36:43.440470 [DEBUG] mod_sofia.c:451 sofia/external/17277762768 Overriding SIP cause 480 with 200 from the other leg
2011-04-07 12:36:43.440470 [DEBUG] mod_sofia.c:457 Channel sofia/external/17277762768 hanging up, cause: NORMAL_CLEARING
2011-04-07 12:36:43.440470 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/internal/18778502010 at bert [CS_EXECUTE] [NORMAL_CLEARING]
2011-04-07 12:36:43.440470 [DEBUG] switch_channel.c:2579 Send signal sofia/internal/18778502010 at bert [KILL]
2011-04-07 12:36:43.440470 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/18778502010 at bert [BREAK]
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/18778502010 at bert) State EXECUTE going to sleep
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/18778502010 at bert) Running State Change CS_HANGUP
2011-04-07 12:36:43.440470 [DEBUG] switch_ivr_async.c:936 Stop recording file /usr/local/freeswitch/recordings/2011-04-07-12-36-27_17277762768_18778502010.wav
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:46 sofia/external/17277762768 Standard HANGUP, cause: NORMAL_CLEARING
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:565 (sofia/external/17277762768) State HANGUP going to sleep
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:356 (sofia/external/17277762768) State Change CS_HANGUP -> CS_REPORTING
2011-04-07 12:36:43.440470 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/17277762768 [BREAK]
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:325 (sofia/external/17277762768) Running State Change CS_REPORTING
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:625 (sofia/external/17277762768) State REPORTING
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:53 sofia/external/17277762768 Standard REPORTING, cause: NORMAL_CLEARING
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:625 (sofia/external/17277762768) State REPORTING going to sleep
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:350 (sofia/external/17277762768) State Change CS_REPORTING -> CS_DESTROY
2011-04-07 12:36:43.440470 [DEBUG] switch_core_media_bug.c:439 Removing BUG from sofia/internal/18778502010 at bert
2011-04-07 12:36:43.440470 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/17277762768 [BREAK]
2011-04-07 12:36:43.440470 [DEBUG] switch_core_session.c:1288 Session 2 (sofia/external/17277762768) Locked, Waiting on external entities
2011-04-07 12:36:43.440470 [NOTICE] switch_core_session.c:1306 Session 2 (sofia/external/17277762768) Ended
2011-04-07 12:36:43.440470 [NOTICE] switch_core_session.c:1308 Close Channel sofia/external/17277762768 [CS_DESTROY]
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:454 (sofia/external/17277762768) Callstate Change HANGUP -> DOWN
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:457 (sofia/external/17277762768) Running State Change CS_DESTROY
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:467 (sofia/external/17277762768) State DESTROY
2011-04-07 12:36:43.440470 [DEBUG] mod_sofia.c:362 sofia/external/17277762768 SOFIA DESTROY
2011-04-07 12:36:43.440470 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/18778502010 at bert) State HANGUP
2011-04-07 12:36:43.440470 [DEBUG] mod_sofia.c:451 sofia/internal/18778502010 at bert Overriding SIP cause 480 with 200 from the other leg
2011-04-07 12:36:43.440470 [DEBUG] mod_sofia.c:457 Channel sofia/internal/18778502010 at bert hanging up, cause: NORMAL_CLEARING
2011-04-07 12:36:43.440470 [DEBUG] mod_sofia.c:500 Sending BYE to sofia/internal/18778502010 at bert
nua: nua_bye: entering
nua(0x7fe6b00839e0): sent signal r_bye
nua(0x7fe6b00839e0): recv signal r_bye
nua: nua_stack_set_params: entering
soa_set_params(static::0x7fe6b004be90, ...) called
soa_terminate(static::0x7fe6b004be90) called
soa_init_offer_answer(static::0x7fe6b004be90) called
nta: selecting scheme sip
tport_tsend(0x7660d0) tpn = Tcp/10.1.0.17:58370
tport_resolve addrinfo = 10.1.0.17:58370
tport_by_addrinfo(0x7660d0): not found by name Tcp/10.1.0.17:58370
tport_alloc_secondary(0x7660d0): new secondary tport 0x7fe6a806f5d0
tport_base_connect(0x7fe6a806f5d0): connecting to tcp/10.1.0.17:58370/sip
tport(0x7fe6a806f5d0): reset timer
tport_queue(0x7fe6a806f5d0): queueing 0x7fe6a801a950 for tcp/10.1.0.17:58370
nta: sent BYE (10753837) to Tcp/10.1.0.17:58370
tport_pend(0x7fe6a806f5d0): pending 0x7fe6a801a950 for tcp/10.1.0.17:58370 (already 0)
nta: timer set to 32000 ms
2011-04-07 12:36:43.444692 [DEBUG] switch_core_state_machine.c:46 sofia/internal/18778502010 at bert Standard HANGUP, cause: NORMAL_CLEARING
2011-04-07 12:36:43.444692 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/18778502010 at bert) State HANGUP going to sleep
2011-04-07 12:36:43.444692 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/18778502010 at bert) State Change CS_HANGUP -> CS_REPORTING
2011-04-07 12:36:43.444692 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/18778502010 at bert [BREAK]
2011-04-07 12:36:43.444692 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/18778502010 at bert) Running State Change CS_REPORTING
2011-04-07 12:36:43.444692 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/18778502010 at bert) State REPORTING
2011-04-07 12:36:43.444692 [DEBUG] switch_core_state_machine.c:53 sofia/internal/18778502010 at bert Standard REPORTING, cause: NORMAL_CLEARING
2011-04-07 12:36:43.444692 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/18778502010 at bert) State REPORTING going to sleep
2011-04-07 12:36:43.444692 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/18778502010 at bert) State Change CS_REPORTING -> CS_DESTROY
2011-04-07 12:36:43.444692 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/18778502010 at bert [BREAK]
2011-04-07 12:36:43.444692 [DEBUG] switch_core_session.c:1288 Session 1 (sofia/internal/18778502010 at bert) Locked, Waiting on external entities
2011-04-07 12:36:43.444692 [NOTICE] switch_core_session.c:1306 Session 1 (sofia/internal/18778502010 at bert) Ended
2011-04-07 12:36:43.444692 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/18778502010 at bert [CS_DESTROY]
2011-04-07 12:36:43.444692 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/18778502010 at bert) Callstate Change HANGUP -> DOWN
2011-04-07 12:36:43.444692 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/18778502010 at bert) Running State Change CS_DESTROY
2011-04-07 12:36:43.444692 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/18778502010 at bert) State DESTROY
2011-04-07 12:36:43.444692 [DEBUG] mod_sofia.c:362 sofia/internal/18778502010 at bert SOFIA DESTROY
2011-04-07 12:36:43.452463 [DEBUG] switch_nat.c:544 unmapped public port 17454 protocol UDP to localport 17454
2011-04-07 12:36:43.460474 [DEBUG] switch_nat.c:544 unmapped public port 24894 protocol UDP to localport 24894
2011-04-07 12:36:43.468467 [DEBUG] switch_nat.c:544 unmapped public port 17455 protocol UDP to localport 17455
2011-04-07 12:36:43.469757 [DEBUG] switch_core_state_machine.c:60 sofia/external/17277762768 Standard DESTROY
2011-04-07 12:36:43.469757 [DEBUG] switch_core_state_machine.c:467 (sofia/external/17277762768) State DESTROY going to sleep
2011-04-07 12:36:43.476473 [DEBUG] switch_nat.c:544 unmapped public port 24895 protocol UDP to localport 24895
2011-04-07 12:36:43.476473 [DEBUG] switch_core_state_machine.c:60 sofia/internal/18778502010 at bert Standard DESTROY
2011-04-07 12:36:43.476473 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/18778502010 at bert) State DESTROY going to sleep
nta: timer set next to 4896 ms
nta: timer I fired, terminate 200 response
incoming_reclaim_all((nil), (nil), 0x40d67e60)
nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free
nta: timer set next to 22305 ms

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com


-----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, April 07, 2011 12:24 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FS does not relay BYE

sofia loglevel all 9

you'll see why its not.

/b

On Apr 7, 2011, at 10:11 AM, Eric Beard wrote:

> Hello,
> 
> I have just started using freeSwitch as a way to terminate calls from Microsoft Speech Server to voip gateways.  I have almost everything working with a few exceptions.  One of the problems I am having is that the final BYE issued by the terminator does not get relayed back to MSS, so MSS keeps the call open for an additional minute, then issues its own BYE, which freeSwitch can't match up to a call because it tore the call down already.
> 
> The sequence:
> 
> 
> -          MSS running on my machine originates a call, sends INVITE to freeSwitch running on a separate machine, with an internal and external NIC.
> 
> -          freeSwitch relays the INVITE to the gateway (in this case Affinity, but I get the same behavior with other gateways).
> 
> -          My cell phone rings, I pick it up, then hang up the call.
> 
> -          The gateway issues a BYE to freeSwitch, freeSwitch says OK and tears down the call without passing on the BYE.
> 
> If I originate a call from my machine with a soft phone, it works fine.  The only difference I can see is that the soft phone uses UDP, while MSS only talks SIP over TCP.
> 
> I have pasted logs for the session at http://pastebin.freeswitch.org/16037.
> 
> Thanks!
> 
> -----------------------
> Eric Z. Beard, CTO
> Loop LLC
> w (877) 850-2010 x9249
> m (727) 776-2768
> eric at loopfx.com<mailto:eric at loopfx.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 avi at avimarcus.net  Thu Apr  7 22:05:17 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 7 Apr 2011 21:05:17 +0300
Subject: [Freeswitch-users] FS -How to limit the max number of
 registration users
In-Reply-To: <BANLkTiniKfdR7e7AYLSs4Ghj0pq1V33emA@mail.gmail.com>
References: <BANLkTi=+4wCkiTCHgoLD+avkjJhVkjbZYw@mail.gmail.com>
	<BANLkTiniKfdR7e7AYLSs4Ghj0pq1V33emA@mail.gmail.com>
Message-ID: <BANLkTinrYCgRmZs=Faz4b89G6mQ7th=U0A@mail.gmail.com>

Can you explain your question?
Do you want each account to only allow X number of clients authed at a time
(other than one)?
Do you mean you want FS to disallow authing once there are e.g. 100 current
registered users on the server?
and.. why exactly would you want to do this? limiting concurrent calls or
calls per second would seem to be a more valuable performance metric.
-Avi

On Thu, Apr 7, 2011 at 7:11 PM, fieldpeak <fieldpeak at gmail.com> wrote:

> Could anyone help advise how to limit the max number of registration users
> on FS?
> Thanks if any advice.
>
> Regards,
> Charles
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Thu Apr  7 22:55:06 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Apr 2011 13:55:06 -0500
Subject: [Freeswitch-users] Job Opening at Barracuda Networks / CudaTel -
	Technical Escalation Engineer
Message-ID: <BANLkTi=wXcu43LLhg0prDkZo2SALKp3zqg@mail.gmail.com>

Looking for someone well-versed in debugging voice and data using
common tools like wireshark etc to take escalation incidents from tech
support at CudaTel.
Should be well versed in unix systems and internet tools and above
average diagnostic skills.

Preferred location or relocation to Ann Arbor MI but will entertain
applicants from the bay area in CA as well.

Please reply with resume to

jobs at freeswitch.org


-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From frank at telonium.com  Thu Apr  7 23:53:25 2011
From: frank at telonium.com (Frank Park)
Date: Thu, 7 Apr 2011 15:53:25 -0400
Subject: [Freeswitch-users] xml_curl response for voicemail_inject
In-Reply-To: <4D9BBF43.7050407@communicatefreely.net>
References: <BANLkTi=AwsfhJw7j6yLBBmy7hAUwVegkbA@mail.gmail.com>
	<BANLkTikvfBZEYeA=UsoKRhXOJdrnMVgJGQ@mail.gmail.com>
	<4D9BBF43.7050407@communicatefreely.net>
Message-ID: <BANLkTik9BHNczmSLz1cvFfBRO4Y2gnHQ1w@mail.gmail.com>

In my case, it see the directory request, and despite the response as
mentioned before, I am getting the same console output.
I am going to start looking at the source code to see if I can find what
it's doing... Can any of the developers replicate this error? Should I be
talking to the dev list for this?

Thank,
Frank

On Tue, Apr 5, 2011 at 9:17 PM, Tim St. Pierre <
fs-list at communicatefreely.net> wrote:

> I'm having the same problem.
>
> I'm returning a complete directory any time it's asked for, but I don't
> see FS requesting anything here.
>
> There is a request when it starts playing the message, but when I choose
> the forwarding option and enter an extension, I don't see any other
> directory requests.
>
> On the console, I get
>
> 2011-04-05 21:11:20.271219 [ERR] mod_voicemail.c:2767 Can't find profile
> 2011-04-05 21:11:20.271219 [ERR] mod_voicemail.c:1550 Failed to Carbon
> Copy to
> 5109
>
> Extension 5109 is the extension I was trying to forward to, and it's in
> the same domain as the extension I'm checking voice mail on.
>
> Why is it looking for the profile?  I would expect FS to do a directory
> lookup on the extension number that I entered, but that doesn't seem to
> be happening.  Any ideas?
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From all.eforums at gmail.com  Thu Apr  7 23:54:41 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Thu, 7 Apr 2011 15:54:41 -0400
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>
	<41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>
	<AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>
Message-ID: <BANLkTi=Mn0xEEojmmLsvyNpts02pR8akNA@mail.gmail.com>

Hello,

On Fri, Jan 28, 2011 at 7:10 PM, Steven Ayre <steveayre at gmail.com> wrote:

> I've been using it on Lenny with no problems for ~2 years, timing works
> fine. It will work. CentOS is the reference platform though.
>
> -Steve
>
>
Can anyone testify that as stated in the installation instructions on FS
Wiki for compiling it on Debian, the kernel still needs to be configured
with

"CONFIG_HZ_1000=y" and "CONFIG_HZ=1000"

If so, then how does one go about re-configuring to create a custom kernel?
I know I can google but if someone has a quick 2 liner on it that'll save me
some time ;)

I see this in my machine:

#> grep CONFIG_HZ config-2.6.32-5-sparc64  | grep 1000
# CONFIG_HZ_1000 is not set
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From paul at cupis.co.uk  Fri Apr  8 00:45:19 2011
From: paul at cupis.co.uk (Paul Cupis)
Date: Thu, 07 Apr 2011 21:45:19 +0100
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <BANLkTi=Mn0xEEojmmLsvyNpts02pR8akNA@mail.gmail.com>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>	<0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>	<41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>	<AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>
	<BANLkTi=Mn0xEEojmmLsvyNpts02pR8akNA@mail.gmail.com>
Message-ID: <4D9E225F.5070404@cupis.co.uk>

On 07/04/11 20:54, A E [Gmail] wrote:
> Can anyone testify that as stated in the installation instructions on FS
> Wiki for compiling it on Debian, the kernel still needs to be configured
> with
> 
> "CONFIG_HZ_1000=y" and "CONFIG_HZ=1000"
> 
> If so, then how does one go about re-configuring to create a custom kernel?
> I know I can google but if someone has a quick 2 liner on it that'll save me
> some time ;)

I'd like to know if this a good/recommomended idea as well. I'd be happy
to write up the procedure for changing the Debian kernel if this change
makes a big difference to FreeSWITCH.

Not sure how the timerfd stuff fits into this CONFIG_HZ issue, either.

Regards,



From anthony.minessale at gmail.com  Fri Apr  8 01:03:12 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 7 Apr 2011 16:03:12 -0500
Subject: [Freeswitch-users] Transfer attempt for a previously a replaced
 call fails
In-Reply-To: <4D9DE0CB.7000402@opensipstack.org>
References: <4D9A9A42.2070804@opensipstack.org>
	<BANLkTimsX4oJStWYDvz2yhPvf1e9F7kezw@mail.gmail.com>
	<4D9AADEB.7040803@opensipstack.org>
	<BANLkTi=XdyGKZcLmHmqe1_kupp5n7ib6KQ@mail.gmail.com>
	<4D9BAD82.3000205@opensipstack.org>
	<4D9DE0CB.7000402@opensipstack.org>
Message-ID: <BANLkTimZT-bBf45GX3YYdV_G6psmrzhtNA@mail.gmail.com>

The case where you are saying it doesnt work, the refer is to a call
on another box so its doing what we call a nightmare transfer where FS
INVITES the remote leg then cross connects them.  According to the log
as soon as they are bridge one leg of the call seems to be hungup or
disconnected. Is this when you press #?


On Thu, Apr 7, 2011 at 11:05 AM, Joegen E. Baclor
<joegen at opensipstack.org> wrote:
> Just bumping this thread.?? If I need to provide more info, just let me
> know.? Or if this is a known bug and a fix is due for a future version that
> is also acceptable.
>
> On 04/06/2011 08:02 AM, Joegen E. Baclor wrote:
>
> I'll keep that in mind.? If more information is needed to get into the
> bottom of this, I will happily oblige.? Thanks for helping.
>
> On 04/06/2011 03:09 AM, Michael Collins wrote:
>
> I'll have to defer to those more experienced than I in such matters.
> However, I can offer two tips:
> #1 - turn off the crazy sofia debugging - it's just noise. All you need to
> do to enable SIP trace is "sofia global siptrace on"
> #2 - when you pastebin the console output use the FreeSWITCH log syntax
> highlighting - it makes it *much* easier to see what's going on.
> -MC
>
> On Mon, Apr 4, 2011 at 10:51 PM, Joegen E. Baclor <joegen at opensipstack.org>
> wrote:
>>
>> Hi Michael,
>>
>> I have pasted both working and none working logs on pastebin.
>>
>> FreeSWITCH Version 1.0.7 (hacked-20110326T123355Z)
>> working:? http://pastebin.freeswitch.org/16008
>> not working:? http://pastebin.freeswitch.org/16009
>>
>> The call flow for the working call is
>> UA1 ->? (FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp)
>> FSIVRApp knows the uuid of the bridge call.? Pressing # on the IVR results
>> to a uuid_deflect on the bridged channel.? This works and call successfully
>> transfers to the new destination.
>>
>> The call flow for the none working call is
>>
>> 1.? UA1 -> UA2? is in conversation
>> 2.? UA1 puts UA2 on hold
>>
>> -- start of FS interaction here --
>>
>> 3.? UA1 ->? (FSBridgeDialPlan) -> (SIP-Loopback) -> (FSIVRApp)? (on line
>> 2)
>> 4.? UA1 sends REFER (replacing its call with UA2) to FSBridgeDialPlan.
>> 5.? Flow is now UA2 ->? ([REPLACED]FSBridgeDialPlan) -> (SIP-Loopback) ->
>> (FSIVRApp)
>> 6.? UA2 presses #.
>> 7.? IVRApp performs uuid_deflect on FSBridgeDialPlan.
>> 8. FSBridgeDialPlan drops call (no REFER is done)
>>
>> Thanks for your help.
>>
>> Joegen
>>
>> On 04/05/2011 12:35 PM, Michael Collins wrote:
>>
>> What do you see on the console when you try this? A console debug log with
>> siptrace would go a long way toward figuring out what is happening.
>> -MC
>>
>> On Mon, Apr 4, 2011 at 9:27 PM, Joegen E. Baclor <joegen at opensipstack.org>
>> wrote:
>>>
>>> Hi List,
>>>
>>> I have a scenario where a bridged call has been replaced due to a
>>> consultative transfer. ?This works pretty well and audio is
>>> bidirectional. ?I have the original uuid of the call in a var
>>> somewhere. ?The trouble begins when I uuid_deflect the bridged call once
>>> again to attempt another transfer. ?Sofia disconnects the channel. ?I am
>>> using the original uuid of the call (uuid prior to replaces). ?Is this
>>> the right way of doing it?
>>>
>>> Joegen
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From javieraristizabal at gmail.com  Fri Apr  8 01:18:37 2011
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Thu, 7 Apr 2011 16:18:37 -0500
Subject: [Freeswitch-users] defunct processes
Message-ID: <BANLkTi=D=5OS4H6KhJd1t-qgVUoJLR8hWw@mail.gmail.com>

Hi folks, I have FS running on a CentOS 5.3 (64 bits) and the last git
source. And i'm using ps -ef to look at the process running on my system and
i notice that i have more than 20 [freeswitch] <defunct> processes. Is this
normal? What exactly do that processes?

Thanks in advance


-- 
Javier Aristiz?bal
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From wstephen80 at gmail.com  Fri Apr  8 02:03:48 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Fri, 8 Apr 2011 00:03:48 +0200
Subject: [Freeswitch-users] Freeswitch
In-Reply-To: <BANLkTikCO-30PnPTgkam7+t4fo9VEAH=pQ@mail.gmail.com>
References: <BANLkTikCO-30PnPTgkam7+t4fo9VEAH=pQ@mail.gmail.com>
Message-ID: <BANLkTinEMBcC9__3X+CkUhN=yW9KJLGKrQ@mail.gmail.com>

Hi Kaushal,
I'm using Sangoma PRI cards and I'm very satisfied with them.
They are also very stable. I can recommend them.

Stephen

On Wed, Apr 6, 2011 at 8:28 AM, Kaushal Shriyan <kaushalshriyan at gmail.com>wrote:

> Hi,
>
> I have couple of questions regarding Asterisk.
>
> a) Does it has Automated Dialing Feature like dialing 1000 and 1000 of
> phone numbers?
> b) Does it Support VoiceXML ?
> c) What PRI Card is recommended for using Asterisk ?
>
> Thanks
>
> Kaushal
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bwibowo at gmail.com  Fri Apr  8 03:01:46 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Fri, 8 Apr 2011 06:01:46 +0700
Subject: [Freeswitch-users] webphone app
In-Reply-To: <93D7DF166EB94D98AEA5C03368283113@e1705>
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
	<32EF1A658EFC4E5393D8D1A3A486DA31@e1705>
	<BANLkTikdW-5-hDz9A+0mDid_KNEQNhzf6Q@mail.gmail.com>
	<93D7DF166EB94D98AEA5C03368283113@e1705>
Message-ID: <BANLkTik8m+5LyzG_OTBY+kmNXrONwKfBiw@mail.gmail.com>

have somebody in boophone? i already send inquiry but still no response

On Thu, Apr 7, 2011 at 10:35 AM, Madovsky <infos at madovsky.org> wrote:

>  this work with freeswitch
>
> ----- Original Message -----
> *From:* budi wibowo <bwibowo at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Wednesday, April 06, 2011 8:01 PM
> *Subject:* Re: [Freeswitch-users] webphone app
>
> thx, but i want to link the webphone to Freeswitch.
> not use any body's sip server
>
>
> thx
>
> budi
>
>
> On Thu, Apr 7, 2011 at 6:54 AM, Madovsky <infos at madovsky.org> wrote:
>
>>  boophone.com
>>
>>   ----- Original Message -----
>> *From:* budi wibowo <bwibowo at gmail.com>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Wednesday, April 06, 2011 7:42 PM
>> *Subject:* [Freeswitch-users] webphone app
>>
>> looking for webphone sip based on flash.
>> any info, please share
>>
>>
>> thx
>>
>> budi wibowo
>>
>> ------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From infos at madovsky.org  Fri Apr  8 03:18:26 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 7 Apr 2011 19:18:26 -0400
Subject: [Freeswitch-users] webphone app
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com><32EF1A658EFC4E5393D8D1A3A486DA31@e1705><BANLkTikdW-5-hDz9A+0mDid_KNEQNhzf6Q@mail.gmail.com><93D7DF166EB94D98AEA5C03368283113@e1705>
	<BANLkTik8m+5LyzG_OTBY+kmNXrONwKfBiw@mail.gmail.com>
Message-ID: <323421759C304B419A3EBB6E24F6B3A6@e1705>

I didn't receive your email.
pleas contact me off list thanks
  ----- Original Message ----- 
  From: budi wibowo 
  To: FreeSWITCH Users Help 
  Sent: Thursday, April 07, 2011 7:01 PM
  Subject: Re: [Freeswitch-users] webphone app


  have somebody in boophone? i already send inquiry but still no response  


  On Thu, Apr 7, 2011 at 10:35 AM, Madovsky <infos at madovsky.org> wrote:

    this work with freeswitch
      ----- Original Message ----- 
      From: budi wibowo 
      To: FreeSWITCH Users Help 
      Sent: Wednesday, April 06, 2011 8:01 PM
      Subject: Re: [Freeswitch-users] webphone app


      thx, but i want to link the webphone to Freeswitch. 
      not use any body's sip server




      thx


      budi 



      On Thu, Apr 7, 2011 at 6:54 AM, Madovsky <infos at madovsky.org> wrote:

        boophone.com
          ----- Original Message ----- 
          From: budi wibowo 
          To: FreeSWITCH Users Help 
          Sent: Wednesday, April 06, 2011 7:42 PM
          Subject: [Freeswitch-users] webphone app


          looking for webphone sip based on flash.  
          any info, please share




          thx


          budi wibowo


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


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



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






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


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



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






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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org
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From joegen at opensipstack.org  Fri Apr  8 04:17:17 2011
From: joegen at opensipstack.org (Joegen E. Baclor)
Date: Fri, 08 Apr 2011 08:17:17 +0800
Subject: [Freeswitch-users] Transfer attempt for a previously a replaced
 call fails
In-Reply-To: <BANLkTimZT-bBf45GX3YYdV_G6psmrzhtNA@mail.gmail.com>
References: <4D9A9A42.2070804@opensipstack.org>	<BANLkTimsX4oJStWYDvz2yhPvf1e9F7kezw@mail.gmail.com>	<4D9AADEB.7040803@opensipstack.org>	<BANLkTi=XdyGKZcLmHmqe1_kupp5n7ib6KQ@mail.gmail.com>	<4D9BAD82.3000205@opensipstack.org>	<4D9DE0CB.7000402@opensipstack.org>
	<BANLkTimZT-bBf45GX3YYdV_G6psmrzhtNA@mail.gmail.com>
Message-ID: <4D9E540D.5050607@opensipstack.org>

Anthony,

Thanks for looking into it.  Yes, it is right after I press pound.  If 
you need me to reproduce using a specific setting let me know.  I can 
pastebin the logs required.  Let me know if this requires a Jira tracker 
as well.

Joegen

On 04/08/2011 05:03 AM, Anthony Minessale wrote:
> The case where you are saying it doesnt work, the refer is to a call
> on another box so its doing what we call a nightmare transfer where FS
> INVITES the remote leg then cross connects them.  According to the log
> as soon as they are bridge one leg of the call seems to be hungup or
> disconnected. Is this when you press #?
>
>
> On Thu, Apr 7, 2011 at 11:05 AM, Joegen E. Baclor
> <joegen at opensipstack.org>  wrote:
>> Just bumping this thread.   If I need to provide more info, just let me
>> know.  Or if this is a known bug and a fix is due for a future version that
>> is also acceptable.
>>
>> On 04/06/2011 08:02 AM, Joegen E. Baclor wrote:
>>
>> I'll keep that in mind.  If more information is needed to get into the
>> bottom of this, I will happily oblige.  Thanks for helping.
>>
>> On 04/06/2011 03:09 AM, Michael Collins wrote:
>>
>> I'll have to defer to those more experienced than I in such matters.
>> However, I can offer two tips:
>> #1 - turn off the crazy sofia debugging - it's just noise. All you need to
>> do to enable SIP trace is "sofia global siptrace on"
>> #2 - when you pastebin the console output use the FreeSWITCH log syntax
>> highlighting - it makes it *much* easier to see what's going on.
>> -MC
>>
>> On Mon, Apr 4, 2011 at 10:51 PM, Joegen E. Baclor<joegen at opensipstack.org>
>> wrote:
>>> Hi Michael,
>>>
>>> I have pasted both working and none working logs on pastebin.
>>>
>>> FreeSWITCH Version 1.0.7 (hacked-20110326T123355Z)
>>> working:  http://pastebin.freeswitch.org/16008
>>> not working:  http://pastebin.freeswitch.org/16009
>>>
>>> The call flow for the working call is
>>> UA1 ->   (FSBridgeDialPlan) ->  (SIP-Loopback) ->  (FSIVRApp)
>>> FSIVRApp knows the uuid of the bridge call.  Pressing # on the IVR results
>>> to a uuid_deflect on the bridged channel.  This works and call successfully
>>> transfers to the new destination.
>>>
>>> The call flow for the none working call is
>>>
>>> 1.  UA1 ->  UA2  is in conversation
>>> 2.  UA1 puts UA2 on hold
>>>
>>> -- start of FS interaction here --
>>>
>>> 3.  UA1 ->   (FSBridgeDialPlan) ->  (SIP-Loopback) ->  (FSIVRApp)  (on line
>>> 2)
>>> 4.  UA1 sends REFER (replacing its call with UA2) to FSBridgeDialPlan.
>>> 5.  Flow is now UA2 ->   ([REPLACED]FSBridgeDialPlan) ->  (SIP-Loopback) ->
>>> (FSIVRApp)
>>> 6.  UA2 presses #.
>>> 7.  IVRApp performs uuid_deflect on FSBridgeDialPlan.
>>> 8. FSBridgeDialPlan drops call (no REFER is done)
>>>
>>> Thanks for your help.
>>>
>>> Joegen
>>>
>>> On 04/05/2011 12:35 PM, Michael Collins wrote:
>>>
>>> What do you see on the console when you try this? A console debug log with
>>> siptrace would go a long way toward figuring out what is happening.
>>> -MC
>>>
>>> On Mon, Apr 4, 2011 at 9:27 PM, Joegen E. Baclor<joegen at opensipstack.org>
>>> wrote:
>>>> Hi List,
>>>>
>>>> I have a scenario where a bridged call has been replaced due to a
>>>> consultative transfer.  This works pretty well and audio is
>>>> bidirectional.  I have the original uuid of the call in a var
>>>> somewhere.  The trouble begins when I uuid_deflect the bridged call once
>>>> again to attempt another transfer.  Sofia disconnects the channel.  I am
>>>> using the original uuid of the call (uuid prior to replaces).  Is this
>>>> the right way of doing it?
>>>>
>>>> Joegen
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org



From fieldpeak at gmail.com  Fri Apr  8 07:28:16 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Fri, 8 Apr 2011 11:28:16 +0800
Subject: [Freeswitch-users] FS -How to limit the max number of
 registration users
In-Reply-To: <BANLkTinrYCgRmZs=Faz4b89G6mQ7th=U0A@mail.gmail.com>
References: <BANLkTi=+4wCkiTCHgoLD+avkjJhVkjbZYw@mail.gmail.com>
	<BANLkTiniKfdR7e7AYLSs4Ghj0pq1V33emA@mail.gmail.com>
	<BANLkTinrYCgRmZs=Faz4b89G6mQ7th=U0A@mail.gmail.com>
Message-ID: <BANLkTimVaEzcw4gTgdyzaQi=X2zU1BCzYw@mail.gmail.com>

Hi Avi,

THanks for you reply.

I mean the second one ('want FS to disallow authing once there are e.g. 100
current registered users on the server' or send 403 forbidden or anything
else to not allow reigister more...), I'm afraid a lots of registeration
users will impact the performance especially without the extra ramdisk used,
thanks.

Regards,
Charles
2011/4/8 Avi Marcus <avi at avimarcus.net>

> Can you explain your question?
> Do you want each account to only allow X number of clients authed at a time
> (other than one)?
> Do you mean you want FS to disallow authing once there are e.g. 100 current
> registered users on the server?
>
and.. why exactly would you want to do this? limiting concurrent calls or
> calls per second would seem to be a more valuable performance metric.
> -Avi
>
> On Thu, Apr 7, 2011 at 7:11 PM, fieldpeak <fieldpeak at gmail.com> wrote:
>
>> Could anyone help advise how to limit the max number of registration users
>> on FS?
>> Thanks if any advice.
>>
>> Regards,
>> Charles
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From frankie.k.yiu at gmail.com  Fri Apr  8 07:37:39 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Thu, 7 Apr 2011 20:37:39 -0700
Subject: [Freeswitch-users] How to replay an audio when a ' * " (star key)
	is pressed?
Message-ID: <BANLkTi=FJOiRuOQJdsk7Kzib_L9aPcWSzg@mail.gmail.com>

Hi there,

I would like to play an audio to the channel and then if the callee presses
a ' * ' and the audio would replay immediately from the beginning, how can I
do that?  I am using C#.

Currently I am calling PlayAndGetDigits() and would play an audio.  I am
also subscribing a DTMF event, when I find a DTMF =  ' * '  I would call
this command "uuid_displace " + <unique ID> + " start " + <file name> + "
20" " to play the audio again.  The problem is that it seems like the event
come in after 6 or 7 sec after the callee presses the key.

What am I doing wrong or is there a better way to do this?

Thank you.

Frankie
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From frankie.k.yiu at gmail.com  Fri Apr  8 14:34:40 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Fri, 8 Apr 2011 03:34:40 -0700
Subject: [Freeswitch-users] how to send a "uuid_displace" command in C++
	code?
Message-ID: <BANLkTim5O3dSfYjd7xJWYNMkMVhdt1-1=g@mail.gmail.com>

Hi there,

I would like to know how I can send a "uuid_displace" command in my c++
code.

In C#, I can call Api.ExecuteString("uuid_displace " + <uuid string> + "
start " + <strFileName> + " 20");

but in C++ how can I send this command?

Hope someone can help me.

Thanks in advance!!

Frankie
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From peter.olsson at visionutveckling.se  Fri Apr  8 16:06:33 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Fri, 8 Apr 2011 14:06:33 +0200
Subject: [Freeswitch-users] how to send a "uuid_displace" command in
	C++	code?
In-Reply-To: <BANLkTim5O3dSfYjd7xJWYNMkMVhdt1-1=g@mail.gmail.com>
References: <BANLkTim5O3dSfYjd7xJWYNMkMVhdt1-1=g@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C4D30C8A@cooper>

If you're building a FS module, just execute core API switch_ivr_displace_session().

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Frankie Yiu
Skickat: den 8 april 2011 12:35
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] how to send a "uuid_displace" command in C++ code?

Hi there,

I would like to know how I can send a "uuid_displace" command in my c++ code.

In C#, I can call Api.ExecuteString("uuid_displace " + <uuid string> + " start " + <strFileName> + " 20");

but in C++ how can I send this command?

Hope someone can help me.

Thanks in advance!!

Frankie
!DSPAM:4d9ee61232761517916214!
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From sameer2k3t at gmail.com  Fri Apr  8 16:17:29 2011
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Fri, 8 Apr 2011 17:17:29 +0500
Subject: [Freeswitch-users] codec negotiation
In-Reply-To: <BANLkTini2w_5uc-LZRJxy4VwdcUKcthk7w@mail.gmail.com>
References: <BANLkTi=mD2mv726VZrmMO+ud2_tMgjf4EA@mail.gmail.com>
	<BANLkTini2w_5uc-LZRJxy4VwdcUKcthk7w@mail.gmail.com>
Message-ID: <BANLkTimuYKR2=2KETv6W-Lr7t75Dd5kOWg@mail.gmail.com>

Thanks for help

here it is

http://pastebin.freeswitch.org/16053



On Thu, Apr 7, 2011 at 5:55 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Can you show the debug-level log output including siptrace?
>
> -Steve
>
>
> On 7 April 2011 12:04, Sameer Khan <sameer2k3t at gmail.com> wrote:
>
>> hello every 1
>> i need help regarding codec negotiation
>> I set abs codec string in my dialplan $xml_output .='<action
>> application="export" data="nolocal:absolute_codec_string=PCMA,PCMU"/>';
>>
>> but still leg B is carrying the same codecs as leg A
>>
>> disable_transcoding is false in my internal sip profile
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pkelly at gmail.com  Fri Apr  8 18:20:46 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Fri, 8 Apr 2011 15:20:46 +0100
Subject: [Freeswitch-users] Setting SIP request URI in bridge
Message-ID: <BANLkTin6EKQkY10MGAuL1P0qaaKdDFKnpA@mail.gmail.com>

Hi

I need to bridge a call to an IP a.b.c.d, however the request URI needs to
be sent to a domain rather than an IP

e.g. the UDP packet is sent to a.b.c.d, but the INVITE looks like "INVITE
sip:12345 at my.domain.com"

This is the syntax I am using for the bridge:

<action application="bridge" data="{sip_invite_req_uri=
sip:12345 at my.domain.com,sip_from_uri=sip:$1 at my.domain.com
,codec_string='G729,PCMU,PCMA'}sofia/mediaserver/12345 at a.b.c.d"/>

however freeswitch just 503's the bridge with "NORMAL_TEMPORARY_FAILURE". If
I remove the sip_invite_req_uri part, the call is bridged fine, but the
INVITE line is incorrect.

Is it possible to do what I need in Freeswitch?

Thanks

Pete
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From pkelly at gmail.com  Fri Apr  8 19:11:41 2011
From: pkelly at gmail.com (Pete Kelly)
Date: Fri, 8 Apr 2011 16:11:41 +0100
Subject: [Freeswitch-users] Setting SIP request URI in bridge
In-Reply-To: <BANLkTin6EKQkY10MGAuL1P0qaaKdDFKnpA@mail.gmail.com>
References: <BANLkTin6EKQkY10MGAuL1P0qaaKdDFKnpA@mail.gmail.com>
Message-ID: <BANLkTikuEe--jXY5WEd3R7Eyg9Af-7ZDGA@mail.gmail.com>

bizarrely, I have removed the definition of a.b.c.d from the local hosts
file and moved it to DNS, and it now works.

Can anyone explain to me what's happening here?

On 8 April 2011 15:20, Pete Kelly <pkelly at gmail.com> wrote:

> Hi
>
> I need to bridge a call to an IP a.b.c.d, however the request URI needs to
> be sent to a domain rather than an IP
>
> e.g. the UDP packet is sent to a.b.c.d, but the INVITE looks like "INVITE
> sip:12345 at my.domain.com"
>
> This is the syntax I am using for the bridge:
>
> <action application="bridge" data="{sip_invite_req_uri=
> sip:12345 at my.domain.com,sip_from_uri=sip:$1 at my.domain.com
> ,codec_string='G729,PCMU,PCMA'}sofia/mediaserver/12345 at a.b.c.d"/>
>
> however freeswitch just 503's the bridge with "NORMAL_TEMPORARY_FAILURE".
> If I remove the sip_invite_req_uri part, the call is bridged fine, but the
> INVITE line is incorrect.
>
> Is it possible to do what I need in Freeswitch?
>
> Thanks
>
> Pete
>
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From mel0torme at gmail.com  Fri Apr  8 19:15:32 2011
From: mel0torme at gmail.com (Tom C)
Date: Fri, 8 Apr 2011 08:15:32 -0700
Subject: [Freeswitch-users] incoming call stop working after a few
	minutes
In-Reply-To: <4D98A62B.5060303@ppmt.org>
References: <4D98A62B.5060303@ppmt.org>
Message-ID: <BANLkTina=CGSCNYhGOU-4FUw9PkqLawovw@mail.gmail.com>

Did you figure this out already?

Changing the hostname can cause the router to assign a new IP address.  If
you have port forwarding set up for the old IP address, incoming SIP
requests would now be lost.  Making an outgoing call could convince the
router to forward those ports to the new IP temporarily.

On Sun, Apr 3, 2011 at 9:54 AM, Philippe Le Toquin <philippe at ppmt.org>wrote:

> Last Friday I decided to rename the hostname of the guruplug and since then
> I have problem with
> incoming calls no longer going through.
>
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From frank at telonium.com  Fri Apr  8 19:36:01 2011
From: frank at telonium.com (Frank Park)
Date: Fri, 8 Apr 2011 11:36:01 -0400
Subject: [Freeswitch-users] incoming call stop working after a few
	minutes
In-Reply-To: <BANLkTina=CGSCNYhGOU-4FUw9PkqLawovw@mail.gmail.com>
References: <4D98A62B.5060303@ppmt.org>
	<BANLkTina=CGSCNYhGOU-4FUw9PkqLawovw@mail.gmail.com>
Message-ID: <BANLkTinEPa-ui=p68ezCosm2k=qxcx=17w@mail.gmail.com>

This can be your router issue. Is the SIP client behind the NAT? Can you
check NAT keepalive settings? As Tom mentioned, looks like the router
reopens the route when it sees the outbound traffic originating from the
LAN. We've had call drops on longer calls before with Polycoms and it ended
up being NAT and router issue

Frank

On Fri, Apr 8, 2011 at 11:15 AM, Tom C <mel0torme at gmail.com> wrote:

> Did you figure this out already?
>
> Changing the hostname can cause the router to assign a new IP address.  If
> you have port forwarding set up for the old IP address, incoming SIP
> requests would now be lost.  Making an outgoing call could convince the
> router to forward those ports to the new IP temporarily.
>
> On Sun, Apr 3, 2011 at 9:54 AM, Philippe Le Toquin <philippe at ppmt.org>wrote:
>
>> Last Friday I decided to rename the hostname of the guruplug and since
>> then I have problem with
>> incoming calls no longer going through.
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

----=======================----
Frank Park
Telonium Communications, LLC
frank at telonium.com
http://www.telonium.com
Follow Us on Twitter: @GetTelonium
404-566-8888 x1001 Office
404-939-4242 Cell
----=======================----
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From anthony.minessale at gmail.com  Fri Apr  8 19:57:56 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Apr 2011 10:57:56 -0500
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <4D9E225F.5070404@cupis.co.uk>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>
	<41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>
	<AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>
	<BANLkTi=Mn0xEEojmmLsvyNpts02pR8akNA@mail.gmail.com>
	<4D9E225F.5070404@cupis.co.uk>
Message-ID: <BANLkTinou6OZAMtCRzVsurULmBmnLLZ99A@mail.gmail.com>

I think its relative to each kernel version.
The safe bet is to enable the 1000hz timer because 1ms is the least
amount of time FS needs to sleep.
Sometimes when you have a kernel that runs even faster the performance
goes down due to the extra cycles.
All I can say is test everything.

Try it both ways with 1000hz and however the default is and if you
support timerfd try that too.
param enable-softtimer-timerfd set to true in switch.conf.xml and/or
using mod_timer_fd and setting rtp_timer_name=timerfd in your sofia
profile.


On Thu, Apr 7, 2011 at 3:45 PM, Paul Cupis <paul at cupis.co.uk> wrote:
> On 07/04/11 20:54, A E [Gmail] wrote:
>> Can anyone testify that as stated in the installation instructions on FS
>> Wiki for compiling it on Debian, the kernel still needs to be configured
>> with
>>
>> "CONFIG_HZ_1000=y" and "CONFIG_HZ=1000"
>>
>> If so, then how does one go about re-configuring to create a custom kernel?
>> I know I can google but if someone has a quick 2 liner on it that'll save me
>> some time ;)
>
> I'd like to know if this a good/recommomended idea as well. I'd be happy
> to write up the procedure for changing the Debian kernel if this change
> makes a big difference to FreeSWITCH.
>
> Not sure how the timerfd stuff fits into this CONFIG_HZ issue, either.
>
> Regards,
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From fieldpeak at gmail.com  Fri Apr  8 20:21:14 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sat, 9 Apr 2011 00:21:14 +0800
Subject: [Freeswitch-users] Failed to startup as daemon on CentOS 5.5 with
	latest GIT head
Message-ID: <BANLkTinv4ob6BLYyFyCDj6bX4n+a2Wi6pA@mail.gmail.com>

I'm a newbie, and trying use scritp for FS to auto startup when OS start, i
follow below link,

http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix
http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora

However, after the OS started for some time, the FS still not startup (by
register from eyebeam failure and use './freeswitch status' to check), if i
manually excute the scritp (./freeswitch start), it works well.
Could anyone advise any clue to to resolve it?

Attached is the startup script, FS running on CentOS 5.5 with latest GIT
head.

Thanks

Regards,
Charles
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From all.eforums at gmail.com  Fri Apr  8 20:44:53 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Fri, 8 Apr 2011 12:44:53 -0400
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <BANLkTinou6OZAMtCRzVsurULmBmnLLZ99A@mail.gmail.com>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>
	<41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>
	<AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>
	<BANLkTi=Mn0xEEojmmLsvyNpts02pR8akNA@mail.gmail.com>
	<4D9E225F.5070404@cupis.co.uk>
	<BANLkTinou6OZAMtCRzVsurULmBmnLLZ99A@mail.gmail.com>
Message-ID: <BANLkTinFCtayvYeO6FG5hB1ONHOADw8vGQ@mail.gmail.com>

On Fri, Apr 8, 2011 at 11:57 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I think its relative to each kernel version.
> The safe bet is to enable the 1000hz timer because 1ms is the least
> amount of time FS needs to sleep.
> Sometimes when you have a kernel that runs even faster the performance
> goes down due to the extra cycles.
> All I can say is test everything.
>
> Try it both ways with 1000hz and however the default is and if you
> support timerfd try that too.
> param enable-softtimer-timerfd set to true in switch.conf.xml and/or
> using mod_timer_fd and setting rtp_timer_name=timerfd in your sofia
> profile.
>
>
Ok, Thanks Anthony. The default on my system was 250Hz.  Have changed that
and re-compiled the kernel. Will try out both and see what happens.

BTW, do we need timerfd in conjunction with the 1000hz timer set in the
kernel or is it either/or? As in does it affect positively or negatively if
we leave the kernel at 250Hz and enable timerfd as the timing source?

Thanks
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From philippe at ppmt.org  Fri Apr  8 19:33:40 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Fri, 08 Apr 2011 11:33:40 -0400
Subject: [Freeswitch-users] incoming call stop working after a few
	minutes
In-Reply-To: <BANLkTina=CGSCNYhGOU-4FUw9PkqLawovw@mail.gmail.com>
References: <4D98A62B.5060303@ppmt.org>
	<BANLkTina=CGSCNYhGOU-4FUw9PkqLawovw@mail.gmail.com>
Message-ID: <4D9F2AD4.4020200@ppmt.org>

Hello,

Thanks for your answer.

As one problem never comes alone I ended up with a damage SD card and
rather than waste too much time I reinstall
the all system from scratch (with the correct hostname!)

since then it is working fine.

I don't have port forwarding but you are right something must have
confused Freeswitch

regards

Philippe

On 11-04-08 11:15 AM, Tom C wrote:
> Did you figure this out already?
>  
> Changing the hostname can cause the router to assign a new IP
> address.  If you have port forwarding set up for the old IP address,
> incoming SIP requests would now be lost.  Making an outgoing call
> could convince the router to forward those ports to the new IP
> temporarily.
>
> On Sun, Apr 3, 2011 at 9:54 AM, Philippe Le Toquin <philippe at ppmt.org
> <mailto:philippe at ppmt.org>> wrote:
>
>     Last Friday I decided to rename the hostname of the guruplug and
>     since then I have problem with
>     incoming calls no longer going through.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From zacw at safisystems.com  Fri Apr  8 22:05:18 2011
From: zacw at safisystems.com (Zac Wolfe)
Date: Fri, 8 Apr 2011 11:05:18 -0700
Subject: [Freeswitch-users] New FreeSWITCH IVR coming, but need HELP!
In-Reply-To: <AANLkTikGiLtxOXJAbb-3KKzC4M_B+1XzMzimjDDcs_HW@mail.gmail.com>
References: <AANLkTi=L+EFi50Q4Mb-=X2HW=SsA_TG8tBcg9k6Dgmth@mail.gmail.com>
	<AANLkTikGiLtxOXJAbb-3KKzC4M_B+1XzMzimjDDcs_HW@mail.gmail.com>
Message-ID: <BANLkTi=anGwd-ZKQjmrA_Dbkt27mm0Cy5A@mail.gmail.com>

Thanks Anthony,

I dropped the ball on this bigtime.  I posted my question and somehow the
messages didnt thread properly in my list and didn't see any responses until
now.

I'll try your suggestions, it sounds like exactly what I was looking for.
Unfortunately we just released our product with FreeSWITCH (alpha) support
so it will have be included in a future update.

For this version, if the Saflet (event handler app) doesn't hang up the call
and the caller remains on the line, in some cases the call remains in park
indefinitely.  I see this as more of an annoyance than a major issue but
definitely something we want to address quickly.

In case you were interested the download site is
http://www.safisystems.com/downloads

We're still lacking somewhat in the documentation department but the
existing screencasts and walkthroughs we have apply to both FreeSWITCH and
Asterisk.

Thanks,
Zac

On Tue, Jan 11, 2011 at 2:45 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> What condition would you want to use to have the park terminate?
>
> Once you app is controlling the session, it would be up to you to
> enforce when it hangs up from the FS side.
>
> Based on what you describe the only issue could be when your remote
> application either misses the event or is restarted while calls are up
> so what I can suggest is this:
>
> in your C app, you could wait there for some timeout period just
> calling switch_ivr_sleep for 1 second up to 10 tries to wait a total
> of 10 seconds.
>
> If your app gets the event it can then transfer it to park using
> uuid_transfer, this would break the sleep loop and you could do
> something at the end of the loop like:
>
> if (switch_channel_ready(channel)) {
>   switch_channel_hangup(channel, SWITCH_CAUSE_DESTINATION_OUT_OF_ORDER);
> }
>
> so when it was already hung or being transfered that could would not
> execute but if your loop ended from waiting too long and it was still
> active it would hangup.
>
> you could do something similar with just dp logic if you used park_timeout
> of 10
> then in your event handler app, use uuid_setvar to unset park_timeout,
> then uuid_transfer it back to park now with now timeout
>
> uuid_transfer <uuid> set:park_timeout=,park inline
>
> You should come and present this at ClueCon if you have it done in time.
>
>
>
> On Mon, Jan 10, 2011 at 4:03 PM, Zac Wolfe <zacw at safisystems.com> wrote:
> > Hi guys,
> >
> > First some good news:  we're finally close to releasing our free IVR
> > Development platform SafiServer/SafiWorkshop (www.safisystems.com) with
> > FreeSWITCH support! It's happening much later than we originally
> anticipated
> > as we've been unexpectedly busy with contracting opportunities but I
> think
> > it will be worth the wait.  Currently everything is working fine with one
> > minor exception:  if the user-created script (we call them Saflets)
> doesn't
> > explicitly hang up the call, the call will remain parked until the caller
> > hangs up.  Some details:
> >
> > In Asterisk we invoke our server-side scripting applications via the
> > extensions.conf using the following syntax:
> >
> > exten =
> > 1111,1,Agi(agi://
> 192.168.0.10:3573/safletEngine.agi?saflet=project1/callflow1)
> >
> > Here '192.168.0.10' is the IP address of the SafiServer and
> > project1/callflow1 identifies the Saflet to be executed.  Asterisk
> creates a
> > socket connection to the SafiServer and, once the socket is disconnected,
> > the call proceeds to the next entry in the dialplan (typically 'hangup').
> >
> > For FreeSWITCH, the process is slightly different.  Currently, rather
> than
> > create a separate socket connection for each incoming call, we're
> invoking
> > an event that informs the SafiServer that there is a new incoming call.
> The
> > event contains the contextual information including the Saflet name.  For
> > example:
> >
> > <extension name="9194">
> >       <condition field="destination_number" expression="9194$">
> >         <action inline="true" application="set" data="saficall=true"/>
> >         <action application="event"
> >
> data="Event-Subclass=saficall::incoming,Event-Name=CUSTOM,saflet_project=test,saflet=flow1,new_saficall=true"/>
> >
> >        <!-- <action application="set" data="park_timeout=10"/> -->
> >         <action application="park" />
> >       </condition>
> >     </extension>
> >
> > So once the event is fired, the call is parked to prevent further
> execution
> > within the dialplan.  From there on, SafiServer is controlling the call
> via
> > Inbound Mod event socket.
> >
> > So this works perfectly, except that if the invoked Saflet doesn't
> > explicitly hang-up the call it will remain parked until the caller hangs
> > up.  My question is, is there a better way to do this?  Is there some
> better
> > alternative to park in this case?  Ideally I'd like to initiate a
> 'session'
> > of some kind when the SafiServer is "controlling" the call and then exit
> > that session as soon as the Saflet is complete, at which point the call
> > would continue on to the next entry in the dialplan.  I understand I
> could
> > use Outbound sockets to achieve this but, as I mentioned, I'd like to
> avoid
> > the overhead of a separate socket connection for each incoming call.
> >
> > I actually have a mod_saficall.c app that does basically the the same
> thing
> > as I described in the dialplan entry.  Perhaps there's something more I
> > could do in code that would allow me to be notified when the session is
> > complete.  Here's the relevant code I have so far:
> >
> >     switch_channel_t *channel = NULL;
> >     switch_event_t *event;
> >     const char *safiCallFlag = NULL;
> >     channel = switch_core_session_get_channel(session);
> >
> >     safiCallFlag = switch_channel_get_variable(channel, "saficall");
> >
> >     if (!safiCallFlag)
> >         switch_channel_set_variable(channel, "saficall", "true");
> >
> >
> >     if (switch_event_create_subclass(&event, SWITCH_EVENT_CUSTOM,
> > "saficall::incoming") == SWITCH_STATUS_SUCCESS) {
> >
> >         switch_event_add_header_string(event, SWITCH_STACK_BOTTOM,
> > "new_saficall", safiCallFlag ? "false" : "true");
> >
> >         switch_channel_event_set_data(channel, event);
> >         switch_event_fire(&event);
> >         switch_ivr_park(session, NULL);
> >     }
> >
> > Any ideas you might have on this are welcome.
> >
> > Thanks,
> > Zac Wolfe
> > Safi Systems LLC
> > www.safisystems.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
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> 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
>



-- 
Zac Wolfe
Safi Systems LLC
www.safisystems.com
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From curriegrad2004 at gmail.com  Fri Apr  8 22:39:40 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 8 Apr 2011 11:39:40 -0700
Subject: [Freeswitch-users] Failed to startup as daemon on CentOS 5.5
 with latest GIT head
In-Reply-To: <BANLkTinv4ob6BLYyFyCDj6bX4n+a2Wi6pA@mail.gmail.com>
References: <BANLkTinv4ob6BLYyFyCDj6bX4n+a2Wi6pA@mail.gmail.com>
Message-ID: <BANLkTimHjcQGmVo1fgfT8EY7D3D4SALt5g@mail.gmail.com>

did you run "chkconfig --add freeswitch && chkconfig --levels 35
freeswitch on" after you added the freeswitch init script to the
init.d directory?

On Fri, Apr 8, 2011 at 9:21 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> I'm a newbie, and trying use scritp for FS to auto startup when OS start, i
> follow below link,
>
> http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix
> http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora
>
> However, after the OS started for some time, the FS still not startup (by
> register from eyebeam failure and use './freeswitch status' to check), if i
> manually excute the scritp (./freeswitch start), it works well.
> Could anyone advise any clue to to resolve it?
>
> Attached is the startup script, FS running on CentOS 5.5 with latest GIT
> head.
>
> Thanks
>
> Regards,
> Charles
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jrichey at itltd.net  Fri Apr  8 22:41:01 2011
From: jrichey at itltd.net (JRichey)
Date: Fri, 8 Apr 2011 11:41:01 -0700 
Subject: [Freeswitch-users] Failed to startup as daemon on CentOS 5.5
	with latest GIT head
Message-ID: <6ECAF1527329364583AB525CF34ABF950B31A549@ms.kallback.com>

What do you see if you run "chkconfig --list | grep freeswitch"?

# chkconfig --list | grep freeswitch
freeswitch      0:off   1:off   2:on    3:on    4:on    5:on    6:off

If you don't get anything you'll need to add it with "chkconfig --add
freeswitch" and "chkconfig freeswitch on".
 

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of fieldpeak
Sent: Friday, April 08, 2011 9:21 AM
To: FreeSWITCH-users
Cc: 13910936628 at 139.com
Subject: [Freeswitch-users] Failed to startup as daemon on CentOS 5.5 with
latest GIT head


I'm a newbie, and trying use scritp for FS to auto startup when OS start, i
follow below link,

http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix
<http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix> 
http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora
<http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora> 

However, after the OS started for some time, the FS still not startup (by
register from eyebeam failure and use './freeswitch status' to check), if i
manually excute the scritp (./freeswitch start), it works well.
Could anyone advise any clue to to resolve it? 

Attached is the startup script, FS running on CentOS 5.5 with latest GIT
head.

Thanks 

Regards,
Charles




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From zacw at safisystems.com  Fri Apr  8 22:48:23 2011
From: zacw at safisystems.com (Zac Wolfe)
Date: Fri, 8 Apr 2011 11:48:23 -0700
Subject: [Freeswitch-users] New FreeSWITCH Graphical IVR Released...Testers
	Needed!
Message-ID: <BANLkTimPz+YQM4BJVoNe=bJwhCrXNHvh3Q@mail.gmail.com>

Hi all,

We (Safi Systems) have just released version Safi Communication Suite (SCS)
1.5.5.Beta, now with FreeSWITCH support!  You can check out the blog posting
here
<http://blog.safisystems.com/2011/04/08/safi-communications-suite-1-5-5-beta-released-now-with-freeswitch-support/>

The product is actually fairly mature and has been around now for over 2
years.  Up until now, however, Asterisk has been the only (non-commercial)
telephony platform supported.   The FreeSWITCH support should be considered
Alpha at this point and I'd like to appeal to you in the FreeSWITCH
community for feedback and suggestions to help up improve the product.

SCS is free to download and use in commercial and non-commercial capacities.

Downloads are available HERE <http://www.safisystems.com/downloads>

You may find our
screencasts<http://www.safisystems.com/screencasts/?pagemode=screencasts>and
wiki <http://wiki.safisystems.com/display/DOCS/Home> useful in getting
started.  Although some screencasts are still Asterisk-specific, most
concepts will apply to FreeSWITCH as well.  If you still have questions or
issues, our forums <http://www.safisystems.com/forums> are active and are an
excellent resource.

Thanks,
Zac Wolfe
Safi Systems LLC
www.safisystems.com
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From frankie.k.yiu at gmail.com  Fri Apr  8 22:51:21 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Fri, 8 Apr 2011 11:51:21 -0700
Subject: [Freeswitch-users] How to stop and replay an audio from the
	beginning?
Message-ID: <BANLkTim03e9hWPTTONcm1ARLCUFrrc+PfQ@mail.gmail.com>

Hi there,

I would like to know what is the preferred way to do the following.
My application is this:  Make a phone call to a person, after the person
picks up the phone an message would play.  If he presses the * key, the
message would stop and start from the beginning again.  I am using C# to
start the call and play an audio using  PlayAndGetDigits() while in my C++
code would check the DTMF event.  If it finds a * key pressed, it will call
the uuid_displace with the same file (but for testing purpose, I am using
different file).  This is not working right because I can hear that the new
audio file is playing on top of the original audio instead of stopping the
original audio and play the new audio.

Anyone has an idea how I should do this?

Thanks in advance.

Frankie
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From curriegrad2004 at gmail.com  Fri Apr  8 22:52:52 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 8 Apr 2011 11:52:52 -0700
Subject: [Freeswitch-users] Failed to startup as daemon on CentOS 5.5
 with latest GIT head
In-Reply-To: <6ECAF1527329364583AB525CF34ABF950B31A549@ms.kallback.com>
References: <6ECAF1527329364583AB525CF34ABF950B31A549@ms.kallback.com>
Message-ID: <BANLkTi=JOZUJmDhtfGdKDFDP++eFj7u3dw@mail.gmail.com>

I've updated the wiki to guide the newbies to get FreeSwitch starting
up automatically on boot for CentOS/Fedora systems.

On Fri, Apr 8, 2011 at 11:41 AM, JRichey <jrichey at itltd.net> wrote:
> What do you see if you run "chkconfig --list | grep freeswitch"?
>
> # chkconfig --list | grep freeswitch
> freeswitch????? 0:off?? 1:off?? 2:on??? 3:on??? 4:on??? 5:on??? 6:off
>
> If you don't get anything you'll need to add it with "chkconfig --add
> freeswitch" and "chkconfig freeswitch on".
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of fieldpeak
> Sent: Friday, April 08, 2011 9:21 AM
> To: FreeSWITCH-users
> Cc: 13910936628 at 139.com
> Subject: [Freeswitch-users] Failed to startup as daemon on CentOS 5.5 with
> latest GIT head
>
> I'm a newbie, and trying use scritp for FS to auto startup when OS start, i
> follow below link,
>
> http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix
> http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora
>
> However, after the OS started for some time, the FS still not startup (by
> register from eyebeam failure and use './freeswitch status' to check), if i
> manually excute the scritp (./freeswitch start), it works well.
> Could anyone advise any clue to to resolve it?
>
> Attached is the startup script, FS running on CentOS 5.5 with latest GIT
> head.
>
> Thanks
>
> Regards,
> Charles
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From krice at freeswitch.org  Fri Apr  8 22:55:02 2011
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 08 Apr 2011 13:55:02 -0500
Subject: [Freeswitch-users] New FreeSWITCH Graphical IVR
 Released...Testers Needed!
In-Reply-To: <BANLkTimPz+YQM4BJVoNe=bJwhCrXNHvh3Q@mail.gmail.com>
Message-ID: <C9C4C436.134EE%krice@freeswitch.org>

Is there a OSX Version?


On 4/8/11 1:48 PM, "Zac Wolfe" <zacw at safisystems.com> wrote:

> Hi all,
> 
> We (Safi Systems) have just released version Safi Communication Suite (SCS)
> 1.5.5.Beta, now with FreeSWITCH support!? You can check out the blog posting
> here  
> <http://blog.safisystems.com/2011/04/08/safi-communications-suite-1-5-5-beta-r
> eleased-now-with-freeswitch-support/>
> 
> The product is actually fairly mature and has been around now for over 2
> years.? Up until now, however, Asterisk has been the only (non-commercial)
> telephony platform supported. ? The FreeSWITCH support should be considered
> Alpha at this point and I'd like to appeal to you in the FreeSWITCH community
> for feedback and suggestions to help up improve the product.?
> 
> SCS is free to download and use in commercial and non-commercial capacities.
> 
> Downloads are available HERE <http://www.safisystems.com/downloads>
> 
> You may find our screencasts
> <http://www.safisystems.com/screencasts/?pagemode=screencasts>  and wiki
> <http://wiki.safisystems.com/display/DOCS/Home>  useful in getting started.?
> Although some screencasts are still Asterisk-specific, most concepts will
> apply to FreeSWITCH as well.? If you still have questions or issues, our
> forums <http://www.safisystems.com/forums>  are active and are an excellent
> resource.
> 
> Thanks,
> Zac Wolfe
> Safi Systems LLC
> www.safisystems.com <http://www.safisystems.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

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From fvillarroel at yahoo.com  Fri Apr  8 22:57:26 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Fri, 8 Apr 2011 11:57:26 -0700 (PDT)
Subject: [Freeswitch-users] Account ACL
Message-ID: <172612.86506.qm@web34308.mail.mud.yahoo.com>

Hi Community.

How i can identifi inbound traffic authorizated on ACL with some variable like Accountcode.

For aoutbound traffic i use:

 <action application="set" data="accountcode=foo"/>

My problem is for inbound traffic how i can identify accounts?

Regards.


From mrene_lists at avgs.ca  Fri Apr  8 23:00:25 2011
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 8 Apr 2011 15:00:25 -0400
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <172612.86506.qm@web34308.mail.mud.yahoo.com>
References: <172612.86506.qm@web34308.mail.mud.yahoo.com>
Message-ID: <578DDCBD-C870-42A1-8B23-2502193D98E6@avgs.ca>

http://wiki.freeswitch.org/wiki/ACL#Users

You can set variables directly in the user's directory entry.

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




On 2011-04-08, at 2:57 PM, FERNANDO VILLARROEL wrote:

> Hi Community.
> 
> How i can identifi inbound traffic authorizated on ACL with some variable like Accountcode.
> 
> For aoutbound traffic i use:
> 
> <action application="set" data="accountcode=foo"/>
> 
> My problem is for inbound traffic how i can identify accounts?
> 
> Regards.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From zacw at safisystems.com  Fri Apr  8 23:01:01 2011
From: zacw at safisystems.com (Zac Wolfe)
Date: Fri, 8 Apr 2011 12:01:01 -0700
Subject: [Freeswitch-users] New FreeSWITCH Graphical IVR
 Released...Testers Needed!
In-Reply-To: <C9C4C436.134EE%krice@freeswitch.org>
References: <BANLkTimPz+YQM4BJVoNe=bJwhCrXNHvh3Q@mail.gmail.com>
	<C9C4C436.134EE%krice@freeswitch.org>
Message-ID: <BANLkTi=+qtVTqEmUQRU7K3iw=Br=zj1bfA@mail.gmail.com>

Sorry no OSX support at the moment.  Currently the graphical designer
(SafiWorkshop) is supported in Windows and Linux (Alpha) and the server
portion (SafiServer) is supported in Linux and Windows.

The product is written entirely in Java and there's no reason why an
OSX-port couldn't be created without too much effort.  If there's enough
interest we'd certainly consider it.

On Fri, Apr 8, 2011 at 11:55 AM, Ken Rice <krice at freeswitch.org> wrote:

>  Is there a OSX Version?
>
>
>
> On 4/8/11 1:48 PM, "Zac Wolfe" <zacw at safisystems.com> wrote:
>
> Hi all,
>
> We (Safi Systems) have just released version Safi Communication Suite (SCS)
> 1.5.5.Beta, now with FreeSWITCH support!  You can check out the blog posting
> here  <
> http://blog.safisystems.com/2011/04/08/safi-communications-suite-1-5-5-beta-released-now-with-freeswitch-support/>
>
>
> The product is actually fairly mature and has been around now for over 2
> years.  Up until now, however, Asterisk has been the only (non-commercial)
> telephony platform supported.   The FreeSWITCH support should be considered
> Alpha at this point and I'd like to appeal to you in the FreeSWITCH
> community for feedback and suggestions to help up improve the product.
>
> SCS is free to download and use in commercial and non-commercial
> capacities.
>
> Downloads are available HERE <http://www.safisystems.com/downloads>
>
> You may find our screencasts <
> http://www.safisystems.com/screencasts/?pagemode=screencasts>  and wiki <
> http://wiki.safisystems.com/display/DOCS/Home>  useful in getting
> started.  Although some screencasts are still Asterisk-specific, most
> concepts will apply to FreeSWITCH as well.  If you still have questions or
> issues, our forums <http://www.safisystems.com/forums>  are active and are
> an excellent resource.
>
>
> Thanks,
> Zac Wolfe
> Safi Systems LLC
> www.safisystems.com <http://www.safisystems.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
>
>


-- 
Zac Wolfe
Safi Systems LLC
www.safisystems.com
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From krice at freeswitch.org  Fri Apr  8 23:23:58 2011
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 08 Apr 2011 14:23:58 -0500
Subject: [Freeswitch-users] New FreeSWITCH Graphical IVR
 Released...Testers Needed!
In-Reply-To: <BANLkTi=+qtVTqEmUQRU7K3iw=Br=zj1bfA@mail.gmail.com>
Message-ID: <C9C4CAFE.134F3%krice@freeswitch.org>

I don?t think the server portion is a big deal, just that many of the FS
Devs and Professional support people out there use OSX instead of windows


On 4/8/11 2:01 PM, "Zac Wolfe" <zacw at safisystems.com> wrote:

> Sorry no OSX support at the moment.? Currently the graphical designer
> (SafiWorkshop) is supported in Windows and Linux (Alpha) and the server
> portion (SafiServer) is supported in Linux and Windows.
> 
> The product is written entirely in Java and there's no reason why an OSX-port
> couldn't be created without too much effort.? If there's enough interest we'd
> certainly consider it.
> 
> On Fri, Apr 8, 2011 at 11:55 AM, Ken Rice <krice at freeswitch.org> wrote:
>> Is there a OSX Version?
>> 
>> 
>> 
>> On 4/8/11 1:48 PM, "Zac Wolfe" <zacw at safisystems.com
>> <http://zacw at safisystems.com> > wrote:
>> 
>>> Hi all,
>>> 
>>> We (Safi Systems) have just released version Safi Communication Suite (SCS)
>>> 1.5.5.Beta, now with FreeSWITCH support!? You can check out the blog posting
>>> here 
>>> ?<http://blog.safisystems.com/2011/04/08/safi-communications-suite-1-5-5-bet
>>> a-released-now-with-freeswitch-support/>
>>> 
>>> The product is actually fairly mature and has been around now for over 2
>>> years.? Up until now, however, Asterisk has been the only (non-commercial)
>>> telephony platform supported. ? The FreeSWITCH support should be considered
>>> Alpha at this point and I'd like to appeal to you in the FreeSWITCH
>>> community for feedback and suggestions to help up improve the product.?
>>> 
>>> SCS is free to download and use in commercial and non-commercial capacities.
>>> 
>>> Downloads are available HERE <http://www.safisystems.com/downloads>
>>> 
>>> You may find our screencasts
>>> <http://www.safisystems.com/screencasts/?pagemode=screencasts> ?and wiki
>>> <http://wiki.safisystems.com/display/DOCS/Home> ?useful in getting started.?
>>> Although some screencasts are still Asterisk-specific, most concepts will
>>> apply to FreeSWITCH as well.? If you still have questions or issues, our
>>> forums <http://www.safisystems.com/forums> ?are active and are an excellent
>>> resource.
>>> 
>>> 
>>> Thanks,
>>> Zac Wolfe
>>> Safi Systems LLC
>>> www.safisystems.com <http://www.safisystems.com>
>>> <http://www.safisystems.com>
>>> 
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> <http://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 

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From moises.silva at gmail.com  Fri Apr  8 23:28:52 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 8 Apr 2011 15:28:52 -0400
Subject: [Freeswitch-users] Cannot compile freetdm!
In-Reply-To: <BANLkTik3PXUY_=4hg1DQ23iGtGipxpG50Q@mail.gmail.com>
References: <BANLkTik3PXUY_=4hg1DQ23iGtGipxpG50Q@mail.gmail.com>
Message-ID: <BANLkTik8YvpSNOD3oDdatdm8WxgLi0SGMw@mail.gmail.com>

Valery,

Where did you get libisdn from? The developer of libisdn (stkn on irc) is
likely working on his own branch of freetdm as I have not seen updates from
him lately. May be you can ask him for an update.

Moises Silva
Senior Software Engineer, Software Development Manager
Sangoma Technologies Inc. | 100 Renfrew Drive, Suite 100, Markham ON L3R 9R6
Canada
t. 1 905 474 1990 x128 | e. moy at sangoma.com


On Wed, Apr 6, 2011 at 9:43 PM, Valery Kalinin <valery.kalinin at gmail.com>wrote:

> # cd /usr/local/freeswitch/libs/freetdm
> # ./configure --with-libisdn
> # make
>
> bla-bla-bla
>
> cc1: warnings being treated as errors
> src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'ftdm_isdn_931_34':
> src/ftmod/ftmod_isdn/ftmod_isdn.c:982: warning: unused variable 'cplen'
> src/ftmod/ftmod_isdn/ftmod_isdn.c: In function 'isdn_configure_span':
> src/ftmod/ftmod_isdn/ftmod_isdn.c:2794: warning: passing argument 2 of
> 'Q931SetLogCB' from incompatible pointer type
> make: *** [ftmod_isdn_la-ftmod_isdn.lo] Error 1
>
> Why?
> libisdn-0.0.1 installed
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From edpimentl at gmail.com  Fri Apr  8 23:39:31 2011
From: edpimentl at gmail.com (EdPimentl)
Date: Fri, 8 Apr 2011 15:39:31 -0400
Subject: [Freeswitch-users] New FreeSWITCH Graphical IVR
 Released...Testers Needed!
In-Reply-To: <C9C4CAFE.134F3%krice@freeswitch.org>
References: <BANLkTi=+qtVTqEmUQRU7K3iw=Br=zj1bfA@mail.gmail.com>
	<C9C4CAFE.134F3%krice@freeswitch.org>
Message-ID: <BANLkTimv-R95Q4vPjf3on2enr2ff3berQw@mail.gmail.com>

Jot us down to test OSX.

-E
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From msc at freeswitch.org  Fri Apr  8 23:58:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 8 Apr 2011 12:58:44 -0700
Subject: [Freeswitch-users] How to stop and replay an audio from the
	beginning?
In-Reply-To: <BANLkTim03e9hWPTTONcm1ARLCUFrrc+PfQ@mail.gmail.com>
References: <BANLkTim03e9hWPTTONcm1ARLCUFrrc+PfQ@mail.gmail.com>
Message-ID: <BANLkTin658eqN-TFuiqXf6mcuy+J5fDmgg@mail.gmail.com>

I don't know if what you're doing is the optimal way or not, but I'm pretty
sure that if you are playing a second file then you need to break out of
playing the first file. Look at the uuid_break API for ideas on how to do
that.

-MC

On Fri, Apr 8, 2011 at 11:51 AM, Frankie Yiu <frankie.k.yiu at gmail.com>wrote:

> Hi there,
>
> I would like to know what is the preferred way to do the following.
> My application is this:  Make a phone call to a person, after the person
> picks up the phone an message would play.  If he presses the * key, the
> message would stop and start from the beginning again.  I am using C# to
> start the call and play an audio using  PlayAndGetDigits() while in my C++
> code would check the DTMF event.  If it finds a * key pressed, it will call
> the uuid_displace with the same file (but for testing purpose, I am using
> different file).  This is not working right because I can hear that the new
> audio file is playing on top of the original audio instead of stopping the
> original audio and play the new audio.
>
> Anyone has an idea how I should do this?
>
> Thanks in advance.
>
> Frankie
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From javieraristizabal at gmail.com  Sat Apr  9 00:00:19 2011
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Fri, 8 Apr 2011 15:00:19 -0500
Subject: [Freeswitch-users] defunct processes
In-Reply-To: <BANLkTi=D=5OS4H6KhJd1t-qgVUoJLR8hWw@mail.gmail.com>
References: <BANLkTi=D=5OS4H6KhJd1t-qgVUoJLR8hWw@mail.gmail.com>
Message-ID: <BANLkTikTS_DLMcGs8uyogKKXrfU0Tdggzw@mail.gmail.com>

Any clue?

Thanks



2011/4/7 Javier Aristiz?bal <javieraristizabal at gmail.com>

> Hi folks, I have FS running on a CentOS 5.3 (64 bits) and the last git
> source. And i'm using ps -ef to look at the process running on my system and
> i notice that i have more than 20 [freeswitch] <defunct> processes. Is
> this normal? What exactly do that processes?
>
> Thanks in advance
>
>
> --
> Javier Aristiz?bal
>
>


-- 
Javier Aristiz?bal
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From marcdecorny at gmail.com  Sat Apr  9 00:08:07 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Fri, 8 Apr 2011 21:08:07 +0100
Subject: [Freeswitch-users] ESL with PHP not working
Message-ID: <BANLkTim4NmFsHarccjRW_edmD-oV0OVZtw@mail.gmail.com>

Hi all

got an issue with ESL I cannot figure out.

I have installed enabled the event socket on the Freeswitch, and it works
locally on the server via PHP

I have a remote server were I compiled the ESL.so and did the php-install.
It is a standard CentOS install with apache.
when I type into the command line :     php test.php   ( the standard test
script that is api status ) I get the correct result.
when I execute the same script from the browser on the remote server I get
an error on the getBody command.
*Fatal error*: Call to a member function getBody() on a non-object in *
/var/www/html/test.php* on line *9*

the only think I can think of as it works from the command line as root but
not as apache is rights or ownerships. but I have changed everything to
apache without any luck.

I'm thinking this must be a common issue. Anybody experienced this ?

Thanks
Marc
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From fvillarroel at yahoo.com  Sat Apr  9 01:05:36 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Fri, 8 Apr 2011 14:05:36 -0700 (PDT)
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <578DDCBD-C870-42A1-8B23-2502193D98E6@avgs.ca>
Message-ID: <453850.62406.qm@web34303.mail.mud.yahoo.com>

Dear Mathieu.

My user are not in ditrectory.

My users are gateways authenticated with ACL.

How i can use some variable for identifi like accountcode for inbound traffic from this gateways???

Regards

--- On Fri, 4/8/11, Mathieu Rene <mrene_lists at avgs.ca> wrote:

From: Mathieu Rene <mrene_lists at avgs.ca>
Subject: Re: [Freeswitch-users] Account ACL
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, April 8, 2011, 4:00 PM

http://wiki.freeswitch.org/wiki/ACL#Users
You can set variables directly in the user's directory entry.

Mathieu ReneAvant-Garde Solutions IncOffice: + 1 (514) 664-1044 x100Cell: +1 (514) 664-1044 x200mrene at avgs.ca





On 2011-04-08, at 2:57 PM, FERNANDO VILLARROEL wrote:
Hi Community.

How i can identifi inbound traffic authorizated on ACL with some variable like Accountcode.

For aoutbound traffic i use:

 <action application="set" data="accountcode=foo"/>

My problem is for inbound traffic how i can identify accounts?

Regards.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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
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From zacw at safisystems.com  Sat Apr  9 01:42:09 2011
From: zacw at safisystems.com (Zac Wolfe)
Date: Fri, 8 Apr 2011 14:42:09 -0700
Subject: [Freeswitch-users] New FreeSWITCH Graphical IVR
 Released...Testers Needed!
In-Reply-To: <BANLkTimv-R95Q4vPjf3on2enr2ff3berQw@mail.gmail.com>
References: <BANLkTi=+qtVTqEmUQRU7K3iw=Br=zj1bfA@mail.gmail.com>
	<C9C4CAFE.134F3%krice@freeswitch.org>
	<BANLkTimv-R95Q4vPjf3on2enr2ff3berQw@mail.gmail.com>
Message-ID: <BANLkTik1w3fNNHwLqXjexiB6CcPJ3X9nfw@mail.gmail.com>

Sounds like we'll need to get a Mac in our labs :)

On Fri, Apr 8, 2011 at 12:39 PM, EdPimentl <edpimentl at gmail.com> wrote:

> Jot us down to test OSX.
>
> -E
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Zac Wolfe
Safi Systems LLC
www.safisystems.com
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From pablosaro at gmail.com  Sat Apr  9 02:07:54 2011
From: pablosaro at gmail.com (Pablo Hernan Saro)
Date: Fri, 8 Apr 2011 19:07:54 -0300
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <453850.62406.qm@web34303.mail.mud.yahoo.com>
References: <578DDCBD-C870-42A1-8B23-2502193D98E6@avgs.ca>
	<453850.62406.qm@web34303.mail.mud.yahoo.com>
Message-ID: <BANLkTimzpJP2KjDsir8H=6q6nb7oJfOCSg@mail.gmail.com>

Probably it's not the best for you, but the first solution that comes to my
mind is setting the accountcode at dialplan. If you are working in a high
performance scenario with lot of gateways, then probably you have an
objection. Anyway, it would be something like this:

<extensio name="Setting account code for gateways">
  <condition field="network_addr" expression="^192\.168\.1\.1$"
break="on-true">
    <action application="set" data="accountcode=gateway1"/>
  </condition>
  <condition field="network_addr" expression="^192\.168\.1\.2$"
break="on-true">
    <action application="set" data="accountcode=gateway2"/>
  </condition>
</extension>

On Fri, Apr 8, 2011 at 6:05 PM, FERNANDO VILLARROEL
<fvillarroel at yahoo.com>wrote:

> Dear Mathieu.
>
> My user are not in ditrectory.
>
> My users are gateways authenticated with ACL.
>
> How i can use some variable for identifi like accountcode for inbound
> traffic from this gateways???
>
> Regards
>
> --- On *Fri, 4/8/11, Mathieu Rene <mrene_lists at avgs.ca>* wrote:
>
>
> From: Mathieu Rene <mrene_lists at avgs.ca>
> Subject: Re: [Freeswitch-users] Account ACL
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Friday, April 8, 2011, 4:00 PM
>
>
> http://wiki.freeswitch.org/wiki/ACL#Users
>
> You can set variables directly in the user's directory entry.
> <http://wiki.freeswitch.org/wiki/ACL#Users>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca <http://mc/compose?to=mrene at avgs.ca>
>
>
>
>
> On 2011-04-08, at 2:57 PM, FERNANDO VILLARROEL wrote:
>
> Hi Community.
>
> How i can identifi inbound traffic authorizated on ACL with some variable
> like Accountcode.
>
> For aoutbound traffic i use:
>
> <action application="set" data="accountcode=foo"/>
>
> My problem is for inbound traffic how i can identify accounts?
>
> Regards.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at 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://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Sat Apr  9 02:16:24 2011
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 08 Apr 2011 17:16:24 -0500
Subject: [Freeswitch-users] New FreeSWITCH Graphical IVR
 Released...Testers Needed!
In-Reply-To: <BANLkTik1w3fNNHwLqXjexiB6CcPJ3X9nfw@mail.gmail.com>
Message-ID: <C9C4F368.13504%krice@freeswitch.org>

Is this an eclipse plugin?


On 4/8/11 4:42 PM, "Zac Wolfe" <zacw at safisystems.com> wrote:

> Sounds like we'll need to get a Mac in our labs :)
> 
> On Fri, Apr 8, 2011 at 12:39 PM, EdPimentl <edpimentl at gmail.com> wrote:
>> Jot us down to test OSX.
>> 
>> -E
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 

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From anthony.minessale at gmail.com  Sat Apr  9 03:08:47 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Apr 2011 18:08:47 -0500
Subject: [Freeswitch-users] How to stop and replay an audio from the
	beginning?
In-Reply-To: <BANLkTim03e9hWPTTONcm1ARLCUFrrc+PfQ@mail.gmail.com>
References: <BANLkTim03e9hWPTTONcm1ARLCUFrrc+PfQ@mail.gmail.com>
Message-ID: <BANLkTinKnkuE=_dL4CqM6N=PQtAW8UBn5g@mail.gmail.com>

you would at least have to disable the displace you already called
since you can have concurrent displace bugs


On Fri, Apr 8, 2011 at 1:51 PM, Frankie Yiu <frankie.k.yiu at gmail.com> wrote:
> Hi there,
>
> I would like to know what is the preferred way to do the following.
> My application is this:? Make a phone call to a person, after the person
> picks up the phone an message would play.??If he presses the * key, the
> message would stop and start from the beginning again.??I am using C# to
> start the call and play an audio using ?PlayAndGetDigits() while in my C++
> code would check the DTMF event.? If it finds a * key pressed, it will call
> the uuid_displace with the same file (but for testing purpose, I am using
> different file).? This is not working right because I can hear that the new
> audio file is playing on top of the original audio instead of stopping the
> original audio and play the new audio.
>
> Anyone has an idea how I should do this?
> Thanks in advance.
>
> Frankie
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From zacw at safisystems.com  Sat Apr  9 03:18:32 2011
From: zacw at safisystems.com (Zac Wolfe)
Date: Fri, 8 Apr 2011 16:18:32 -0700
Subject: [Freeswitch-users] New FreeSWITCH Graphical IVR
 Released...Testers Needed!
In-Reply-To: <C9C4F368.13504%krice@freeswitch.org>
References: <BANLkTik1w3fNNHwLqXjexiB6CcPJ3X9nfw@mail.gmail.com>
	<C9C4F368.13504%krice@freeswitch.org>
Message-ID: <BANLkTimb67L8kyy51OWv3My7nPKaqndL2g@mail.gmail.com>

It was built using Eclipse RCP but it's not a plug-in.

On Fri, Apr 8, 2011 at 3:16 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Is this an eclipse plugin?
>
>
> On 4/8/11 4:42 PM, "Zac Wolfe" <zacw at safisystems.com> wrote:
>
> Sounds like we'll need to get a Mac in our labs :)
>
> On Fri, Apr 8, 2011 at 12:39 PM, EdPimentl <edpimentl at gmail.com> wrote:
>
> Jot us down to test OSX.
>
> -E
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Zac Wolfe
Safi Systems LLC
www.safisystems.com
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From anthony.minessale at gmail.com  Sat Apr  9 04:06:05 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 8 Apr 2011 19:06:05 -0500
Subject: [Freeswitch-users] defunct processes
In-Reply-To: <BANLkTikTS_DLMcGs8uyogKKXrfU0Tdggzw@mail.gmail.com>
References: <BANLkTi=D=5OS4H6KhJd1t-qgVUoJLR8hWw@mail.gmail.com>
	<BANLkTikTS_DLMcGs8uyogKKXrfU0Tdggzw@mail.gmail.com>
Message-ID: <BANLkTimKpDjGkW3_mCqw4NUD0p9V0C7W5A@mail.gmail.com>

Are you using the exec directive? There was a recent fix related to that
that could fit your question.
On Apr 8, 2011 3:01 PM, "Javier Aristiz?bal" <javieraristizabal at gmail.com>
wrote:
> Any clue?
>
> Thanks
>
>
>
> 2011/4/7 Javier Aristiz?bal <javieraristizabal at gmail.com>
>
>> Hi folks, I have FS running on a CentOS 5.3 (64 bits) and the last git
>> source. And i'm using ps -ef to look at the process running on my system
and
>> i notice that i have more than 20 [freeswitch] <defunct> processes. Is
>> this normal? What exactly do that processes?
>>
>> Thanks in advance
>>
>>
>> --
>> Javier Aristiz?bal
>>
>>
>
>
> --
> Javier Aristiz?bal
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From bwibowo at gmail.com  Sat Apr  9 05:13:39 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Sat, 9 Apr 2011 08:13:39 +0700
Subject: [Freeswitch-users] webphone app
In-Reply-To: <1302165236411-6249102.post@n2.nabble.com>
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
	<1302165236411-6249102.post@n2.nabble.com>
Message-ID: <BANLkTi=6SL4irBVFshO9NQqJQc9Nhyo9gw@mail.gmail.com>

check flashphoner,.. why flash sip phone seems need specific app server? how
about connecting sip flash phone to generic sip server

On Thu, Apr 7, 2011 at 3:33 PM, peely <freeswitch at peely.com> wrote:

> http://www.flashphoner.com/
>
> It's commercial, but well worth it and very stable, unlike red5. It's built
> on the wowza media server.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/webphone-app-tp6248061p6249102.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
>
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From kris at livecall.com  Sat Apr  9 05:27:25 2011
From: kris at livecall.com (Kris)
Date: Fri, 8 Apr 2011 18:27:25 -0700
Subject: [Freeswitch-users] How to stop and replay an audio from
	thebeginning?
References: <BANLkTim03e9hWPTTONcm1ARLCUFrrc+PfQ@mail.gmail.com>
	<BANLkTin658eqN-TFuiqXf6mcuy+J5fDmgg@mail.gmail.com>
Message-ID: <76367CB43C5241C1A91B5E2029918D32@stor1>

After the PlayAndGetDigits()  terminates on the *, loop around and play 
again with PlayAndGetDigits() . don't use displace.

----- Original Message ----- 
From: "Michael Collins" <msc at freeswitch.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, April 08, 2011 12:58 PM
Subject: Re: [Freeswitch-users] How to stop and replay an audio from 
thebeginning?


>I don't know if what you're doing is the optimal way or not, but I'm pretty
> sure that if you are playing a second file then you need to break out of
> playing the first file. Look at the uuid_break API for ideas on how to do
> that.
>
> -MC
>
> On Fri, Apr 8, 2011 at 11:51 AM, Frankie Yiu 
> <frankie.k.yiu at gmail.com>wrote:
>
>> Hi there,
>>
>> I would like to know what is the preferred way to do the following.
>> My application is this:  Make a phone call to a person, after the person
>> picks up the phone an message would play.  If he presses the * key, the
>> message would stop and start from the beginning again.  I am using C# to
>> start the call and play an audio using  PlayAndGetDigits() while in my 
>> C++
>> code would check the DTMF event.  If it finds a * key pressed, it will 
>> call
>> the uuid_displace with the same file (but for testing purpose, I am using
>> different file).  This is not working right because I can hear that the 
>> new
>> audio file is playing on top of the original audio instead of stopping 
>> the
>> original audio and play the new audio.
>>
>> Anyone has an idea how I should do this?
>>
>> Thanks in advance.
>>
>> Frankie
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 



From krice at freeswitch.org  Sat Apr  9 06:04:02 2011
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 08 Apr 2011 21:04:02 -0500
Subject: [Freeswitch-users] webphone app
In-Reply-To: <BANLkTi=6SL4irBVFshO9NQqJQc9Nhyo9gw@mail.gmail.com>
Message-ID: <C9C528C2.1350E%krice@freeswitch.org>

Flash Phones need a specific app server backend because they leverage RTMP
then on the server side use a plug-in for the app server to convert from
AMF-RTMP to SIP/RTP




On 4/8/11 8:13 PM, "budi wibowo" <bwibowo at gmail.com> wrote:

> check flashphoner,.. why flash sip phone seems need specific app server? how
> about connecting sip flash phone to generic sip server
> 
> On Thu, Apr 7, 2011 at 3:33 PM, peely <freeswitch at peely.com> wrote:
>> http://www.flashphoner.com/
>> 
>> It's commercial, but well worth it and very stable, unlike red5. It's built
>> on the wowza media server.
>> 
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/webphone-app-tp6248061p6249102.
>> 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

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From all.eforums at gmail.com  Sat Apr  9 06:18:33 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Fri, 8 Apr 2011 22:18:33 -0400
Subject: [Freeswitch-users] Optimal configuration for compiling on 64-bit
	platforms
Message-ID: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>

Hello,

I'm sure this has been talked about several times and searching through the
email archives etc. I have seen 64-bit been mentioned many times, but never
seen a targetted instructions as to what are the best/optimal parameters to
give the configure script to compile on a 64-bit Linux platform. In the
wiki, the following is given:


 CFLAGS=-m64 CXXFLAGS=-m64 LDFLAGS=-m64 ./configure
--prefix=/opt/freeswitch --enable-core-odbc-support \
--enable-core-libedit-support --enable-64 --with-openssl=/usr/sfw


But it's for compiling in Solaris. Do these flags/settings work on Linux?

Can we just use '--enable-64' and that would take care of everything or do
we need the extra CFLAGS, CXXFLAGS and LDFLAGS as well?

I'm using Debian 64-bit

Thanks
AEG
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From curriegrad2004 at gmail.com  Sat Apr  9 07:28:49 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 8 Apr 2011 20:28:49 -0700
Subject: [Freeswitch-users] Optimal configuration for compiling on
	64-bit platforms
In-Reply-To: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>
References: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>
Message-ID: <BANLkTim9D6JdC=0Eg0TATP=iocGqumJ0vg@mail.gmail.com>

You can specifiy -march options on the CXX and CFLAGS section, but
I've only seen noticeable performance increases on the 32-bit
platforms. I personally have "-O2 -g -march=pentium3" on the CXX and
CFLAGS section as I run a small home office setup on my P3 router ;)

On Fri, Apr 8, 2011 at 7:18 PM, A E [Gmail] <all.eforums at gmail.com> wrote:
> Hello,
> I'm sure this has been talked about several times and searching through the
> email archives etc. I have seen 64-bit been mentioned many times, but never
> seen a targetted instructions as to what are the best/optimal parameters to
> give the configure script to compile on a 64-bit Linux platform. In the
> wiki, the following is given:
>
>  CFLAGS=-m64 CXXFLAGS=-m64 LDFLAGS=-m64 ./configure --prefix=/opt/freeswitch
> --enable-core-odbc-support \
> --enable-core-libedit-support --enable-64 --with-openssl=/usr/sfw
>
> But it's for compiling in Solaris. Do these flags/settings work on Linux?
> Can we just use '--enable-64' and that would take care of everything or do
> we need the extra CFLAGS, CXXFLAGS and LDFLAGS as well?
> I'm using Debian 64-bit
> Thanks
> AEG
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From fieldpeak at gmail.com  Sat Apr  9 09:37:19 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sat, 9 Apr 2011 13:37:19 +0800
Subject: [Freeswitch-users] Failed to startup as daemon on CentOS 5.5
 with latest GIT head
In-Reply-To: <BANLkTimHjcQGmVo1fgfT8EY7D3D4SALt5g@mail.gmail.com>
References: <BANLkTinv4ob6BLYyFyCDj6bX4n+a2Wi6pA@mail.gmail.com>
	<BANLkTimHjcQGmVo1fgfT8EY7D3D4SALt5g@mail.gmail.com>
Message-ID: <BANLkTimkYykN5NbyqNMZizttDstL0qHJ8Q@mail.gmail.com>

It works after added "chkconfig --add freeswitch && chkconfig --levels 35
> freeswitch on", Thanks all for help!

Regards,
Charles
2011/4/9, curriegrad2004 <curriegrad2004 at gmail.com>:
> did you run "chkconfig --add freeswitch && chkconfig --levels 35
> freeswitch on" after you added the freeswitch init script to the
> init.d directory?
>
> On Fri, Apr 8, 2011 at 9:21 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>> I'm a newbie, and trying use scritp for FS to auto startup when OS start,
>> i
>> follow below link,
>>
>> http://wiki.freeswitch.org/wiki/Installation_Guide#Linux_and_Unix
>> http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora
>>
>> However, after the OS started for some time, the FS still not startup (by
>> register from eyebeam failure and use './freeswitch status' to check), if
>> i
>> manually excute the scritp (./freeswitch start), it works well.
>> Could anyone advise any clue to to resolve it?
>>
>> Attached is the startup script, FS running on CentOS 5.5 with latest GIT
>> head.
>>
>> Thanks
>>
>> Regards,
>> Charles
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From fvillarroel at yahoo.com  Sat Apr  9 19:17:31 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Sat, 9 Apr 2011 08:17:31 -0700 (PDT)
Subject: [Freeswitch-users] SIP to H323 on FS
Message-ID: <714881.75652.qm@web34307.mail.mud.yahoo.com>

Dear all.

I have FreeSWITCH Version 1.0.trunk (16526)

I need receive traffic SIP from a Gateway A and forward h323 traffic to another Gateway B like this:

Gateway A----SIP---> My FS ---H323----> Gateway B

It's possible and how i can do?

I am seeing the Wiki http://wiki.freeswitch.org/wiki/FreeSwitch_H323

What i need install to my FS Box?

How i can do in the Dialplan for send h323 call?
<action application="bridge" data="opal/h323:${dialed_extension}@${ip_gateway_B}"/>

It's Fine?
I hope you can help me

Fernando


From anthony.minessale at gmail.com  Sat Apr  9 23:54:35 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 9 Apr 2011 14:54:35 -0500
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <BANLkTinFCtayvYeO6FG5hB1ONHOADw8vGQ@mail.gmail.com>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>
	<41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>
	<AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>
	<BANLkTi=Mn0xEEojmmLsvyNpts02pR8akNA@mail.gmail.com>
	<4D9E225F.5070404@cupis.co.uk>
	<BANLkTinou6OZAMtCRzVsurULmBmnLLZ99A@mail.gmail.com>
	<BANLkTinFCtayvYeO6FG5hB1ONHOADw8vGQ@mail.gmail.com>
Message-ID: <BANLkTin7JtrE_kci927ffiF+FRKj04fxbA@mail.gmail.com>

The timerfd stuff is experimental so play around and see.
with 1000hz timer you don't need timerfd but it won't hurt.
Its very dependant on the motherboard and cpu etc.
 On Apr 8, 2011 11:46 AM, "A E [Gmail]" <all.eforums at gmail.com> wrote:
> On Fri, Apr 8, 2011 at 11:57 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> I think its relative to each kernel version.
>> The safe bet is to enable the 1000hz timer because 1ms is the least
>> amount of time FS needs to sleep.
>> Sometimes when you have a kernel that runs even faster the performance
>> goes down due to the extra cycles.
>> All I can say is test everything.
>>
>> Try it both ways with 1000hz and however the default is and if you
>> support timerfd try that too.
>> param enable-softtimer-timerfd set to true in switch.conf.xml and/or
>> using mod_timer_fd and setting rtp_timer_name=timerfd in your sofia
>> profile.
>>
>>
> Ok, Thanks Anthony. The default on my system was 250Hz. Have changed that
> and re-compiled the kernel. Will try out both and see what happens.
>
> BTW, do we need timerfd in conjunction with the 1000hz timer set in the
> kernel or is it either/or? As in does it affect positively or negatively
if
> we leave the kernel at 250Hz and enable timerfd as the timing source?
>
> Thanks
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From all.eforums at gmail.com  Sun Apr 10 00:21:40 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Sat, 9 Apr 2011 16:21:40 -0400
Subject: [Freeswitch-users] time_test on Centos 5.5
In-Reply-To: <BANLkTin7JtrE_kci927ffiF+FRKj04fxbA@mail.gmail.com>
References: <AANLkTiksCg70iEdZiN1+CnfyBW87mg7T4jDHjeq7JW5Y@mail.gmail.com>
	<0E4A540B-3E61-4940-8246-6FBF67CF91D8@ipeva.fr>
	<41F744A2-90FF-405F-AF62-5E7B8FB5128F@carmickle.com>
	<AANLkTi=NEPwvbF70y3j0p_jfSgLg=xBxW7Es1k7RgAVm@mail.gmail.com>
	<BANLkTi=Mn0xEEojmmLsvyNpts02pR8akNA@mail.gmail.com>
	<4D9E225F.5070404@cupis.co.uk>
	<BANLkTinou6OZAMtCRzVsurULmBmnLLZ99A@mail.gmail.com>
	<BANLkTinFCtayvYeO6FG5hB1ONHOADw8vGQ@mail.gmail.com>
	<BANLkTin7JtrE_kci927ffiF+FRKj04fxbA@mail.gmail.com>
Message-ID: <BANLkTiktJHK3nE_-K8ADBUT5Bx93tjQB=g@mail.gmail.com>

On Sat, Apr 9, 2011 at 3:54 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The timerfd stuff is experimental so play around and see.
> with 1000hz timer you don't need timerfd but it won't hurt.
> Its very dependant on the motherboard and cpu etc.
>
> Ok Cool. Will report back once I can get this going. Right now I can't even
get FS to compile on this platform. It can't find the libssl and I have all
packages installed. Like it finds it, but complains that those ones are
incompatible perhaps coz they're 32-bit and I'm trying to compile it using
the 64-bit switches fed to the configure switch. Ugh...wasted so much time
on this :(
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From gmaruzz at gmail.com  Sun Apr 10 02:23:54 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 10 Apr 2011 00:23:54 +0200
Subject: [Freeswitch-users] Optimal configuration for compiling on
	64-bit platforms
In-Reply-To: <BANLkTim9D6JdC=0Eg0TATP=iocGqumJ0vg@mail.gmail.com>
References: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>
	<BANLkTim9D6JdC=0Eg0TATP=iocGqumJ0vg@mail.gmail.com>
Message-ID: <BANLkTimTvaxy8DqSnDqnLxdt6CCBntt8xA@mail.gmail.com>

Compilation flags (if you do not compile on one machine kind and run
on another kind) are taken care of automatically by the
bootstrap/configure/make mechanism.
You are not supposed to do anything (actually, fiddling with those
options can damage performance/stability).
-giovanni


On 4/9/11, curriegrad2004 <curriegrad2004 at gmail.com> wrote:
> You can specifiy -march options on the CXX and CFLAGS section, but
> I've only seen noticeable performance increases on the 32-bit
> platforms. I personally have "-O2 -g -march=pentium3" on the CXX and
> CFLAGS section as I run a small home office setup on my P3 router ;)
>
> On Fri, Apr 8, 2011 at 7:18 PM, A E [Gmail] <all.eforums at gmail.com> wrote:
>> Hello,
>> I'm sure this has been talked about several times and searching through
>> the
>> email archives etc. I have seen 64-bit been mentioned many times, but
>> never
>> seen a targetted instructions as to what are the best/optimal parameters
>> to
>> give the configure script to compile on a 64-bit Linux platform. In the
>> wiki, the following is given:
>>
>>  CFLAGS=-m64 CXXFLAGS=-m64 LDFLAGS=-m64 ./configure
>> --prefix=/opt/freeswitch
>> --enable-core-odbc-support \
>> --enable-core-libedit-support --enable-64 --with-openssl=/usr/sfw
>>
>> But it's for compiling in Solaris. Do these flags/settings work on Linux?
>> Can we just use '--enable-64' and that would take care of everything or do
>> we need the extra CFLAGS, CXXFLAGS and LDFLAGS as well?
>> I'm using Debian 64-bit
>> Thanks
>> AEG
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From all.eforums at gmail.com  Sun Apr 10 02:38:49 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Sat, 9 Apr 2011 18:38:49 -0400
Subject: [Freeswitch-users] Optimal configuration for compiling on
	64-bit platforms
In-Reply-To: <BANLkTimTvaxy8DqSnDqnLxdt6CCBntt8xA@mail.gmail.com>
References: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>
	<BANLkTim9D6JdC=0Eg0TATP=iocGqumJ0vg@mail.gmail.com>
	<BANLkTimTvaxy8DqSnDqnLxdt6CCBntt8xA@mail.gmail.com>
Message-ID: <BANLkTi=SzJXjOSbcCF-+a-moFHCz4xpi8g@mail.gmail.com>

On Sat, Apr 9, 2011 at 6:23 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> Compilation flags (if you do not compile on one machine kind and run
> on another kind) are taken care of automatically by the
> bootstrap/configure/make mechanism.
> You are not supposed to do anything (actually, fiddling with those
> options can damage performance/stability).
> -giovanni
>
>
Thanks Giovanni. How about the -"-enable-64"  option in the configure
script? is that needed or should I just do a standard
./bootstrap.sh && ./configure && make && make install?

Thanks
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From rebel.pappas at gmail.com  Sat Apr  9 21:52:11 2011
From: rebel.pappas at gmail.com (alex pappas)
Date: Sat, 9 Apr 2011 20:52:11 +0300
Subject: [Freeswitch-users] Esl outbound connection with java
Message-ID: <BANLkTinfjTyBbvcoLVk+ZC63zt7U63arPw@mail.gmail.com>

Dear all,

I'm trying to start and learn - use the esl interface for creating
applications. I try this in java.
>From the esl example I see that i need to invoke the SocketClient in order
to start a socket and accept connections.
I don't understand the AbstractOutboundPipelineFactory class at all..

Does anyone can provide an example on how to start a socket deamon for this
use ? Like a kick start example in which I will be able to start and listen
for FreeSwitch connections.
Also I would like to ask if there is any limitation in using java for this.

Thank you in advance
Alex Pappas
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From curriegrad2004 at gmail.com  Sun Apr 10 02:41:42 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sat, 9 Apr 2011 15:41:42 -0700
Subject: [Freeswitch-users] Optimal configuration for compiling on
	64-bit platforms
In-Reply-To: <BANLkTimTvaxy8DqSnDqnLxdt6CCBntt8xA@mail.gmail.com>
References: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>
	<BANLkTim9D6JdC=0Eg0TATP=iocGqumJ0vg@mail.gmail.com>
	<BANLkTimTvaxy8DqSnDqnLxdt6CCBntt8xA@mail.gmail.com>
Message-ID: <BANLkTik1e-=HKvHwHdujGQXvZ177YfxkCA@mail.gmail.com>

It seems for me on a PIII under SL6/Fedora 12-13 that the
auto-optmization doesn't happen at all

On Sat, Apr 9, 2011 at 3:23 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> Compilation flags (if you do not compile on one machine kind and run
> on another kind) are taken care of automatically by the
> bootstrap/configure/make mechanism.
> You are not supposed to do anything (actually, fiddling with those
> options can damage performance/stability).
> -giovanni
>
>
> On 4/9/11, curriegrad2004 <curriegrad2004 at gmail.com> wrote:
>> You can specifiy -march options on the CXX and CFLAGS section, but
>> I've only seen noticeable performance increases on the 32-bit
>> platforms. I personally have "-O2 -g -march=pentium3" on the CXX and
>> CFLAGS section as I run a small home office setup on my P3 router ;)
>>
>> On Fri, Apr 8, 2011 at 7:18 PM, A E [Gmail] <all.eforums at gmail.com> wrote:
>>> Hello,
>>> I'm sure this has been talked about several times and searching through
>>> the
>>> email archives etc. I have seen 64-bit been mentioned many times, but
>>> never
>>> seen a targetted instructions as to what are the best/optimal parameters
>>> to
>>> give the configure script to compile on a 64-bit Linux platform. In the
>>> wiki, the following is given:
>>>
>>> ?CFLAGS=-m64 CXXFLAGS=-m64 LDFLAGS=-m64 ./configure
>>> --prefix=/opt/freeswitch
>>> --enable-core-odbc-support \
>>> --enable-core-libedit-support --enable-64 --with-openssl=/usr/sfw
>>>
>>> But it's for compiling in Solaris. Do these flags/settings work on Linux?
>>> Can we just use '--enable-64' and that would take care of everything or do
>>> we need the extra CFLAGS, CXXFLAGS and LDFLAGS as well?
>>> I'm using Debian 64-bit
>>> Thanks
>>> AEG
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>
> 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 all.eforums at gmail.com  Sun Apr 10 03:16:56 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Sat, 9 Apr 2011 19:16:56 -0400
Subject: [Freeswitch-users] Optimal configuration for compiling on
	64-bit platforms
In-Reply-To: <BANLkTik1e-=HKvHwHdujGQXvZ177YfxkCA@mail.gmail.com>
References: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>
	<BANLkTim9D6JdC=0Eg0TATP=iocGqumJ0vg@mail.gmail.com>
	<BANLkTimTvaxy8DqSnDqnLxdt6CCBntt8xA@mail.gmail.com>
	<BANLkTik1e-=HKvHwHdujGQXvZ177YfxkCA@mail.gmail.com>
Message-ID: <BANLkTi=TwAUzx9m1Mea4iwobSKnoDyQSaQ@mail.gmail.com>

On Sat, Apr 9, 2011 at 6:41 PM, curriegrad2004 <curriegrad2004 at gmail.com>wrote:

> It seems for me on a PIII under SL6/Fedora 12-13 that the
> auto-optmization doesn't happen at all
>
>
Well, I tried  the simple straight-forward

./bootstrap.sh && ./configure && make but got a WHOLE bunch of other errors,
which look much worse than the ones I got with compiling with the switches
forcing 64-bit build.

Looks like my only solution here might be to download source of openssl and
compile it on this machine and see if that gets accepted :(
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From msc at freeswitch.org  Sun Apr 10 03:22:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 9 Apr 2011 16:22:16 -0700
Subject: [Freeswitch-users] ESL with PHP not working
In-Reply-To: <BANLkTim4NmFsHarccjRW_edmD-oV0OVZtw@mail.gmail.com>
References: <BANLkTim4NmFsHarccjRW_edmD-oV0OVZtw@mail.gmail.com>
Message-ID: <BANLkTinccdzwfppewEUjJ=q8JkforE+J2g@mail.gmail.com>

On Fri, Apr 8, 2011 at 1:08 PM, Marc de Corny <marcdecorny at gmail.com> wrote:

> Hi all
>
> got an issue with ESL I cannot figure out.
>
> I have installed enabled the event socket on the Freeswitch, and it works
> locally on the server via PHP
>
> I have a remote server were I compiled the ESL.so and did the php-install.
> It is a standard CentOS install with apache.
> when I type into the command line :     php test.php   ( the standard test
> script that is api status ) I get the correct result.
> when I execute the same script from the browser on the remote server I get
> an error on the getBody command.
> *Fatal error*: Call to a member function getBody() on a non-object in *
> /var/www/html/test.php* on line *9*
>
> This is the exact same error that occurs when the script fails to connect
to the FS event socket, so something must be different when you are calling
this from a script. Can you confirm that the script is actually trying to
connect to the FreeSWITCH server? I don't know much about debugging PHP, but
at the very least you could do a tcpdump on the server running the php
script and confirm that the ESL connection is being attempted, whether it is
successful, etc.

-MC
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From gmaruzz at gmail.com  Sun Apr 10 03:40:24 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 10 Apr 2011 01:40:24 +0200
Subject: [Freeswitch-users] Optimal configuration for compiling on
	64-bit platforms
In-Reply-To: <BANLkTi=TwAUzx9m1Mea4iwobSKnoDyQSaQ@mail.gmail.com>
References: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>
	<BANLkTim9D6JdC=0Eg0TATP=iocGqumJ0vg@mail.gmail.com>
	<BANLkTimTvaxy8DqSnDqnLxdt6CCBntt8xA@mail.gmail.com>
	<BANLkTik1e-=HKvHwHdujGQXvZ177YfxkCA@mail.gmail.com>
	<BANLkTi=TwAUzx9m1Mea4iwobSKnoDyQSaQ@mail.gmail.com>
Message-ID: <BANLkTi=MVnyPTMdgDwkTLxbTbann6N959w@mail.gmail.com>

Compiler options have nothing to do with libraries incompatibilities.
Why don't you use the standard libraries given by your distro?
Never heard someone had to compile ssl libs for FreeSWITCH.
Also never heard someone used any compiler options on standard machines.
FS is targeted toward 64bit OSes, most people use CentOS, Ubuntu and
Debian. CentOS is by far the most used and the reference platform. On
64bit.
That said, if you're trying to optimize it for a PIII, an EPIA 5000 or
an ARM, or whatever is not a standard server class machine, yes you
better know how to optimize :).

On 4/10/11, A E [Gmail] <all.eforums at gmail.com> wrote:
> On Sat, Apr 9, 2011 at 6:41 PM, curriegrad2004
> <curriegrad2004 at gmail.com>wrote:
>
>> It seems for me on a PIII under SL6/Fedora 12-13 that the
>> auto-optmization doesn't happen at all
>>
>>
> Well, I tried  the simple straight-forward
>
> ./bootstrap.sh && ./configure && make but got a WHOLE bunch of other errors,
> which look much worse than the ones I got with compiling with the switches
> forcing 64-bit build.
>
> Looks like my only solution here might be to download source of openssl and
> compile it on this machine and see if that gets accepted :(
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From all.eforums at gmail.com  Sun Apr 10 03:51:44 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Sat, 9 Apr 2011 19:51:44 -0400
Subject: [Freeswitch-users] Optimal configuration for compiling on
	64-bit platforms
In-Reply-To: <BANLkTi=MVnyPTMdgDwkTLxbTbann6N959w@mail.gmail.com>
References: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>
	<BANLkTim9D6JdC=0Eg0TATP=iocGqumJ0vg@mail.gmail.com>
	<BANLkTimTvaxy8DqSnDqnLxdt6CCBntt8xA@mail.gmail.com>
	<BANLkTik1e-=HKvHwHdujGQXvZ177YfxkCA@mail.gmail.com>
	<BANLkTi=TwAUzx9m1Mea4iwobSKnoDyQSaQ@mail.gmail.com>
	<BANLkTi=MVnyPTMdgDwkTLxbTbann6N959w@mail.gmail.com>
Message-ID: <BANLkTimMTV0BbxU0HhD3JkqVrLyagtfxuw@mail.gmail.com>

On Sat, Apr 9, 2011 at 7:40 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> Compiler options have nothing to do with libraries incompatibilities.
> Why don't you use the standard libraries given by your distro?
> Never heard someone had to compile ssl libs for FreeSWITCH.
> Also never heard someone used any compiler options on standard machines.
> FS is targeted toward 64bit OSes, most people use CentOS, Ubuntu and
> Debian. CentOS is by far the most used and the reference platform. On
> 64bit.
> That said, if you're trying to optimize it for a PIII, an EPIA 5000 or
> an ARM, or whatever is not a standard server class machine, yes you
> better know how to optimize :).
>
>
Trust me, I wouldn't want to be messing around with compiler options, custom
libraries and  custom kernel, but so far I've had to do ALL of that. I am
just having the hardest time having FS even compile for me. I wasted a bunch
of days on various versions of Solaris and then gave up and moved on to
linux thinking this will at least have me up and running and EVEN if there
was any proof that FS would run faster/better on Sparc hardware with the
latest Solaris then I'd compromise with lesser performing OS and since
Debian seems to be the choice of the masses when it comes to using anything
other than Solaris on a 64-bit Sparc hardware, I went with that. And now I'm
having trouble compiling it on this too. Such a shame! And I can't even
pin-point exactly which module it's crashing in...all I see is it's trying
to build libfreeswitch.la and goes through the installed ssl packages and
says they're incompatible and then fails since it can't find -lssl *sigh*

Like I said, I've already tried doing the vanilla ./bootstrap.sh &&
./configure && make but that don't work. So I HAVE to mess around with
compiler options and env variables...but which ones and what exactly?
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From m.sobkow at marketelsystems.com  Sun Apr 10 03:49:22 2011
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Sat, 9 Apr 2011 17:49:22 -0600 (CST)
Subject: [Freeswitch-users] Can someone point me to examples of how to
 program message-based calls?
Message-ID: <32121300.77951302392962541.JavaMail.root@julie.marketel>

Recently the API behaviour changed for Erlang such that when you initiate a call, the routine immediately returns with a UUID of the launched call, rather than waiting for the call to be answered.  I've been able to program for this behaviour in the call-connected case, but I'm at a loss as to what to do for detecting calls that go unanswered, time out, or which cannot be placed for various technical reasons.

As a result, my call queue is filling up and choking -- I only launch so many calls at a time, and only the connected calls are getting properly processed.  Unanswered and timed out calls are getting "stuck" because I don't know what events to catch and how to evaluate them.

The old behaviour wasn't event-based and thereby easier to program -- but I can definitely see the advantages of shifting to a pure event-driven model.  I just need to learn how to _use_ it.

Thanks for any assistance you can provide.


From gmaruzz at celliax.org  Sun Apr 10 04:15:15 2011
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 10 Apr 2011 02:15:15 +0200
Subject: [Freeswitch-users] Optimal configuration for compiling on
	64-bit platforms
In-Reply-To: <BANLkTimMTV0BbxU0HhD3JkqVrLyagtfxuw@mail.gmail.com>
References: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>
	<BANLkTim9D6JdC=0Eg0TATP=iocGqumJ0vg@mail.gmail.com>
	<BANLkTimTvaxy8DqSnDqnLxdt6CCBntt8xA@mail.gmail.com>
	<BANLkTik1e-=HKvHwHdujGQXvZ177YfxkCA@mail.gmail.com>
	<BANLkTi=TwAUzx9m1Mea4iwobSKnoDyQSaQ@mail.gmail.com>
	<BANLkTi=MVnyPTMdgDwkTLxbTbann6N959w@mail.gmail.com>
	<BANLkTimMTV0BbxU0HhD3JkqVrLyagtfxuw@mail.gmail.com>
Message-ID: <BANLkTikTGDpWc6WAekMuGmq4sdBD3zPhiQ@mail.gmail.com>

Believe me, no.
Many people is using and compiling FS on debian without any hitch.
Is clearly described on the wiki.
Install the prerequisites, then bootstrap configure make install.
Maybe you have not installed the prerequisites, or your Debian
installation is somehow botched or damaged.
Maybe you want to reinstall Debian, then compile FS again.
-giovanni

On 4/10/11, A E [Gmail] <all.eforums at gmail.com> wrote:
> On Sat, Apr 9, 2011 at 7:40 PM, Giovanni Maruzzelli
> <gmaruzz at gmail.com>wrote:
>
>> Compiler options have nothing to do with libraries incompatibilities.
>> Why don't you use the standard libraries given by your distro?
>> Never heard someone had to compile ssl libs for FreeSWITCH.
>> Also never heard someone used any compiler options on standard machines.
>> FS is targeted toward 64bit OSes, most people use CentOS, Ubuntu and
>> Debian. CentOS is by far the most used and the reference platform. On
>> 64bit.
>> That said, if you're trying to optimize it for a PIII, an EPIA 5000 or
>> an ARM, or whatever is not a standard server class machine, yes you
>> better know how to optimize :).
>>
>>
> Trust me, I wouldn't want to be messing around with compiler options, custom
> libraries and  custom kernel, but so far I've had to do ALL of that. I am
> just having the hardest time having FS even compile for me. I wasted a bunch
> of days on various versions of Solaris and then gave up and moved on to
> linux thinking this will at least have me up and running and EVEN if there
> was any proof that FS would run faster/better on Sparc hardware with the
> latest Solaris then I'd compromise with lesser performing OS and since
> Debian seems to be the choice of the masses when it comes to using anything
> other than Solaris on a 64-bit Sparc hardware, I went with that. And now I'm
> having trouble compiling it on this too. Such a shame! And I can't even
> pin-point exactly which module it's crashing in...all I see is it's trying
> to build libfreeswitch.la and goes through the installed ssl packages and
> says they're incompatible and then fails since it can't find -lssl *sigh*
>
> Like I said, I've already tried doing the vanilla ./bootstrap.sh &&
> ./configure && make but that don't work. So I HAVE to mess around with
> compiler options and env variables...but which ones and what exactly?
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Sun Apr 10 04:35:27 2011
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 10 Apr 2011 02:35:27 +0200
Subject: [Freeswitch-users] Optimal configuration for compiling on
	64-bit platforms
In-Reply-To: <BANLkTikTGDpWc6WAekMuGmq4sdBD3zPhiQ@mail.gmail.com>
References: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>
	<BANLkTim9D6JdC=0Eg0TATP=iocGqumJ0vg@mail.gmail.com>
	<BANLkTimTvaxy8DqSnDqnLxdt6CCBntt8xA@mail.gmail.com>
	<BANLkTik1e-=HKvHwHdujGQXvZ177YfxkCA@mail.gmail.com>
	<BANLkTi=TwAUzx9m1Mea4iwobSKnoDyQSaQ@mail.gmail.com>
	<BANLkTi=MVnyPTMdgDwkTLxbTbann6N959w@mail.gmail.com>
	<BANLkTimMTV0BbxU0HhD3JkqVrLyagtfxuw@mail.gmail.com>
	<BANLkTikTGDpWc6WAekMuGmq4sdBD3zPhiQ@mail.gmail.com>
Message-ID: <BANLkTi=kUXvbE5iFRnaq3ECOWQbiHAHCuA@mail.gmail.com>

Also, maybe in one of your previous attempts you botched the FS source.
Delete it completely and git clone it again.
-giovanni

On 4/10/11, Giovanni Maruzzelli <gmaruzz at celliax.org> wrote:
> Believe me, no.
> Many people is using and compiling FS on debian without any hitch.
> Is clearly described on the wiki.
> Install the prerequisites, then bootstrap configure make install.
> Maybe you have not installed the prerequisites, or your Debian
> installation is somehow botched or damaged.
> Maybe you want to reinstall Debian, then compile FS again.
> -giovanni
>
> On 4/10/11, A E [Gmail] <all.eforums at gmail.com> wrote:
>> On Sat, Apr 9, 2011 at 7:40 PM, Giovanni Maruzzelli
>> <gmaruzz at gmail.com>wrote:
>>
>>> Compiler options have nothing to do with libraries incompatibilities.
>>> Why don't you use the standard libraries given by your distro?
>>> Never heard someone had to compile ssl libs for FreeSWITCH.
>>> Also never heard someone used any compiler options on standard machines.
>>> FS is targeted toward 64bit OSes, most people use CentOS, Ubuntu and
>>> Debian. CentOS is by far the most used and the reference platform. On
>>> 64bit.
>>> That said, if you're trying to optimize it for a PIII, an EPIA 5000 or
>>> an ARM, or whatever is not a standard server class machine, yes you
>>> better know how to optimize :).
>>>
>>>
>> Trust me, I wouldn't want to be messing around with compiler options,
>> custom
>> libraries and  custom kernel, but so far I've had to do ALL of that. I am
>> just having the hardest time having FS even compile for me. I wasted a
>> bunch
>> of days on various versions of Solaris and then gave up and moved on to
>> linux thinking this will at least have me up and running and EVEN if
>> there
>> was any proof that FS would run faster/better on Sparc hardware with the
>> latest Solaris then I'd compromise with lesser performing OS and since
>> Debian seems to be the choice of the masses when it comes to using
>> anything
>> other than Solaris on a 64-bit Sparc hardware, I went with that. And now
>> I'm
>> having trouble compiling it on this too. Such a shame! And I can't even
>> pin-point exactly which module it's crashing in...all I see is it's
>> trying
>> to build libfreeswitch.la and goes through the installed ssl packages and
>> says they're incompatible and then fails since it can't find -lssl *sigh*
>>
>> Like I said, I've already tried doing the vanilla ./bootstrap.sh &&
>> ./configure && make but that don't work. So I HAVE to mess around with
>> compiler options and env variables...but which ones and what exactly?
>>
>
> --
> Sent from my mobile device
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From all.eforums at gmail.com  Sun Apr 10 05:10:11 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Sat, 9 Apr 2011 21:10:11 -0400
Subject: [Freeswitch-users] Optimal configuration for compiling on
	64-bit platforms
In-Reply-To: <BANLkTikTGDpWc6WAekMuGmq4sdBD3zPhiQ@mail.gmail.com>
References: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>
	<BANLkTim9D6JdC=0Eg0TATP=iocGqumJ0vg@mail.gmail.com>
	<BANLkTimTvaxy8DqSnDqnLxdt6CCBntt8xA@mail.gmail.com>
	<BANLkTik1e-=HKvHwHdujGQXvZ177YfxkCA@mail.gmail.com>
	<BANLkTi=TwAUzx9m1Mea4iwobSKnoDyQSaQ@mail.gmail.com>
	<BANLkTi=MVnyPTMdgDwkTLxbTbann6N959w@mail.gmail.com>
	<BANLkTimMTV0BbxU0HhD3JkqVrLyagtfxuw@mail.gmail.com>
	<BANLkTikTGDpWc6WAekMuGmq4sdBD3zPhiQ@mail.gmail.com>
Message-ID: <BANLkTin6FSFZJDhmopqtojB4q5FJxG3xow@mail.gmail.com>

On Sat, Apr 9, 2011 at 8:15 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Believe me, no.
> Many people is using and compiling FS on debian without any hitch.
> Is clearly described on the wiki.
> Install the prerequisites, then bootstrap configure make install.
> Maybe you have not installed the prerequisites, or your Debian
> installation is somehow botched or damaged.
> Maybe you want to reinstall Debian, then compile FS again.
> -giovanni
>
>
Haha wow, you feel really strongly about that. I am not denying that many
people out there might be running successfully on debian 64-bit, I guess I
have the extra issue of it being on Sparc, which may OR may not have ALL
packages and the packages FS finds aren't to some compatibility level, which
I have no idea what that is. This is a fresh debian install so I have no
idea what might be botched on it but I'm cleaning out FS SRC and tried to
make it, and even though this time the 'configure' script finds the ssl libs
(seeing from grepping for them in config.log), it crashes way earlier than
before in trying to build sqlite. Here's the dump

make[1]: Entering directory `/home/fs/freeswitch/libs/sqlite'
./libtool --tag=CC --mode=link gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1
-DHAVE_FDATASYNC=1 -I. -I./src -DNDEBUG   -DTHREADSAFE=1
-DSQLITE_THREAD_OVERRIDE_LOCK=-1 -DSQLITE_OMIT_LOAD_EXTENSION=1
-DHAVE_READLINE=0  -lpthread  \
                -o sqlite3 ./src/shell.c libsqlite3.la \
                -lncurses
libtool: link: gcc -g -O2 -g -O2 -DOS_UNIX=1 -DHAVE_USLEEP=1
-DHAVE_FDATASYNC=1 -I. -I./src -DNDEBUG -DTHREADSAFE=1
-DSQLITE_THREAD_OVERRIDE_LOCK=-1 -DSQLITE_OMIT_LOAD_EXTENSION=1
-DHAVE_READLINE=0 -o sqlite3 ./src/shell.c  ./.libs/libsqlite3.a -lpthread
-lncurses
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(complete.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(main.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(os_unix.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(prepare.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(printf.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(random.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(table.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(tokenize.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(trigger.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(update.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(util.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9a architecture of input file
`./.libs/libsqlite3.a(vdbeapi.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(vdbeaux.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(vdbefifo.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9a architecture of input file
`./.libs/libsqlite3.a(vdbemem.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9a architecture of input file
`./.libs/libsqlite3.a(where.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(utf.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(legacy.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(vtab.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(analyze.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(attach.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(auth.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(btree.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(build.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(callback.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(delete.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(expr.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9a architecture of input file
`./.libs/libsqlite3.a(func.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(hash.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(insert.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(opcodes.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(os.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(pager.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(parse.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(pragma.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(select.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(vacuum.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9a architecture of input file
`./.libs/libsqlite3.a(vdbe.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(alter.o)' is incompatible with sparc:v8plus output
/usr/bin/ld: sparc:v9 architecture of input file
`./.libs/libsqlite3.a(date.o)' is incompatible with sparc:v8plus output
./.libs/libsqlite3.a(printf.o): In function `et_getdigit':
printf.c:(.text+0x144): undefined reference to `_Qp_qtoi'
printf.c:(.text+0x168): undefined reference to `_Qp_itoq'
printf.c:(.text+0x1a8): undefined reference to `_Qp_sub'
printf.c:(.text+0x1f4): undefined reference to `_Qp_mul'
./.libs/libsqlite3.a(printf.o): In function `vxprintf':
printf.c:(.text+0xd48): undefined reference to `_Qp_dtoq'
printf.c:(.text+0xdcc): undefined reference to `_Qp_flt'
printf.c:(.text+0xf14): undefined reference to `_Qp_dtoq'
printf.c:(.text+0xf54): undefined reference to `_Qp_add'
printf.c:(.text+0xfa0): undefined reference to `_Qp_fgt'
printf.c:(.text+0x1014): undefined reference to `_Qp_mul'
printf.c:(.text+0x1074): undefined reference to `_Qp_fge'
printf.c:(.text+0x10f8): undefined reference to `_Qp_mul'
printf.c:(.text+0x1164): undefined reference to `_Qp_fge'
printf.c:(.text+0x11e8): undefined reference to `_Qp_mul'
printf.c:(.text+0x1254): undefined reference to `_Qp_fge'
printf.c:(.text+0x12d8): undefined reference to `_Qp_mul'
printf.c:(.text+0x1344): undefined reference to `_Qp_flt'
printf.c:(.text+0x13c8): undefined reference to `_Qp_mul'
printf.c:(.text+0x1434): undefined reference to `_Qp_flt'
printf.c:(.text+0x14f0): undefined reference to `_Qp_dtoq'
printf.c:(.text+0x1530): undefined reference to `_Qp_add'
printf.c:(.text+0x1584): undefined reference to `_Qp_fge'
printf.c:(.text+0x15f0): undefined reference to `_Qp_mul'
./.libs/libsqlite3.a(util.o): In function `sqlite3AtoF':
util.c:(.text+0x1078): undefined reference to `_Qp_mul'
util.c:(.text+0x10ac): undefined reference to `_Qp_itoq'
util.c:(.text+0x10e4): undefined reference to `_Qp_add'
util.c:(.text+0x11c4): undefined reference to `_Qp_mul'
util.c:(.text+0x11f8): undefined reference to `_Qp_itoq'
util.c:(.text+0x1230): undefined reference to `_Qp_add'
util.c:(.text+0x126c): undefined reference to `_Qp_mul'
util.c:(.text+0x12d8): undefined reference to `_Qp_div'
util.c:(.text+0x1474): undefined reference to `_Qp_mul'
util.c:(.text+0x14d4): undefined reference to `_Qp_mul'
util.c:(.text+0x1534): undefined reference to `_Qp_mul'
util.c:(.text+0x1594): undefined reference to `_Qp_mul'
util.c:(.text+0x15e0): undefined reference to `_Qp_div'
util.c:(.text+0x1608): undefined reference to `_Qp_mul'
util.c:(.text+0x164c): undefined reference to `_Qp_qtod'
collect2: ld returned 1 exit status
make[1]: *** [sqlite3] Error 1
make[1]: Leaving directory `/home/fs/freeswitch/libs/sqlite'
make: *** [libs/sqlite/libsqlite3.la] Error 2
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From dave at dchorton.com  Sun Apr 10 08:29:02 2011
From: dave at dchorton.com (Dave Horton)
Date: Sun, 10 Apr 2011 00:29:02 -0400
Subject: [Freeswitch-users] how to pass arbitrary headers from A leg to B
	leg when bridging
Message-ID: <3D1E1744-EF86-442D-A632-93FEA621D822@dchorton.com>


Is there a way to pass any desired SIP header that I receive on an A leg through to the B leg?  I have looked at sofia_glue.c, and I see either special cases for certain headers being handled (e.g., Alert-Info), or support for only X- or P- custom headers.  I have a specific need to pass the Proxy-Authorization header that I receive on an A leg and have that same header and value contained on the B leg INVITE I generate (I have authentication turned off for this configuration, because I don't want FS generating challenges, I just want it to pass through the relevant headers).

Is this possible without hacking the code?

Dave



From dave at dchorton.com  Sun Apr 10 08:25:30 2011
From: dave at dchorton.com (Dave Horton)
Date: Sun, 10 Apr 2011 00:25:30 -0400
Subject: [Freeswitch-users] problem setting request uri params when bridging
	a call
Message-ID: <288AA948-04A4-4E2D-96AE-058ED7CC87A7@dchorton.com>

I'm relatively new to FS, so apologies if I'm missing something obvious..

I want to bridge an incoming call on the internal profile and have it go out to a configured gateway through the external profile, and I want any request-uri parameters that come in on the A leg to be also contained on the request-uri for the B leg.  My first attempt at doing that was to create this dialplan in the public context:

<extension name="epB" >
   <condition field="${sofia_profile_name}" expression="internal">
       <action application="set" data="export_vars=sip_req_params"/>
      <action application="bridge" data="sofia/gateway/gwB/${destination_number}"/>
   </condition>
</extension>

This did not work -- the INVITE went out from the external sip profile without the parameters that came in on the received INVITE.  The log messages did however indicate that the parameters were being exported:

EXECUTE sofia/internal/5684935846 at 204.215.65.202 set(export_vars=sip_req_params)
2011-04-10 00:17:07.350172 [DEBUG] mod_dptools.c:1059 sofia/internal/5684935846 at 204.215.65.202 SET [export_vars]=[sip_req_params]
EXECUTE sofia/internal/5684935846 at 204.215.65.202 bridge(sofia/gateway/gwB/15086160900)
2011-04-10 00:17:07.350172 [DEBUG] switch_channel.c:918 sofia/internal/5684935846 at 204.215.65.202 EXPORTING[export_vars] [sip_req_params]=[target=pcs_voip_originate] to event
2011-04-10 00:17:07.350172 [NOTICE] switch_channel.c:812 New Channel sofia/external/15086160900 [674ec775-befb-4b19-8e02-3b1f4159e1d6]

However, if I put them in as dial string variables, it works:

<extension name="epB" >
   <condition field="${sofia_profile_name}" expression="internal">
      <action application="bridge" data="{sip_invite_params=${sip_req_params}}sofia/gateway/gwB/${destination_number}"/>
   </condition>
</extension>

Shouldn't I be able to use the 'set' application and 'export_vars' to get variables from the A leg to be passed to the B leg and to be incorporated appropriately into the B leg INVITE?

(I am running a recent head build:  FreeSWITCH-mod_sofia/1.0.head-git-244fd68 2011-03-21 14-27-57 -0400)

Thanks in advance.

Dave

From marcdecorny at gmail.com  Sun Apr 10 11:04:58 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Sun, 10 Apr 2011 08:04:58 +0100
Subject: [Freeswitch-users] ESL with PHP not working
In-Reply-To: <BANLkTinccdzwfppewEUjJ=q8JkforE+J2g@mail.gmail.com>
References: <BANLkTim4NmFsHarccjRW_edmD-oV0OVZtw@mail.gmail.com>
	<BANLkTinccdzwfppewEUjJ=q8JkforE+J2g@mail.gmail.com>
Message-ID: <BANLkTin0b1ahnUHMt3cmCs=oHmnnb1tNiQ@mail.gmail.com>

Hi Michael,

thanks for your help. Yes that's what I thought, so I added a test on
connected() and realised it was not connecting. I also took a tcpdump on
both servers and could not see any packets being sent. It looks as if it is
stuck on the php server, but don't know where to start looking.

If there were information missing in php.ini would it work from the command
line ?
Is there a  break down of what that phpmod_install does so that I can do
back and check each step?

Thanks
Marc

On Sun, Apr 10, 2011 at 12:22 AM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Fri, Apr 8, 2011 at 1:08 PM, Marc de Corny <marcdecorny at gmail.com>wrote:
>
>> Hi all
>>
>> got an issue with ESL I cannot figure out.
>>
>> I have installed enabled the event socket on the Freeswitch, and it works
>> locally on the server via PHP
>>
>> I have a remote server were I compiled the ESL.so and did the php-install.
>> It is a standard CentOS install with apache.
>> when I type into the command line :     php test.php   ( the standard test
>> script that is api status ) I get the correct result.
>> when I execute the same script from the browser on the remote server I get
>> an error on the getBody command.
>> *Fatal error*: Call to a member function getBody() on a non-object in *
>> /var/www/html/test.php* on line *9*
>>
>> This is the exact same error that occurs when the script fails to connect
> to the FS event socket, so something must be different when you are calling
> this from a script. Can you confirm that the script is actually trying to
> connect to the FreeSWITCH server? I don't know much about debugging PHP, but
> at the very least you could do a tcpdump on the server running the php
> script and confirm that the ESL connection is being attempted, whether it is
> successful, etc.
>
> -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
>
>
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From peter.olsson at visionutveckling.se  Sun Apr 10 11:11:23 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 10 Apr 2011 09:11:23 +0200
Subject: [Freeswitch-users] Can someone point me to examples of how to
 program message-based calls?
In-Reply-To: <32121300.77951302392962541.JavaMail.root@julie.marketel>
References: <32121300.77951302392962541.JavaMail.root@julie.marketel>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDB9@cooper>

Are you sure it changed? Can't find anything about this in the git commit log recently.

I've never used Erlang myself, but this sounds like you originate a call, without the "ignore_early_media=true" flag. If this is not specified the originate command will return immediately when early media is detected, if you specify this channel variable it will wait for the call to be answered.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mark Sobkow [m.sobkow at marketelsystems.com]
Skickat: den 10 april 2011 01:49
Till: freeswitch-users
?mne: [Freeswitch-users] Can someone point me to examples of how to program message-based calls?

Recently the API behaviour changed for Erlang such that when you initiate a call, the routine immediately returns with a UUID of the launched call, rather than waiting for the call to be answered.  I've been able to program for this behaviour in the call-connected case, but I'm at a loss as to what to do for detecting calls that go unanswered, time out, or which cannot be placed for various technical reasons.

As a result, my call queue is filling up and choking -- I only launch so many calls at a time, and only the connected calls are getting properly processed.  Unanswered and timed out calls are getting "stuck" because I don't know what events to catch and how to evaluate them.

The old behaviour wasn't event-based and thereby easier to program -- but I can definitely see the advantages of shifting to a pure event-driven model.  I just need to learn how to _use_ it.

Thanks for any assistance you can provide.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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!DSPAM:4da0f41032761398017734!



From infos at madovsky.org  Sun Apr 10 18:55:33 2011
From: infos at madovsky.org (Madovsky)
Date: Sun, 10 Apr 2011 10:55:33 -0400
Subject: [Freeswitch-users] ESL with PHP not working
References: <BANLkTim4NmFsHarccjRW_edmD-oV0OVZtw@mail.gmail.com><BANLkTinccdzwfppewEUjJ=q8JkforE+J2g@mail.gmail.com>
	<BANLkTin0b1ahnUHMt3cmCs=oHmnnb1tNiQ@mail.gmail.com>
Message-ID: <5E9D852558FB432185306E05CDED18F0@e1705>

Marc,

depend how php has been compiled.
did you compile yourself ?
  ----- Original Message ----- 
  From: Marc de Corny 
  To: FreeSWITCH Users Help 
  Sent: Sunday, April 10, 2011 3:04 AM
  Subject: Re: [Freeswitch-users] ESL with PHP not working


  Hi Michael,


  thanks for your help. Yes that's what I thought, so I added a test on connected() and realised it was not connecting. I also took a tcpdump on both servers and could not see any packets being sent. It looks as if it is stuck on the php server, but don't know where to start looking.


  If there were information missing in php.ini would it work from the command line ?
  Is there a  break down of what that phpmod_install does so that I can do back and check each step?


  Thanks
  Marc


  On Sun, Apr 10, 2011 at 12:22 AM, Michael Collins <msc at freeswitch.org> wrote:




    On Fri, Apr 8, 2011 at 1:08 PM, Marc de Corny <marcdecorny at gmail.com> wrote:

      Hi all


      got an issue with ESL I cannot figure out.


      I have installed enabled the event socket on the Freeswitch, and it works locally on the server via PHP


      I have a remote server were I compiled the ESL.so and did the php-install. It is a standard CentOS install with apache.
      when I type into the command line :     php test.php   ( the standard test script that is api status ) I get the correct result.
      when I execute the same script from the browser on the remote server I get an error on the getBody command.
      Fatal error: Call to a member function getBody() on a non-object in /var/www/html/test.php on line 9


    This is the exact same error that occurs when the script fails to connect to the FS event socket, so something must be different when you are calling this from a script. Can you confirm that the script is actually trying to connect to the FreeSWITCH server? I don't know much about debugging PHP, but at the very least you could do a tcpdump on the server running the php script and confirm that the ESL connection is being attempted, whether it is successful, etc. 


    -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






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From jocke29 at gmail.com  Sun Apr 10 10:04:21 2011
From: jocke29 at gmail.com (jocke eriksson)
Date: Sun, 10 Apr 2011 08:04:21 +0200
Subject: [Freeswitch-users] FreeSwitch SCXML
Message-ID: <BANLkTik+4Mj-aXm08sS0VSdy1H0JY=SSng@mail.gmail.com>

Hello FreeSwitch user. I'm the owner of the freeswitch-scxml project,
the project is housed at google-code address
http://code.google.com/p/freeswitch-scxml/.
My intention with this project is to make users able to write SCXML
documents to handle there IVR. SCXML is small XML dialect that will
manage your
states and according to me it is well suited for IVR. The project is
far from feature complete but pretty easy to extend. I will be adding
more features so please feel free to drop in a feature request in the
issue tracker.

Regards Jocke Eriksson.


From fvillarroel at yahoo.com  Sun Apr 10 19:58:06 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Sun, 10 Apr 2011 08:58:06 -0700 (PDT)
Subject: [Freeswitch-users] Compiling mod_opal error
Message-ID: <178487.56361.qm@web34302.mail.mud.yahoo.com>

Dear.

I am trying of use mod_opal, i am use the the follow how to:

http://wiki.freeswitch.org/wiki/FreeSwitch_H323

But when i try of install mod_opal i receive the follow error:

cc:/usr/src/freeswitch.trunk# make mod_opal

making all mod_opal
Compiling /usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp...
Compiling /usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp ...
/usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp:67: error: invalid conversion from ???switch_call_cause_t (*)(switch_core_session_t*, switch_event_t*, switch_caller_profile_t*, switch_core_session_t**, switch_memory_pool_t**, switch_originate_flag_t)??? to ???switch_call_cause_t (*)(switch_core_session_t*, switch_event_t*, switch_caller_profile_t*, switch_core_session_t**, switch_memory_pool_t**, switch_originate_flag_t, switch_call_cause_t*)???
make[4]: *** [mod_opal.lo] Error 1
make[3]: *** [all] Error 1
make[2]: *** [mod_opal-all] Error 1
make[1]: *** [mod_opal] Error 2
make: *** [mod_opal] Error 2

I am using FreesWITCH version:

FreeSWITCH Version 1.0.trunk (16526)

I need support H323 on my FS. Anyone could help me.

Regards.


From fvillarroel at yahoo.com  Sun Apr 10 20:10:20 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Sun, 10 Apr 2011 09:10:20 -0700 (PDT)
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <BANLkTimzpJP2KjDsir8H=6q6nb7oJfOCSg@mail.gmail.com>
Message-ID: <386335.42948.qm@web34307.mail.mud.yahoo.com>

Dear Pablo.

Yes i can use condition but accountcode variable is assigned to only outbound gateway, but i need setup accoundcode for inbound gateway too. I not know how i can do this it's.


?????? <action application="set" data="accountcode=gateway1"/>

Regards.

--- On Fri, 4/8/11, Pablo Hernan Saro <pablosaro at gmail.com> wrote:

From: Pablo Hernan Saro <pablosaro at gmail.com>
Subject: Re: [Freeswitch-users] Account ACL
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Cc: "FERNANDO VILLARROEL" <fvillarroel at yahoo.com>
Date: Friday, April 8, 2011, 7:07 PM

Probably it's not the best for you, but the first solution that comes to my mind is setting the accountcode at dialplan. If you are working in a high performance scenario with lot of gateways, then probably you have an objection. Anyway, it would be something like this:


<extensio name="Setting account code for gateways">??<condition field="network_addr" expression="^192\.168\.1\.1$" break="on-true">
?? ?<action application="set" data="accountcode=gateway1"/>
??</condition>
??<condition field="network_addr" expression="^192\.168\.1\.2$" break="on-true">?? ?<action application="set" data="accountcode=gateway2"/>

??</condition></extension>
On Fri, Apr 8, 2011 at 6:05 PM, FERNANDO VILLARROEL <fvillarroel at yahoo.com> wrote:


Dear Mathieu.



My user are not in ditrectory.

My users are gateways authenticated with ACL.

How i can use some variable for identifi like accountcode for inbound traffic from this gateways???

Regards

--- On Fri, 4/8/11, Mathieu Rene <mrene_lists at avgs.ca> wrote:



From: Mathieu Rene <mrene_lists at avgs.ca>
Subject: Re: [Freeswitch-users] Account ACL


To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Friday, April 8, 2011, 4:00 PM



http://wiki.freeswitch.org/wiki/ACL#Users
You can set variables directly in the user's directory
 entry.

Mathieu ReneAvant-Garde Solutions IncOffice: + 1 (514) 664-1044 x100Cell: +1 (514) 664-1044 x200mrene at avgs.ca







On 2011-04-08, at 2:57 PM, FERNANDO VILLARROEL wrote:
Hi Community.

How i can identifi inbound traffic authorizated on ACL with some variable like Accountcode.



For aoutbound traffic i use:

 <action application="set" data="accountcode=foo"/>

My problem is for inbound traffic how i can identify accounts?

Regards.

_______________________________________________


FreeSWITCH-users mailing list
FreeSWITCH-users at 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
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FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org


http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users


http://www.freeswitch.org

_______________________________________________

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FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

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From peter.olsson at visionutveckling.se  Sun Apr 10 20:12:23 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 10 Apr 2011 18:12:23 +0200
Subject: [Freeswitch-users] Compiling mod_opal error
In-Reply-To: <178487.56361.qm@web34302.mail.mud.yahoo.com>
References: <178487.56361.qm@web34302.mail.mud.yahoo.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDBA@cooper>

Before trying anything else, SVN 16526 is really old, and the latest code is in git now. So please try latest GIT, and then report back.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r FERNANDO VILLARROEL [fvillarroel at yahoo.com]
Skickat: den 10 april 2011 17:58
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Compiling mod_opal error

Dear.

I am trying of use mod_opal, i am use the the follow how to:

http://wiki.freeswitch.org/wiki/FreeSwitch_H323

But when i try of install mod_opal i receive the follow error:

cc:/usr/src/freeswitch.trunk# make mod_opal

making all mod_opal
Compiling /usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp...
Compiling /usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp ...
/usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp:67: error: invalid conversion from ???switch_call_cause_t (*)(switch_core_session_t*, switch_event_t*, switch_caller_profile_t*, switch_core_session_t**, switch_memory_pool_t**, switch_originate_flag_t)??? to ???switch_call_cause_t (*)(switch_core_session_t*, switch_event_t*, switch_caller_profile_t*, switch_core_session_t**, switch_memory_pool_t**, switch_originate_flag_t, switch_call_cause_t*)???
make[4]: *** [mod_opal.lo] Error 1
make[3]: *** [all] Error 1
make[2]: *** [mod_opal-all] Error 1
make[1]: *** [mod_opal] Error 2
make: *** [mod_opal] Error 2

I am using FreesWITCH version:

FreeSWITCH Version 1.0.trunk (16526)

I need support H323 on my FS. Anyone could help me.

Regards.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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:4da1d49e32761530917628!



From boris at tagnet.ru  Sun Apr 10 20:18:53 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sun, 10 Apr 2011 22:18:53 +0600
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <386335.42948.qm@web34307.mail.mud.yahoo.com>
References: <386335.42948.qm@web34307.mail.mud.yahoo.com>
Message-ID: <4DA1D86D.5030402@tagnet.ru>

Hello!

     Why You can't use directory entry like this:

<user id="xxx" cidr="yyyy">
<variables>
<variable name="accountcode" value="zzz"/>
</variable>
</user>

> Dear Pablo.
>
> Yes i can use condition but accountcode variable is assigned to only 
> outbound gateway, but i need setup accoundcode for inbound gateway 
> too. I not know how i can do this it's.
>
>
> <action application="set" data="accountcode=gateway1"/>
>
> Regards.
>
> --- On *Fri, 4/8/11, Pablo Hernan Saro /<pablosaro at gmail.com>/* wrote:
>
>
>     From: Pablo Hernan Saro <pablosaro at gmail.com>
>     Subject: Re: [Freeswitch-users] Account ACL
>     To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>     Cc: "FERNANDO VILLARROEL" <fvillarroel at yahoo.com>
>     Date: Friday, April 8, 2011, 7:07 PM
>
>     Probably it's not the best for you, but the first solution that
>     comes to my mind is setting the accountcode at dialplan. If you
>     are working in a high performance scenario with lot of gateways,
>     then probably you have an objection. Anyway, it would be something
>     like this:
>
>     <extensio name="Setting account code for gateways">
>     <condition field="network_addr" expression="^192\.168\.1\.1$"
>     break="on-true">
>     <action application="set" data="accountcode=gateway1"/>
>     </condition>
>     <condition field="network_addr" expression="^192\.168\.1\.2$"
>     break="on-true">
>     <action application="set" data="accountcode=gateway2"/>
>     </condition>
>     </extension>
>
>     On Fri, Apr 8, 2011 at 6:05 PM, FERNANDO VILLARROEL
>     <fvillarroel at yahoo.com </mc/compose?to=fvillarroel at yahoo.com>> wrote:
>
>         Dear Mathieu.
>
>         My user are not in ditrectory.
>
>         My users are gateways authenticated with ACL.
>
>         How i can use some variable for identifi like accountcode for
>         inbound traffic from this gateways???
>
>         Regards
>
>         --- On *Fri, 4/8/11, Mathieu Rene /<mrene_lists at avgs.ca
>         </mc/compose?to=mrene_lists at avgs.ca>>/* wrote:
>
>
>             From: Mathieu Rene <mrene_lists at avgs.ca
>             </mc/compose?to=mrene_lists at avgs.ca>>
>             Subject: Re: [Freeswitch-users] Account ACL
>             To: "FreeSWITCH Users Help"
>             <freeswitch-users at lists.freeswitch.org
>             </mc/compose?to=freeswitch-users at lists.freeswitch.org>>
>             Date: Friday, April 8, 2011, 4:00 PM
>
>
>             http://wiki.freeswitch.org/wiki/ACL#Users
>
>             You can set variables directly in the user's directory entry.
>
>             Mathieu Rene
>             Avant-Garde Solutions Inc
>             Office: + 1 (514) 664-1044 x100
>             Cell: +1 (514) 664-1044 x200
>             mrene at avgs.ca <http://mc/compose?to=mrene at avgs.ca>
>
>
>
>
>             On 2011-04-08, at 2:57 PM, FERNANDO VILLARROEL wrote:
>
>>             Hi Community.
>>
>>             How i can identifi inbound traffic authorizated on ACL
>>             with some variable like Accountcode.
>>
>>             For aoutbound traffic i use:
>>
>>             <action application="set" data="accountcode=foo"/>
>>
>>             My problem is for inbound traffic how i can identify
>>             accounts?
>>
>>             Regards.
>>
>>             _______________________________________________
>>             FreeSWITCH-users mailing list
>>             FreeSWITCH-users at lists.freeswitch.org
>>             <http://mc/compose?to=FreeSWITCH-users at 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://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>             http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>             UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>             http://www.freeswitch.org
>
>
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         </mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991


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From peter.olsson at visionutveckling.se  Sun Apr 10 20:17:39 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 10 Apr 2011 18:17:39 +0200
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <386335.42948.qm@web34307.mail.mud.yahoo.com>
References: <BANLkTimzpJP2KjDsir8H=6q6nb7oJfOCSg@mail.gmail.com>,
	<386335.42948.qm@web34307.mail.mud.yahoo.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDBB@cooper>

If you use the dial plan example suggested it will set that variable on incoming calls from your gateways. If the gateway IP matched the IP entered it will be executed.

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r FERNANDO VILLARROEL [fvillarroel at yahoo.com]
Skickat: den 10 april 2011 18:10
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] Account ACL

Dear Pablo.

Yes i can use condition but accountcode variable is assigned to only outbound gateway, but i need setup accoundcode for inbound gateway too. I not know how i can do this it's.


       <action application="set" data="accountcode=gateway1"/>

Regards.

--- On Fri, 4/8/11, Pablo Hernan Saro <pablosaro at gmail.com> wrote:

From: Pablo Hernan Saro <pablosaro at gmail.com>
Subject: Re: [Freeswitch-users] Account ACL
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Cc: "FERNANDO VILLARROEL" <fvillarroel at yahoo.com>
Date: Friday, April 8, 2011, 7:07 PM

Probably it's not the best for you, but the first solution that comes to my mind is setting the accountcode at dialplan. If you are working in a high performance scenario with lot of gateways, then probably you have an objection. Anyway, it would be something like this:

<extensio name="Setting account code for gateways">
  <condition field="network_addr" expression="^192\.168\.1\.1$" break="on-true">
    <action application="set" data="accountcode=gateway1"/>
  </condition>
  <condition field="network_addr" expression="^192\.168\.1\.2$" break="on-true">
    <action application="set" data="accountcode=gateway2"/>
  </condition>
</extension>

On Fri, Apr 8, 2011 at 6:05 PM, FERNANDO VILLARROEL <fvillarroel at yahoo.com</mc/compose?to=fvillarroel at yahoo.com>> wrote:
Dear Mathieu.

My user are not in ditrectory.

My users are gateways authenticated with ACL.

How i can use some variable for identifi like accountcode for inbound traffic from this gateways???

Regards

--- On Fri, 4/8/11, Mathieu Rene <mrene_lists at avgs.ca</mc/compose?to=mrene_lists at avgs.ca>> wrote:

From: Mathieu Rene <mrene_lists at avgs.ca</mc/compose?to=mrene_lists at avgs.ca>>
Subject: Re: [Freeswitch-users] Account ACL
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org</mc/compose?to=freeswitch-users at lists.freeswitch.org>>
Date: Friday, April 8, 2011, 4:00 PM


http://wiki.freeswitch.org/wiki/ACL#Users

You can set variables directly in the user's directory entry.
<http://wiki.freeswitch.org/wiki/ACL#Users>
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<http://mc/compose?to=mrene at avgs.ca>




On 2011-04-08, at 2:57 PM, FERNANDO VILLARROEL wrote:

Hi Community.

How i can identifi inbound traffic authorizated on ACL with some variable like Accountcode.

For aoutbound traffic i use:

<action application="set" data="accountcode=foo"/>

My problem is for inbound traffic how i can identify accounts?

Regards.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at 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-----


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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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_______________________________________________
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FreeSWITCH-users at lists.freeswitch.org</mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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!DSPAM:4da1d72c32761895375181!


From peter.olsson at visionutveckling.se  Sun Apr 10 20:19:46 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 10 Apr 2011 18:19:46 +0200
Subject: [Freeswitch-users] Compiling mod_opal error
In-Reply-To: <178487.56361.qm@web34302.mail.mud.yahoo.com>
References: <178487.56361.qm@web34302.mail.mud.yahoo.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDBC@cooper>

And also - bugs go into Jira - not the mailing list :)

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r FERNANDO VILLARROEL [fvillarroel at yahoo.com]
Skickat: den 10 april 2011 17:58
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Compiling mod_opal error

Dear.

I am trying of use mod_opal, i am use the the follow how to:

http://wiki.freeswitch.org/wiki/FreeSwitch_H323

But when i try of install mod_opal i receive the follow error:

cc:/usr/src/freeswitch.trunk# make mod_opal

making all mod_opal
Compiling /usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp...
Compiling /usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp ...
/usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp:67: error: invalid conversion from ???switch_call_cause_t (*)(switch_core_session_t*, switch_event_t*, switch_caller_profile_t*, switch_core_session_t**, switch_memory_pool_t**, switch_originate_flag_t)??? to ???switch_call_cause_t (*)(switch_core_session_t*, switch_event_t*, switch_caller_profile_t*, switch_core_session_t**, switch_memory_pool_t**, switch_originate_flag_t, switch_call_cause_t*)???
make[4]: *** [mod_opal.lo] Error 1
make[3]: *** [all] Error 1
make[2]: *** [mod_opal-all] Error 1
make[1]: *** [mod_opal] Error 2
make: *** [mod_opal] Error 2

I am using FreesWITCH version:

FreeSWITCH Version 1.0.trunk (16526)

I need support H323 on my FS. Anyone could help me.

Regards.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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:4da1d49e32761530917628!



From pablosaro at gmail.com  Sun Apr 10 21:45:10 2011
From: pablosaro at gmail.com (Pablo Hernan Saro)
Date: Sun, 10 Apr 2011 14:45:10 -0300
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDBB@cooper>
References: <BANLkTimzpJP2KjDsir8H=6q6nb7oJfOCSg@mail.gmail.com>
	<386335.42948.qm@web34307.mail.mud.yahoo.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDBB@cooper>
Message-ID: <BANLkTi=-uSqA6j62H4qjMhtswMOA8Re8Xw@mail.gmail.com>

Hi Fernando, using the dialplan I've suggested you will have the account
code set on cdr when one of the conditions match (that is when incoming
calls arrive to your FS box from one of the IP addresses in the conditions).
I'm not sure if it can be rewritten by outbound gateway definition (is that
your problem?).
Try that and check your cdr to see which accountcode it's being saved (the
one you set at dialplan or the one set in outbound gateway definition).

On Sun, Apr 10, 2011 at 1:17 PM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> If you use the dial plan example suggested it will set that variable on
> incoming calls from your gateways. If the gateway IP matched the IP entered
> it will be executed.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r FERNANDO
> VILLARROEL [fvillarroel at yahoo.com]
> Skickat: den 10 april 2011 18:10
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: Re: [Freeswitch-users] Account ACL
>
> Dear Pablo.
>
> Yes i can use condition but accountcode variable is assigned to only
> outbound gateway, but i need setup accoundcode for inbound gateway too. I
> not know how i can do this it's.
>
>
>       <action application="set" data="accountcode=gateway1"/>
>
> Regards.
>
> --- On Fri, 4/8/11, Pablo Hernan Saro <pablosaro at gmail.com> wrote:
>
> From: Pablo Hernan Saro <pablosaro at gmail.com>
> Subject: Re: [Freeswitch-users] Account ACL
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Cc: "FERNANDO VILLARROEL" <fvillarroel at yahoo.com>
> Date: Friday, April 8, 2011, 7:07 PM
>
> Probably it's not the best for you, but the first solution that comes to my
> mind is setting the accountcode at dialplan. If you are working in a high
> performance scenario with lot of gateways, then probably you have an
> objection. Anyway, it would be something like this:
>
> <extensio name="Setting account code for gateways">
>  <condition field="network_addr" expression="^192\.168\.1\.1$"
> break="on-true">
>    <action application="set" data="accountcode=gateway1"/>
>  </condition>
>  <condition field="network_addr" expression="^192\.168\.1\.2$"
> break="on-true">
>    <action application="set" data="accountcode=gateway2"/>
>  </condition>
> </extension>
>
> On Fri, Apr 8, 2011 at 6:05 PM, FERNANDO VILLARROEL <fvillarroel at yahoo.com
> </mc/compose?to=fvillarroel at yahoo.com>> wrote:
> Dear Mathieu.
>
> My user are not in ditrectory.
>
> My users are gateways authenticated with ACL.
>
> How i can use some variable for identifi like accountcode for inbound
> traffic from this gateways???
>
> Regards
>
> --- On Fri, 4/8/11, Mathieu Rene <mrene_lists at avgs.ca</mc/compose?to=
> mrene_lists at avgs.ca>> wrote:
>
> From: Mathieu Rene <mrene_lists at avgs.ca</mc/compose?to=mrene_lists at avgs.ca
> >>
> Subject: Re: [Freeswitch-users] Account ACL
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org
> </mc/compose?to=freeswitch-users at lists.freeswitch.org>>
> Date: Friday, April 8, 2011, 4:00 PM
>
>
> http://wiki.freeswitch.org/wiki/ACL#Users
>
> You can set variables directly in the user's directory entry.
> <http://wiki.freeswitch.org/wiki/ACL#Users>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca<http://mc/compose?to=mrene at avgs.ca>
>
>
>
>
> On 2011-04-08, at 2:57 PM, FERNANDO VILLARROEL wrote:
>
> Hi Community.
>
> How i can identifi inbound traffic authorizated on ACL with some variable
> like Accountcode.
>
> For aoutbound traffic i use:
>
> <action application="set" data="accountcode=foo"/>
>
> My problem is for inbound traffic how i can identify accounts?
>
> Regards.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<
> http://mc/compose?to=FreeSWITCH-users at 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://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org</mc/compose?to=
> FreeSWITCH-users at 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:4da1d72c32761895375181!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From marcdecorny at gmail.com  Sun Apr 10 23:10:51 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Sun, 10 Apr 2011 20:10:51 +0100
Subject: [Freeswitch-users] ESL with PHP not working
In-Reply-To: <5E9D852558FB432185306E05CDED18F0@e1705>
References: <BANLkTim4NmFsHarccjRW_edmD-oV0OVZtw@mail.gmail.com>
	<BANLkTinccdzwfppewEUjJ=q8JkforE+J2g@mail.gmail.com>
	<BANLkTin0b1ahnUHMt3cmCs=oHmnnb1tNiQ@mail.gmail.com>
	<5E9D852558FB432185306E05CDED18F0@e1705>
Message-ID: <BANLkTi=SpZSLFpuZSPNFNHLky8criP69EQ@mail.gmail.com>

no I am using Centos and downloaded it with yum
thanks
Marc

On Sun, Apr 10, 2011 at 3:55 PM, Madovsky <infos at madovsky.org> wrote:

>  Marc,
>
> depend how php has been compiled.
> did you compile yourself ?
>
> ----- Original Message -----
> *From:* Marc de Corny <marcdecorny at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Sunday, April 10, 2011 3:04 AM
> *Subject:* Re: [Freeswitch-users] ESL with PHP not working
>
> Hi Michael,
>
> thanks for your help. Yes that's what I thought, so I added a test on
> connected() and realised it was not connecting. I also took a tcpdump on
> both servers and could not see any packets being sent. It looks as if it is
> stuck on the php server, but don't know where to start looking.
>
> If there were information missing in php.ini would it work from the command
> line ?
> Is there a  break down of what that phpmod_install does so that I can do
> back and check each step?
>
> Thanks
> Marc
>
> On Sun, Apr 10, 2011 at 12:22 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>>  On Fri, Apr 8, 2011 at 1:08 PM, Marc de Corny <marcdecorny at gmail.com>wrote:
>>
>>> Hi all
>>>
>>> got an issue with ESL I cannot figure out.
>>>
>>> I have installed enabled the event socket on the Freeswitch, and it works
>>> locally on the server via PHP
>>>
>>> I have a remote server were I compiled the ESL.so and did the
>>> php-install. It is a standard CentOS install with apache.
>>> when I type into the command line :     php test.php   ( the standard
>>> test script that is api status ) I get the correct result.
>>> when I execute the same script from the browser on the remote server I
>>> get an error on the getBody command.
>>> *Fatal error*: Call to a member function getBody() on a non-object in *
>>> /var/www/html/test.php* on line *9*
>>>
>>> This is the exact same error that occurs when the script fails to connect
>> to the FS event socket, so something must be different when you are calling
>> this from a script. Can you confirm that the script is actually trying to
>> connect to the FreeSWITCH server? I don't know much about debugging PHP, but
>> at the very least you could do a tcpdump on the server running the php
>> script and confirm that the ESL connection is being attempted, whether it is
>> successful, etc.
>>
>> -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
>
>
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From fvillarroel at yahoo.com  Mon Apr 11 02:27:46 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Sun, 10 Apr 2011 15:27:46 -0700 (PDT)
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <BANLkTi=-uSqA6j62H4qjMhtswMOA8Re8Xw@mail.gmail.com>
Message-ID: <952419.41944.qm@web34308.mail.mud.yahoo.com>

Dear Pablo.

Thank you for you help.

My problem is setup accountcode for outbound gateway (sip_profiles/external ). I will try and inform to you my tests results.

For inbound gateway the acoountcode is setup fine the  like this:

?????? <action application="set" data="accountcode=foo"/>
?????? <action application="bridge" data="sofia/gateway/ms6/$1$2"/>

I need setup CDR accountcode for my sip_profiles/external or in this example for gateway named ms6.

<include>
??? <gateway name="ms6" cidr="XXX.xx.XXX.170/32">
????? <param name="username" value="foo"/>
????? <param name="password" value="foo at fool"/>
????? <param name="realm" value="XXX.XX.XXX.170/32"/>
????? <param name="proxy" value="XXX.XX.XXX.170/32"/>
????? <param name="register" value="true"/>
????? <param name="context" value="public"/>
??? </gateway>
</include>

Regards.


Regards
--- On Sun, 4/10/11, Pablo Hernan Saro <pablosaro at gmail.com> wrote:

From: Pablo Hernan Saro <pablosaro at gmail.com>
Subject: Re: [Freeswitch-users] Account ACL
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Sunday, April 10, 2011, 2:45 PM

Hi Fernando, using the dialplan I've suggested you will have the account code set on cdr when one of the conditions match (that is when incoming calls arrive to your FS box from one of the IP addresses in the conditions). I'm not sure if it can be rewritten by outbound gateway definition (is that your problem?).

Try that and check your cdr to see which accountcode it's being saved (the one you set at dialplan or the one set in outbound gateway definition).

On Sun, Apr 10, 2011 at 1:17 PM, Peter Olsson <peter.olsson at visionutveckling.se> wrote:


If you use the dial plan example suggested it will set that variable on incoming calls from your gateways. If the gateway IP matched the IP entered it will be executed.





/Peter

________________________________________

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r FERNANDO VILLARROEL [fvillarroel at yahoo.com]



Skickat: den 10 april 2011 18:10

Till: freeswitch-users at lists.freeswitch.org

?mne: Re: [Freeswitch-users] Account ACL



Dear Pablo.



Yes i can use condition but accountcode variable is assigned to only outbound gateway, but i need setup accoundcode for inbound gateway too. I not know how i can do this it's.





 ? ? ? <action application="set" data="accountcode=gateway1"/>



Regards.



--- On Fri, 4/8/11, Pablo Hernan Saro <pablosaro at gmail.com> wrote:



From: Pablo Hernan Saro <pablosaro at gmail.com>

Subject: Re: [Freeswitch-users] Account ACL

To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

Cc: "FERNANDO VILLARROEL" <fvillarroel at yahoo.com>

Date: Friday, April 8, 2011, 7:07 PM



Probably it's not the best for you, but the first solution that comes to my mind is setting the accountcode at dialplan. If you are working in a high performance scenario with lot of gateways, then probably you have an objection. Anyway, it would be something like this:





<extensio name="Setting account code for gateways">

 ?<condition field="network_addr" expression="^192\.168\.1\.1$" break="on-true">

 ? ?<action application="set" data="accountcode=gateway1"/>

 ?</condition>

 ?<condition field="network_addr" expression="^192\.168\.1\.2$" break="on-true">

 ? ?<action application="set" data="accountcode=gateway2"/>

 ?</condition>

</extension>



On Fri, Apr 8, 2011 at 6:05 PM, FERNANDO VILLARROEL <fvillarroel at yahoo.com</mc/compose?to=fvillarroel at yahoo.com>> wrote:



Dear Mathieu.



My user are not in ditrectory.



My users are gateways authenticated with ACL.



How i can use some variable for identifi like accountcode for inbound traffic from this gateways???



Regards



--- On Fri, 4/8/11, Mathieu Rene <mrene_lists at avgs.ca</mc/compose?to=mrene_lists at avgs.ca>> wrote:



From: Mathieu Rene <mrene_lists at avgs.ca</mc/compose?to=mrene_lists at avgs.ca>>

Subject: Re: [Freeswitch-users] Account ACL

To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org</mc/compose?to=freeswitch-users at lists.freeswitch.org>>



Date: Friday, April 8, 2011, 4:00 PM





http://wiki.freeswitch.org/wiki/ACL#Users



You can set variables directly in the user's directory entry.

<http://wiki.freeswitch.org/wiki/ACL#Users>

Mathieu Rene

Avant-Garde Solutions Inc

Office: + 1 (514) 664-1044 x100

Cell: +1 (514) 664-1044 x200

mrene at avgs.ca<http://mc/compose?to=mrene at avgs.ca>









On 2011-04-08, at 2:57 PM, FERNANDO VILLARROEL wrote:



Hi Community.



How i can identifi inbound traffic authorizated on ACL with some variable like Accountcode.



For aoutbound traffic i use:



<action application="set" data="accountcode=foo"/>



My problem is for inbound traffic how i can identify accounts?



Regards.



_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at 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://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>



http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

http://www.freeswitch.org





_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at lists.freeswitch.org</mc/compose?to=FreeSWITCH-users at 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:4da1d72c32761895375181!



_______________________________________________

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FreeSWITCH-users at lists.freeswitch.org

http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

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From all.eforums at gmail.com  Mon Apr 11 03:00:51 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Sun, 10 Apr 2011 19:00:51 -0400
Subject: [Freeswitch-users] Optimal configuration for compiling on
	64-bit platforms
In-Reply-To: <BANLkTi=MVnyPTMdgDwkTLxbTbann6N959w@mail.gmail.com>
References: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>
	<BANLkTim9D6JdC=0Eg0TATP=iocGqumJ0vg@mail.gmail.com>
	<BANLkTimTvaxy8DqSnDqnLxdt6CCBntt8xA@mail.gmail.com>
	<BANLkTik1e-=HKvHwHdujGQXvZ177YfxkCA@mail.gmail.com>
	<BANLkTi=TwAUzx9m1Mea4iwobSKnoDyQSaQ@mail.gmail.com>
	<BANLkTi=MVnyPTMdgDwkTLxbTbann6N959w@mail.gmail.com>
Message-ID: <BANLkTikU9K5FCaCFJ9h=i3FqU1OccX7eyg@mail.gmail.com>

On Sat, Apr 9, 2011 at 7:40 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> Compiler options have nothing to do with libraries incompatibilities.
> Why don't you use the standard libraries given by your distro?
> Never heard someone had to compile ssl libs for FreeSWITCH.
> Also never heard someone used any compiler options on standard machines.
> FS is targeted toward 64bit OSes, most people use CentOS, Ubuntu and
> Debian. CentOS is by far the most used and the reference platform. On
> 64bit.
> That said, if you're trying to optimize it for a PIII, an EPIA 5000 or
> an ARM, or whatever is not a standard server class machine, yes you
> better know how to optimize :).
>
>
Well 'you told me so' Giovanni ;)

It's finally built! I deleted the entire source directory, did a brand new
git clone, bootstrap and the whole process with NO switches, flags or
options given to 'configure'

But the 'configure' etc did NOT take care of the fact that it's a 64-bit
platform, proven by the following:

$ file ./.libs/freeswitch
./.libs/freeswitch: ELF 32-bit MSB executable, SPARC32PLUS, V8+ Required,
version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux
2.6.18, not stripped

Now I have heard from people in the debian-sparc forum that in the case of
debian port, sparc32 is in fact faster than sparc64 and as a result the
current userland is actually 32-bit. If that applies to FS built as 32-bit
in this environment to be faster or comparable in performance to 64-bit,
then I'm golden. Else, I guess I have no other choice as I can't figure out
how to build it and get past the ssl problem I was having with all those m64
flags!
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From msc at freeswitch.org  Mon Apr 11 05:57:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 10 Apr 2011 18:57:40 -0700
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <952419.41944.qm@web34308.mail.mud.yahoo.com>
References: <BANLkTi=-uSqA6j62H4qjMhtswMOA8Re8Xw@mail.gmail.com>
	<952419.41944.qm@web34308.mail.mud.yahoo.com>
Message-ID: <BANLkTimLGyyv6AtSaS2k3YKjRzhtyZHcxA@mail.gmail.com>

On Sun, Apr 10, 2011 at 3:27 PM, FERNANDO VILLARROEL
<fvillarroel at yahoo.com>wrote:

> Dear Pablo.
>
> Thank you for you help.
>
> My problem is setup accountcode for outbound gateway (sip_profiles/external
> ). I will try and inform to you my tests results.
>
> For inbound gateway the acoountcode is setup fine the like this:
>
>
>        <action application="set" data="accountcode=foo"/>
>        <action application="bridge" data="sofia/gateway/ms6/$1$2"/>
>
> I need setup CDR accountcode for my sip_profiles/external or in this
> example for gateway named ms6.
>

So, you need the variable "accountcode" to be "foo" on the B leg of this
call? If so, just use "export" instead of "set" and it will be done
 <action application="export" data="accountcode=foo"/>

If that's not what you want then I recommend that you try explaining from
the beginning (again) your problem because we are having a difficult time
understanding what you are asking for.

Thanks,
MC
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From jason at jasonjgw.net  Mon Apr 11 05:46:48 2011
From: jason at jasonjgw.net (Jason White)
Date: Mon, 11 Apr 2011 01:46:48 +0000 (UTC)
Subject: [Freeswitch-users] FreeSWITCH sometimes binds to loopback interface
	during boot
Message-ID: <intmi8$e9b$1@dough.gmane.org>

I have FreeSWITCH installed under Debian; it is started by the init script
during the boot process.

External connectivity is provided via an ADSL line, attached to a PCI modem
card in the machine. (Specifically, it's a Traverse Technologies Solos card.)
Pppd handles the ADSL interface.

Now for the problem: sometimes, the external SIP profile binds to 127.0.0.1,
presumably because FreeSWITCH is started after the network has been brought up
but before the ppp0 interface is established. Restarting the profile with
sofia profile external restart does not correct it - I have to restart
FreeSWITCH.

Is there a better solution - preferably, a means of binding to the correct
address without restarting FreeSWITCH? I can create scripts to execute a
FreeSWITCH command whenever the PPP interface comes up, but I'd rather avoid a
total shutdown/restart in that case.




From all.eforums at gmail.com  Mon Apr 11 08:40:57 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 11 Apr 2011 00:40:57 -0400
Subject: [Freeswitch-users] Core Dump after compiling on Debian for Sparc
Message-ID: <BANLkTin3moUaFNVY-wtFOxX5TW1cKWGpMA@mail.gmail.com>

Hi Guys,

So I finally got Freeswitch to compile on my Sparc machine running 64-bit
debian. As was mentioned earlier, it was compiled with NO CXX flags or other
options provided to the configure script. However, when I start freeswitch,
it core dumps.

All relevant information is available on:
http://pastebin.freeswitch.org/16064

<http://pastebin.freeswitch.org/16064>Can someone please look at it and let
me know if this is just me or this hints at a bug that no one else ran into.
I'm running the latest from Git.

Thanks so much
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From steveayre at gmail.com  Mon Apr 11 10:50:50 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Apr 2011 07:50:50 +0100
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <453850.62406.qm@web34303.mail.mud.yahoo.com>
References: <453850.62406.qm@web34303.mail.mud.yahoo.com>
Message-ID: <F01DDEF9-D207-4838-85E6-3C238F79373D@gmail.com>

You can set channel variables in the <gateway> section, and you can specify the direction too so that variables get set either on inbound, outbound or both.

I *think* the syntax is:

<gateway>
   ...
   <variables>
       <variable name="accountcode" value="ABC" direction="inbound"/>
       ...
   </variables>
</gateway>

Steve on iPhone

On 8 Apr 2011, at 22:05, FERNANDO VILLARROEL <fvillarroel at yahoo.com> wrote:

> Dear Mathieu.
> 
> My user are not in ditrectory.
> 
> My users are gateways authenticated with ACL.
> 
> How i can use some variable for identifi like accountcode for inbound traffic from this gateways???
> 
> Regards
> 
> --- On Fri, 4/8/11, Mathieu Rene <mrene_lists at avgs.ca> wrote:
> 
> From: Mathieu Rene <mrene_lists at avgs.ca>
> Subject: Re: [Freeswitch-users] Account ACL
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Friday, April 8, 2011, 4:00 PM
> 
> http://wiki.freeswitch.org/wiki/ACL#Users
> 
> You can set variables directly in the user's directory entry.
> 
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
> 
> 
> 
> 
> On 2011-04-08, at 2:57 PM, FERNANDO VILLARROEL wrote:
> 
>> Hi Community.
>> 
>> How i can identifi inbound traffic authorizated on ACL with some variable like Accountcode.
>> 
>> For aoutbound traffic i use:
>> 
>> <action application="set" data="accountcode=foo"/>
>> 
>> My problem is for inbound traffic how i can identify accounts?
>> 
>> Regards.
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
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From steveayre at gmail.com  Mon Apr 11 10:54:08 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Apr 2011 07:54:08 +0100
Subject: [Freeswitch-users] Compiling mod_opal error
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDBC@cooper>
References: <178487.56361.qm@web34302.mail.mud.yahoo.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDBC@cooper>
Message-ID: <C3859669-D3B8-4622-BAE9-0C37FA5DC5FA@gmail.com>

You can also look at mod

Steve on iPhone

On 10 Apr 2011, at 17:19, Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> And also - bugs go into Jira - not the mailing list :)
> 
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r FERNANDO VILLARROEL [fvillarroel at yahoo.com]
> Skickat: den 10 april 2011 17:58
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] Compiling mod_opal error
> 
> Dear.
> 
> I am trying of use mod_opal, i am use the the follow how to:
> 
> http://wiki.freeswitch.org/wiki/FreeSwitch_H323
> 
> But when i try of install mod_opal i receive the follow error:
> 
> cc:/usr/src/freeswitch.trunk# make mod_opal
> 
> making all mod_opal
> Compiling /usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp...
> Compiling /usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp ...
> /usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp:67: error: invalid conversion from ???switch_call_cause_t (*)(switch_core_session_t*, switch_event_t*, switch_caller_profile_t*, switch_core_session_t**, switch_memory_pool_t**, switch_originate_flag_t)??? to ???switch_call_cause_t (*)(switch_core_session_t*, switch_event_t*, switch_caller_profile_t*, switch_core_session_t**, switch_memory_pool_t**, switch_originate_flag_t, switch_call_cause_t*)???
> make[4]: *** [mod_opal.lo] Error 1
> make[3]: *** [all] Error 1
> make[2]: *** [mod_opal-all] Error 1
> make[1]: *** [mod_opal] Error 2
> make: *** [mod_opal] Error 2
> 
> I am using FreesWITCH version:
> 
> FreeSWITCH Version 1.0.trunk (16526)
> 
> I need support H323 on my FS. Anyone could help me.
> 
> Regards.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:4da1d49e32761530917628!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Mon Apr 11 10:55:26 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Apr 2011 07:55:26 +0100
Subject: [Freeswitch-users] Compiling mod_opal error
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDBC@cooper>
References: <178487.56361.qm@web34302.mail.mud.yahoo.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDBC@cooper>
Message-ID: <3A9338FB-4135-41FF-8651-69208BF1D92C@gmail.com>

You can also look at mod_h323 for h323 support.

Mod_h323 uses the h323plus library
Mod_opal uses the opal library

If you have problems with one you might have more success with the other.

Steve on iPhone

On 10 Apr 2011, at 17:19, Peter Olsson <peter.olsson at visionutveckling.se> wrote:

> And also - bugs go into Jira - not the mailing list :)
> 
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r FERNANDO VILLARROEL [fvillarroel at yahoo.com]
> Skickat: den 10 april 2011 17:58
> Till: freeswitch-users at lists.freeswitch.org
> ?mne: [Freeswitch-users] Compiling mod_opal error
> 
> Dear.
> 
> I am trying of use mod_opal, i am use the the follow how to:
> 
> http://wiki.freeswitch.org/wiki/FreeSwitch_H323
> 
> But when i try of install mod_opal i receive the follow error:
> 
> cc:/usr/src/freeswitch.trunk# make mod_opal
> 
> making all mod_opal
> Compiling /usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp...
> Compiling /usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp ...
> /usr/src/freeswitch.trunk/src/mod/endpoints/mod_opal/mod_opal.cpp:67: error: invalid conversion from ???switch_call_cause_t (*)(switch_core_session_t*, switch_event_t*, switch_caller_profile_t*, switch_core_session_t**, switch_memory_pool_t**, switch_originate_flag_t)??? to ???switch_call_cause_t (*)(switch_core_session_t*, switch_event_t*, switch_caller_profile_t*, switch_core_session_t**, switch_memory_pool_t**, switch_originate_flag_t, switch_call_cause_t*)???
> make[4]: *** [mod_opal.lo] Error 1
> make[3]: *** [all] Error 1
> make[2]: *** [mod_opal-all] Error 1
> make[1]: *** [mod_opal] Error 2
> make: *** [mod_opal] Error 2
> 
> I am using FreesWITCH version:
> 
> FreeSWITCH Version 1.0.trunk (16526)
> 
> I need support H323 on my FS. Anyone could help me.
> 
> Regards.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:4da1d49e32761530917628!
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 Apr 11 11:31:17 2011
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 11 Apr 2011 09:31:17 +0200
Subject: [Freeswitch-users] Optimal configuration for compiling on
	64-bit platforms
In-Reply-To: <BANLkTikU9K5FCaCFJ9h=i3FqU1OccX7eyg@mail.gmail.com>
References: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>
	<BANLkTim9D6JdC=0Eg0TATP=iocGqumJ0vg@mail.gmail.com>
	<BANLkTimTvaxy8DqSnDqnLxdt6CCBntt8xA@mail.gmail.com>
	<BANLkTik1e-=HKvHwHdujGQXvZ177YfxkCA@mail.gmail.com>
	<BANLkTi=TwAUzx9m1Mea4iwobSKnoDyQSaQ@mail.gmail.com>
	<BANLkTi=MVnyPTMdgDwkTLxbTbann6N959w@mail.gmail.com>
	<BANLkTikU9K5FCaCFJ9h=i3FqU1OccX7eyg@mail.gmail.com>
Message-ID: <BANLkTi=PghrX9QfOzdk604a-bZX0m30JQg@mail.gmail.com>

Sorry, I had not understood you were trying to use sparc, I was
writing as you were using standard Intel.
No idea how to help with your experimentations.
-giovanni


On 4/11/11, A E [Gmail] <all.eforums at gmail.com> wrote:
> On Sat, Apr 9, 2011 at 7:40 PM, Giovanni Maruzzelli
> <gmaruzz at gmail.com>wrote:
>
>> Compiler options have nothing to do with libraries incompatibilities.
>> Why don't you use the standard libraries given by your distro?
>> Never heard someone had to compile ssl libs for FreeSWITCH.
>> Also never heard someone used any compiler options on standard machines.
>> FS is targeted toward 64bit OSes, most people use CentOS, Ubuntu and
>> Debian. CentOS is by far the most used and the reference platform. On
>> 64bit.
>> That said, if you're trying to optimize it for a PIII, an EPIA 5000 or
>> an ARM, or whatever is not a standard server class machine, yes you
>> better know how to optimize :).
>>
>>
> Well 'you told me so' Giovanni ;)
>
> It's finally built! I deleted the entire source directory, did a brand new
> git clone, bootstrap and the whole process with NO switches, flags or
> options given to 'configure'
>
> But the 'configure' etc did NOT take care of the fact that it's a 64-bit
> platform, proven by the following:
>
> $ file ./.libs/freeswitch
> ./.libs/freeswitch: ELF 32-bit MSB executable, SPARC32PLUS, V8+ Required,
> version 1 (SYSV), dynamically linked (uses shared libs), for GNU/Linux
> 2.6.18, not stripped
>
> Now I have heard from people in the debian-sparc forum that in the case of
> debian port, sparc32 is in fact faster than sparc64 and as a result the
> current userland is actually 32-bit. If that applies to FS built as 32-bit
> in this environment to be faster or comparable in performance to 64-bit,
> then I'm golden. Else, I guess I have no other choice as I can't figure out
> how to build it and get past the ssl problem I was having with all those m64
> flags!
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From all.eforums at gmail.com  Mon Apr 11 11:58:26 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 11 Apr 2011 03:58:26 -0400
Subject: [Freeswitch-users] Optimal configuration for compiling on
	64-bit platforms
In-Reply-To: <BANLkTi=PghrX9QfOzdk604a-bZX0m30JQg@mail.gmail.com>
References: <BANLkTikfmXMUXwkHGzTtsC1roDqX6Xo=fw@mail.gmail.com>
	<BANLkTim9D6JdC=0Eg0TATP=iocGqumJ0vg@mail.gmail.com>
	<BANLkTimTvaxy8DqSnDqnLxdt6CCBntt8xA@mail.gmail.com>
	<BANLkTik1e-=HKvHwHdujGQXvZ177YfxkCA@mail.gmail.com>
	<BANLkTi=TwAUzx9m1Mea4iwobSKnoDyQSaQ@mail.gmail.com>
	<BANLkTi=MVnyPTMdgDwkTLxbTbann6N959w@mail.gmail.com>
	<BANLkTikU9K5FCaCFJ9h=i3FqU1OccX7eyg@mail.gmail.com>
	<BANLkTi=PghrX9QfOzdk604a-bZX0m30JQg@mail.gmail.com>
Message-ID: <BANLkTim2XvwyqUSNDyYf0D0mH7JspVmcyA@mail.gmail.com>

On Mon, Apr 11, 2011 at 3:31 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Sorry, I had not understood you were trying to use sparc, I was
> writing as you were using standard Intel.
> No idea how to help with your experimentations.
> -giovanni
>
>
No worries :) Well you were still right. Now I'm trying to fight through the
core dump! I only wish this was "experimentation". I have half a rack full
of SunFire V100 and V240 machines and I have to make this work on them as
all intel based machines are too weak and the relatively powerful ones are
already running critical windows software.
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From rebel.pappas at gmail.com  Mon Apr 11 12:55:17 2011
From: rebel.pappas at gmail.com (alex pappas)
Date: Mon, 11 Apr 2011 11:55:17 +0300
Subject: [Freeswitch-users] Calling card test
Message-ID: <BANLkTikMO8Jg_Yx9HtTxC7OXWFPjbJh4Lg@mail.gmail.com>

Dear all,

I'm very new to Freeswitch and I'm trying to find the best way to implement
an application which will handle calling cards.

The scenario is simple. A user makes a call, the call get answered and the
user gives the pin number and if is correct  then the user can make a call.

My question is how is the best way to implement this?

1. Through a script which will be called from the dialplan
2. From Inbound ESL
3. From Outbound ESL

Thank you in advance

Alex
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From marcdecorny at gmail.com  Mon Apr 11 13:44:29 2011
From: marcdecorny at gmail.com (Marc de Corny)
Date: Mon, 11 Apr 2011 10:44:29 +0100
Subject: [Freeswitch-users] Mod_fifo outbound strategy enterprise
In-Reply-To: <AANLkTim4XgQXUYRqMZstsEMd6oimbfU6iHG9Y0MRSkvU@mail.gmail.com>
References: <AANLkTiku=gKQAVSy2of8kSZoF+b+zXM5u_yHS45ojUd5@mail.gmail.com>
	<AANLkTim4XgQXUYRqMZstsEMd6oimbfU6iHG9Y0MRSkvU@mail.gmail.com>
Message-ID: <BANLkTin9SJ7UzVgY2XF7Uaiq4=gaFWbLcg@mail.gmail.com>

Hi Anthony

Thanks for your response, sorry for the delay, but for some reason I did not
see it until now.

To be honest I think the whole point of fifo is to keep it simple and let
mod_callcenter do the more advanced stuff and that it specifically why I
went for it after having read both wiki pages quite thoroughly.

But there is still something I don't understand on how the default behaviour
ring-all with CLI works. According the logic you have described below. it
looks as if the call gets only presented to one agent at a time. however
when testing it, it looks as is the calls gets sent out with the CLI to all
agents who are available. Is that what you'd expect?

Thanks
Marc



On Mon, Mar 21, 2011 at 10:26 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The ringall strategy is based on giving every queue a fair chance
> based on its configured importance.
>
> The following happens nonstop:
>
> *) Loop all the active call queues containing waiting callers not
> already receiving a call sorted on queue priority:
>       *) Select agents from a db (omitting any who are on a call or
> who are in wrap-up) sorted by
>            least amount of consecutive missed calls then least amount
> of answered calls.
> *) Place an outbound call to this agent with the caller data of the
> waiting customer
>
> This distributes the opportunities to have your call answered across
> all queues and chooses the most-likely-to-answer-the-phone agent.
>
>
> The shortcoming of this strategy is simply the fact that you must
> pre-allocate agents for each caller to ensure that you can supply the
> caller's caller id to the agent when you call them.
>
> The enterprise one is accelerated by figuring out how many callers are
> waiting and calling that many agents at once with no caller id info
> and inserting them into the queue to service the next customer in
> line. Since there is no need to pre-match the caller and agent the
> order of what agent and caller are paired is moot.  This was the
> original behavior but the importance of caller id seems to prevail so
> we made the other method the default.
>
> It would be possible to code in other strategies but to maintain the
> simplicity of fifo I did not really bother with any more.
>
>
>
>
>
>
>
>
> On Mon, Mar 21, 2011 at 12:59 PM, Marc de Corny <marcdecorny at gmail.com>
> wrote:
> > Hi all,
> > I am getting the hang of the fifo now and it is proving very stable which
> is
> > what we all need.
> > Got one question however regarding mod_fifo and the outbound strategies.
> > From looking at the outbput of the fifo list commands I can see that that
> > parameter is always set to "ringall". which is fine, but I can also see
> many
> > stats on that number of calls and the last call that the members have
> taken.
> > Is there also a way of sending the calls to the longest idle or something
> of
> > that nature? Is that done by setting the outbound_strategy to
> "enterprise"
> > or is there another way to achieve that.
> > Is there also a command to change that value without editing the XML as I
> am
> > trying to make everything dynamic.
> > any input is very much appreciated.
> > thanks
> > Marc
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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
> googletalk:conf+888 at conference.freeswitch.org
> pstn:+19193869900
>
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From boris at tagnet.ru  Mon Apr 11 16:35:59 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Mon, 11 Apr 2011 18:35:59 +0600
Subject: [Freeswitch-users] Clarify about gateways please
Message-ID: <4DA2F5AF.8030709@tagnet.ru>

Hello!

     Reading documentation about gateways and misunderstanding. Please 
clarify: when there is inbound call from a gateway the 
destination_number is set to:
||

<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
<!--<param name="extension" value="cluecon"/>-->

But in this case how may I know what destination_number gateway is calling?

-- 
Regards,
Boris


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From rhuddleston at gmail.com  Mon Apr 11 17:06:06 2011
From: rhuddleston at gmail.com (rhuddleston at gmail.com)
Date: Mon, 11 Apr 2011 09:06:06 -0400
Subject: [Freeswitch-users] Calling card test
In-Reply-To: <BANLkTikMO8Jg_Yx9HtTxC7OXWFPjbJh4Lg@mail.gmail.com>
References: <BANLkTikMO8Jg_Yx9HtTxC7OXWFPjbJh4Lg@mail.gmail.com>
Message-ID: <8436C9B0-A124-4492-ABEC-D83FE1CCAB62@gmail.com>

I have a custom written LUA script which utilizes mod_lcr and mod_nibblebill. I'm also using xml cdr support.

On Apr 11, 2011, at 4:55 AM, alex pappas <rebel.pappas at gmail.com> wrote:

> Dear all,
> 
> I'm very new to Freeswitch and I'm trying to find the best way to implement an application which will handle calling cards.
> 
> The scenario is simple. A user makes a call, the call get answered and the user gives the pin number and if is correct  then the user can make a call.
> 
> My question is how is the best way to implement this?
> 
> 1. Through a script which will be called from the dialplan
> 2. From Inbound ESL
> 3. From Outbound ESL
> 
> Thank you in advance
> 
> Alex
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From avi at avimarcus.net  Mon Apr 11 17:16:00 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 11 Apr 2011 16:16:00 +0300
Subject: [Freeswitch-users] Calling card test
In-Reply-To: <8436C9B0-A124-4492-ABEC-D83FE1CCAB62@gmail.com>
References: <BANLkTikMO8Jg_Yx9HtTxC7OXWFPjbJh4Lg@mail.gmail.com>
	<8436C9B0-A124-4492-ABEC-D83FE1CCAB62@gmail.com>
Message-ID: <BANLkTimiYN7ta3-T4q_ZTx8Y13b=awDjig@mail.gmail.com>

You can do everything from the dial plan, with a few queries and/or mod_lcr.
You'll have to adjust the account balance post-call, though, via the cdr.
For making sure it is prepaid, you can use api_sched_hangup/transfer and
then you don't need mod_nibblebill.

-Avi

On Mon, Apr 11, 2011 at 4:06 PM, <rhuddleston at gmail.com> wrote:

> I have a custom written LUA script which utilizes mod_lcr and
> mod_nibblebill. I'm also using xml cdr support.
>
> On Apr 11, 2011, at 4:55 AM, alex pappas <rebel.pappas at gmail.com> wrote:
>
> > Dear all,
> >
> > I'm very new to Freeswitch and I'm trying to find the best way to
> implement an application which will handle calling cards.
> >
> > The scenario is simple. A user makes a call, the call get answered and
> the user gives the pin number and if is correct  then the user can make a
> call.
> >
> > My question is how is the best way to implement this?
> >
> > 1. Through a script which will be called from the dialplan
> > 2. From Inbound ESL
> > 3. From Outbound ESL
> >
> > Thank you in advance
> >
> > Alex
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Mon Apr 11 17:41:08 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Apr 2011 14:41:08 +0100
Subject: [Freeswitch-users] Clarify about gateways please
In-Reply-To: <4DA2F5AF.8030709@tagnet.ru>
References: <4DA2F5AF.8030709@tagnet.ru>
Message-ID: <BANLkTinQg9wuf94fDJOmeozHRk6n-1hG+Q@mail.gmail.com>

Try the auto_to_user value.

<!--/// extension for inbound calls: *optional* Same as username, if blank.
To use what's in ${sip_to_user}, set it to the value "auto_to_user" ///-->
<param name="extension" value=""/>

Source: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files

-Steve


On 11 April 2011 13:35, Boris Kovalenko <boris at tagnet.ru> wrote:

>  Hello!
>
>     Reading documentation about gateways and misunderstanding. Please
> clarify: when there is inbound call from a gateway the destination_number is
> set to:
>
> <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
> <!--<param name="extension" value="cluecon"/>-->
>
> But in this case how may I know what destination_number gateway is calling?
>
> --
> Regards,
> Boris
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Mon Apr 11 17:42:54 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 11 Apr 2011 14:42:54 +0100
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <F01DDEF9-D207-4838-85E6-3C238F79373D@gmail.com>
References: <453850.62406.qm@web34303.mail.mud.yahoo.com>
	<F01DDEF9-D207-4838-85E6-3C238F79373D@gmail.com>
Message-ID: <BANLkTinQqgqEGSR+Q_-aAPnMqeSHbCGO4Q@mail.gmail.com>

Just checked and this is indeed the correct syntax. It's documented on the
Wiki:
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Gateway
Scroll down to the example just above the Settings (
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Settings) header.

Does that help?

-Steve



On 11 April 2011 07:50, Steven Ayre <steveayre at gmail.com> wrote:

> You can set channel variables in the <gateway> section, and you can specify
> the direction too so that variables get set either on inbound, outbound or
> both.
>
> I *think* the syntax is:
>
> <gateway>
>    ...
>    <variables>
>        <variable name="accountcode" value="ABC" direction="inbound"/>
>        ...
>    </variables>
> </gateway>
>
> Steve on iPhone
>
> On 8 Apr 2011, at 22:05, FERNANDO VILLARROEL <fvillarroel at yahoo.com>
> wrote:
>
> Dear Mathieu.
>
> My user are not in ditrectory.
>
> My users are gateways authenticated with ACL.
>
> How i can use some variable for identifi like accountcode for inbound
> traffic from this gateways???
>
> Regards
>
> --- On *Fri, 4/8/11, Mathieu Rene <mrene_lists at avgs.ca>* wrote:
>
>
> From: Mathieu Rene <mrene_lists at avgs.ca>
> Subject: Re: [Freeswitch-users] Account ACL
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Friday, April 8, 2011, 4:00 PM
>
> <http://wiki.freeswitch.org/wiki/ACL#Users>
> http://wiki.freeswitch.org/wiki/ACL#Users
>
> You can set variables directly in the user's directory entry.
> <http://wiki.freeswitch.org/wiki/ACL#Users>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> <http://mc/compose?to=mrene at avgs.ca>mrene at avgs.ca
>
>
>
>
> On 2011-04-08, at 2:57 PM, FERNANDO VILLARROEL wrote:
>
> Hi Community.
>
> How i can identifi inbound traffic authorizated on ACL with some variable
> like Accountcode.
>
> For aoutbound traffic i use:
>
> <action application="set" data="accountcode=foo"/>
>
> My problem is for inbound traffic how i can identify accounts?
>
> Regards.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> <http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> FreeSWITCH-users at 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
> <http://mc/compose?to=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
>
>
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From rebel.pappas at gmail.com  Mon Apr 11 17:50:54 2011
From: rebel.pappas at gmail.com (alex pappas)
Date: Mon, 11 Apr 2011 16:50:54 +0300
Subject: [Freeswitch-users] Calling card test
In-Reply-To: <BANLkTimiYN7ta3-T4q_ZTx8Y13b=awDjig@mail.gmail.com>
References: <BANLkTikMO8Jg_Yx9HtTxC7OXWFPjbJh4Lg@mail.gmail.com>
	<8436C9B0-A124-4492-ABEC-D83FE1CCAB62@gmail.com>
	<BANLkTimiYN7ta3-T4q_ZTx8Y13b=awDjig@mail.gmail.com>
Message-ID: <BANLkTinr+P=1H9WW5LpmQQFRkMkrYf-LWg@mail.gmail.com>

Hi,
My approach is that all the logic would be in the application and in the
backend(Database).

What I want to understand is which is the best way to run an application
with Freeswitch concerning the performance ofc.
If I was doing this in Asterisk I would try with AGI for example.

Thanks
\Alx

In my understanding inside the application will seat the business logic of
the prepaid. In every step I can play dynamically sound files and in every
step I would save a custom CDR to my Database


On Mon, Apr 11, 2011 at 4:16 PM, Avi Marcus <avi at avimarcus.net> wrote:

> You can do everything from the dial plan, with a few queries and/or
> mod_lcr. You'll have to adjust the account balance post-call, though, via
> the cdr.
> For making sure it is prepaid, you can use api_sched_hangup/transfer and
> then you don't need mod_nibblebill.
>
> -Avi
>
>
> On Mon, Apr 11, 2011 at 4:06 PM, <rhuddleston at gmail.com> wrote:
>
>> I have a custom written LUA script which utilizes mod_lcr and
>> mod_nibblebill. I'm also using xml cdr support.
>>
>> On Apr 11, 2011, at 4:55 AM, alex pappas <rebel.pappas at gmail.com> wrote:
>>
>> > Dear all,
>> >
>> > I'm very new to Freeswitch and I'm trying to find the best way to
>> implement an application which will handle calling cards.
>> >
>> > The scenario is simple. A user makes a call, the call get answered and
>> the user gives the pin number and if is correct  then the user can make a
>> call.
>> >
>> > My question is how is the best way to implement this?
>> >
>> > 1. Through a script which will be called from the dialplan
>> > 2. From Inbound ESL
>> > 3. From Outbound ESL
>> >
>> > Thank you in advance
>> >
>> > Alex
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rhuddleston at gmail.com  Mon Apr 11 17:59:43 2011
From: rhuddleston at gmail.com (rhuddleston at gmail.com)
Date: Mon, 11 Apr 2011 09:59:43 -0400
Subject: [Freeswitch-users] Calling card test
In-Reply-To: <BANLkTinr+P=1H9WW5LpmQQFRkMkrYf-LWg@mail.gmail.com>
References: <BANLkTikMO8Jg_Yx9HtTxC7OXWFPjbJh4Lg@mail.gmail.com>
	<8436C9B0-A124-4492-ABEC-D83FE1CCAB62@gmail.com>
	<BANLkTimiYN7ta3-T4q_ZTx8Y13b=awDjig@mail.gmail.com>
	<BANLkTinr+P=1H9WW5LpmQQFRkMkrYf-LWg@mail.gmail.com>
Message-ID: <B6DE24D8-DB8E-487B-AFEF-405B56F509B7@gmail.com>

Whats wrong with nibblebill? I use lua script because i have business logic that has to be checked.
I like how nibblebill keeps the balance updated on the fly

On Apr 11, 2011, at 9:50 AM, alex pappas <rebel.pappas at gmail.com> wrote:

> Hi,
> My approach is that all the logic would be in the application and in the backend(Database).
> 
> What I want to understand is which is the best way to run an application with Freeswitch concerning the performance ofc.
> If I was doing this in Asterisk I would try with AGI for example.
> 
> Thanks
> \Alx
> 
> In my understanding inside the application will seat the business logic of the prepaid. In every step I can play dynamically sound files and in every step I would save a custom CDR to my Database 
> 
> 
> On Mon, Apr 11, 2011 at 4:16 PM, Avi Marcus <avi at avimarcus.net> wrote:
> You can do everything from the dial plan, with a few queries and/or mod_lcr. You'll have to adjust the account balance post-call, though, via the cdr.
> For making sure it is prepaid, you can use api_sched_hangup/transfer and then you don't need mod_nibblebill.
> 
> -Avi
> 
> 
> On Mon, Apr 11, 2011 at 4:06 PM,  <rhuddleston at gmail.com> wrote:
> I have a custom written LUA script which utilizes mod_lcr and mod_nibblebill. I'm also using xml cdr support.
> 
> On Apr 11, 2011, at 4:55 AM, alex pappas <rebel.pappas at gmail.com> wrote:
> 
> > Dear all,
> >
> > I'm very new to Freeswitch and I'm trying to find the best way to implement an application which will handle calling cards.
> >
> > The scenario is simple. A user makes a call, the call get answered and the user gives the pin number and if is correct  then the user can make a call.
> >
> > My question is how is the best way to implement this?
> >
> > 1. Through a script which will be called from the dialplan
> > 2. From Inbound ESL
> > 3. From Outbound ESL
> >
> > Thank you in advance
> >
> > Alex
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From rebel.pappas at gmail.com  Mon Apr 11 18:10:07 2011
From: rebel.pappas at gmail.com (alex pappas)
Date: Mon, 11 Apr 2011 17:10:07 +0300
Subject: [Freeswitch-users] Calling card test
In-Reply-To: <B6DE24D8-DB8E-487B-AFEF-405B56F509B7@gmail.com>
References: <BANLkTikMO8Jg_Yx9HtTxC7OXWFPjbJh4Lg@mail.gmail.com>
	<8436C9B0-A124-4492-ABEC-D83FE1CCAB62@gmail.com>
	<BANLkTimiYN7ta3-T4q_ZTx8Y13b=awDjig@mail.gmail.com>
	<BANLkTinr+P=1H9WW5LpmQQFRkMkrYf-LWg@mail.gmail.com>
	<B6DE24D8-DB8E-487B-AFEF-405B56F509B7@gmail.com>
Message-ID: <BANLkTi=FPfiixcyvmn94Oz0PU=SW5NsBsg@mail.gmail.com>

Nothing wrong with nibblebill. I'm trying to simulate an existing system and
I need to start everything from scratch. That's why I'm asking about how I
can have performance with Freeswitch.

\Alx



On Mon, Apr 11, 2011 at 4:59 PM, <rhuddleston at gmail.com> wrote:

> Whats wrong with nibblebill? I use lua script because i have business logic
> that has to be checked.
> I like how nibblebill keeps the balance updated on the fly
>
>
> On Apr 11, 2011, at 9:50 AM, alex pappas <rebel.pappas at gmail.com> wrote:
>
> Hi,
> My approach is that all the logic would be in the application and in the
> backend(Database).
>
> What I want to understand is which is the best way to run an application
> with Freeswitch concerning the performance ofc.
> If I was doing this in Asterisk I would try with AGI for example.
>
> Thanks
> \Alx
>
> In my understanding inside the application will seat the business logic of
> the prepaid. In every step I can play dynamically sound files and in every
> step I would save a custom CDR to my Database
>
>
> On Mon, Apr 11, 2011 at 4:16 PM, Avi Marcus < <avi at avimarcus.net>
> avi at avimarcus.net> wrote:
>
>> You can do everything from the dial plan, with a few queries and/or
>> mod_lcr. You'll have to adjust the account balance post-call, though, via
>> the cdr.
>> For making sure it is prepaid, you can use api_sched_hangup/transfer and
>> then you don't need mod_nibblebill.
>>
>> -Avi
>>
>>
>> On Mon, Apr 11, 2011 at 4:06 PM, < <rhuddleston at gmail.com>
>> rhuddleston at gmail.com> wrote:
>>
>>> I have a custom written LUA script which utilizes mod_lcr and
>>> mod_nibblebill. I'm also using xml cdr support.
>>>
>>> On Apr 11, 2011, at 4:55 AM, alex pappas < <rebel.pappas at gmail.com>
>>> rebel.pappas at gmail.com> wrote:
>>>
>>> > Dear all,
>>> >
>>> > I'm very new to Freeswitch and I'm trying to find the best way to
>>> implement an application which will handle calling cards.
>>> >
>>> > The scenario is simple. A user makes a call, the call get answered and
>>> the user gives the pin number and if is correct  then the user can make a
>>> call.
>>> >
>>> > My question is how is the best way to implement this?
>>> >
>>> > 1. Through a script which will be called from the dialplan
>>> > 2. From Inbound ESL
>>> > 3. From Outbound ESL
>>> >
>>> > Thank you in advance
>>> >
>>> > Alex
>>> >
>>> >
>>> > _______________________________________________
>>> > 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
>
>
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From anthony.minessale at gmail.com  Mon Apr 11 18:45:07 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Apr 2011 09:45:07 -0500
Subject: [Freeswitch-users] Core Dump after compiling on Debian for Sparc
In-Reply-To: <BANLkTin3moUaFNVY-wtFOxX5TW1cKWGpMA@mail.gmail.com>
References: <BANLkTin3moUaFNVY-wtFOxX5TW1cKWGpMA@mail.gmail.com>
Message-ID: <BANLkTinY3sCWAHuyfeGQsuTJA7=gwYV=ZA@mail.gmail.com>

We have limited support of sparc because we don't have any machines to work on.
You should consider finalizing your support contract and maybe let us
find you a consultant to help you with that.


On Sun, Apr 10, 2011 at 11:40 PM, A E [Gmail] <all.eforums at gmail.com> wrote:
> Hi Guys,
> So I finally got Freeswitch to compile on my Sparc machine running 64-bit
> debian. As was mentioned earlier, it was compiled with NO CXX flags or other
> options provided to the configure script. However, when I start freeswitch,
> it core dumps.
> All relevant information is available
> on:?http://pastebin.freeswitch.org/16064
> Can someone please look at it and let me know if this is just me or this
> hints at a bug that no one else ran into. I'm running the latest from Git.
> Thanks so much
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From boris at tagnet.ru  Mon Apr 11 19:26:15 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Mon, 11 Apr 2011 21:26:15 +0600
Subject: [Freeswitch-users] incoming calls from gateways without auth
Message-ID: <4DA31D97.6050803@tagnet.ru>

Hello!

     I have a profile where auth-calls param is true. Is there a way to 
have incoming calls from gateways without auth in this profile? 
Something like cidr="xxx" for directory entry? Or the only way is to 
write ACL with gateways' ips and use it with apply-inbound-acl?

-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From msc at freeswitch.org  Mon Apr 11 20:21:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Apr 2011 09:21:52 -0700
Subject: [Freeswitch-users] Calling card test
In-Reply-To: <BANLkTi=FPfiixcyvmn94Oz0PU=SW5NsBsg@mail.gmail.com>
References: <BANLkTikMO8Jg_Yx9HtTxC7OXWFPjbJh4Lg@mail.gmail.com>
	<8436C9B0-A124-4492-ABEC-D83FE1CCAB62@gmail.com>
	<BANLkTimiYN7ta3-T4q_ZTx8Y13b=awDjig@mail.gmail.com>
	<BANLkTinr+P=1H9WW5LpmQQFRkMkrYf-LWg@mail.gmail.com>
	<B6DE24D8-DB8E-487B-AFEF-405B56F509B7@gmail.com>
	<BANLkTi=FPfiixcyvmn94Oz0PU=SW5NsBsg@mail.gmail.com>
Message-ID: <BANLkTinwW15pLdjACgCWc+wnxV+Zr9eqag@mail.gmail.com>

On Mon, Apr 11, 2011 at 7:10 AM, alex pappas <rebel.pappas at gmail.com> wrote:

> Nothing wrong with nibblebill. I'm trying to simulate an existing system
> and I need to start everything from scratch. That's why I'm asking about how
> I can have performance with Freeswitch.
>

Scripting right from the dialplan has a lower barrier to entry but will not
scale as well as using outbound event socket. If I were in your shoes I
would roll up my sleeves and learn ESL. Pick your favorite language. ESL has
bindings for:
C/C++
Perl
PHP
Python
Ruby
TCL

I highly recommend getting the FreeSWITCH book and reading chapter 9. Of
course, chapters 1 through 6 are also important for getting a foundation for
using FreeSWITCH, but chapter 9 has a lot of solid information about using
the event socket and ESL.

Note: you asked about using inbound vs. outbound event socket. If I
understand your situation correctly you need to handle the case where an
outside party calls in to your FS server. This means you need outbound event
socket. (See the 'socket' dp tool on the wiki.)

Enjoy!
-MC
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From m.sobkow at marketelsystems.com  Mon Apr 11 20:38:03 2011
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 11 Apr 2011 10:38:03 -0600
Subject: [Freeswitch-users] Can someone point me to examples of how to
 program message-based calls?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDB9@cooper>
References: <32121300.77951302392962541.JavaMail.root@julie.marketel>
	<549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDB9@cooper>
Message-ID: <4DA32E6B.2020700@marketelsystems.com>

Possible, but the code had been working for over six months, then 
stopped working when we'd updated Freeswitch, so I thought things had 
(finally!) switched over to a pure event-based implementation.

I've got most of the code in place to do event-based processing instead, 
just getting our users to test it this morning.  If it works (catching 
EVENT_HANGUP_COMPLETE), I'll add in the extra glue logic to classify the 
reason for the hangup and update our call stats accordingly.

On 10/04/2011 1:11 AM, Peter Olsson wrote:
> Are you sure it changed? Can't find anything about this in the git commit log recently.
>
> I've never used Erlang myself, but this sounds like you originate a call, without the "ignore_early_media=true" flag. If this is not specified the originate command will return immediately when early media is detected, if you specify this channel variable it will wait for the call to be answered.
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mark Sobkow [m.sobkow at marketelsystems.com]
> Skickat: den 10 april 2011 01:49
> Till: freeswitch-users
> ?mne: [Freeswitch-users] Can someone point me to examples of how to program message-based calls?
>
> Recently the API behaviour changed for Erlang such that when you initiate a call, the routine immediately returns with a UUID of the launched call, rather than waiting for the call to be answered.  I've been able to program for this behaviour in the call-connected case, but I'm at a loss as to what to do for detecting calls that go unanswered, time out, or which cannot be placed for various technical reasons.
>
> As a result, my call queue is filling up and choking -- I only launch so many calls at a time, and only the connected calls are getting properly processed.  Unanswered and timed out calls are getting "stuck" because I don't know what events to catch and how to evaluate them.
>
> The old behaviour wasn't event-based and thereby easier to program -- but I can definitely see the advantages of shifting to a pure event-driven model.  I just need to learn how to _use_ it.
>
> Thanks for any assistance you can provide.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:4da0f41032761398017734!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
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

Visit our Blog for industry related information.
http://marketel-systems.blogspot.com/



From m.sobkow at marketelsystems.com  Mon Apr 11 20:39:06 2011
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Mon, 11 Apr 2011 10:39:06 -0600
Subject: [Freeswitch-users] Can someone point me to examples of how to
 program message-based calls?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDB9@cooper>
References: <32121300.77951302392962541.JavaMail.root@julie.marketel>
	<549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDB9@cooper>
Message-ID: <4DA32EAA.8070104@marketelsystems.com>

Sorry, that was CHANNEL_HANGUP_COMPLETE

-- 
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

Visit our Blog for industry related information.
http://marketel-systems.blogspot.com/



From ovvenkatesan at gmail.com  Mon Apr 11 21:32:40 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Mon, 11 Apr 2011 23:02:40 +0530
Subject: [Freeswitch-users] Freeswitch Server Down
Message-ID: <BANLkTimerjRCutH8LFaRjvCfSju6byO1ow@mail.gmail.com>

Hi to all,

Today, All my IVR are stopped working.
When I check the freeSwitch it was down.
I dont know the reason why Its Down.
How I can find the reason,  why my freeSwitch went down?


Regards,
Venkat.
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From brian at freeswitch.org  Mon Apr 11 21:41:35 2011
From: brian at freeswitch.org (Brian West)
Date: Mon, 11 Apr 2011 12:41:35 -0500
Subject: [Freeswitch-users] FreeSWITCH sometimes binds to loopback
	interface during boot
In-Reply-To: <intmi8$e9b$1@dough.gmane.org>
References: <intmi8$e9b$1@dough.gmane.org>
Message-ID: <CB1CEFA1-0F43-451C-8E6F-A5AC5EC95728@freeswitch.org>

have you tried "sofia profile external restart"

/b

On Apr 10, 2011, at 8:46 PM, Jason White wrote:

>  I have to restart
> FreeSWITCH.



From msc at freeswitch.org  Mon Apr 11 21:49:36 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Apr 2011 10:49:36 -0700
Subject: [Freeswitch-users] Freeswitch Server Down
In-Reply-To: <BANLkTimerjRCutH8LFaRjvCfSju6byO1ow@mail.gmail.com>
References: <BANLkTimerjRCutH8LFaRjvCfSju6byO1ow@mail.gmail.com>
Message-ID: <BANLkTikLGEqSnL7yYnvVmBE6AvYgCBTXmg@mail.gmail.com>

On Mon, Apr 11, 2011 at 10:32 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:

> Hi to all,
>
> Today, All my IVR are stopped working.
> When I check the freeSwitch it was down.
> I dont know the reason why Its Down.
> How I can find the reason,  why my freeSwitch went down?
>

You'll need to check logs to track down what happened. I know that sometimes
you'll see lots of log lines in freeswitch.log and then all of a sudden
nothing, so that will help you pinpoint when things went wrong. Possibly
you'll see some log lines with errors or warnings. Or you might see that the
"fsctl shutdown" command was sent to it. (Then you'll need to go hunt down
whoever did that. :)

-MC
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From nick.rosier at gmail.com  Mon Apr 11 23:53:17 2011
From: nick.rosier at gmail.com (Nick Rosier)
Date: Mon, 11 Apr 2011 21:53:17 +0200
Subject: [Freeswitch-users] Gateway with dynamic IP address
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58C43A41E5@cooper>
References: <C9C8E1CC-4621-44FF-8D19-D8DB35D41530@gmail.com>
	<AANLkTinQNJpc0UP4hSjnSaa5zqcqgVoLhV0ScrPDvwNN@mail.gmail.com>
	<828493E7-A5E7-4896-844F-271AB72AD38B@gmail.com>
	<AANLkTinZWWijJBBWr0ArbtSGjHuv1cpHDDxKiZqj4Agg@mail.gmail.com>
	<BANLkTik1B7XhyCnDe2zs3Pqob4MD4evaPQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58C43A41E5@cooper>
Message-ID: <BANLkTi=4wX_cc9hPU0S1-O0UT0RE_Tom0Q@mail.gmail.com>

On 5 April 2011 22:45, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> What you wan't to do is to add a user. Then you dial this user, which by then is registered in FreeSWITCH, and it will find the path.
>
> So no gateway in this case, it's when you want to register to an external server, a user is when someone registers to you, and you wan't to be able to dial outside through this.
>
> /Peter

Can someone help me with the URI. It's driving me crazy.
This is what I've got but it's not working:

<action application="bridge"
data="sofia/sipinterface_1/trunk1 at pbx.domain.com/$1"/>

What am I doing wrong?

N.


From all.eforums at gmail.com  Tue Apr 12 00:35:01 2011
From: all.eforums at gmail.com (A E [Gmail])
Date: Mon, 11 Apr 2011 16:35:01 -0400
Subject: [Freeswitch-users] Core Dump after compiling on Debian for Sparc
In-Reply-To: <BANLkTinY3sCWAHuyfeGQsuTJA7=gwYV=ZA@mail.gmail.com>
References: <BANLkTin3moUaFNVY-wtFOxX5TW1cKWGpMA@mail.gmail.com>
	<BANLkTinY3sCWAHuyfeGQsuTJA7=gwYV=ZA@mail.gmail.com>
Message-ID: <BANLkTinDMzuf9TPXpJpQew2un4e5cbuNsw@mail.gmail.com>

On Mon, Apr 11, 2011 at 10:45 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> We have limited support of sparc because we don't have any machines to work
> on.
> You should consider finalizing your support contract and maybe let us
> find you a consultant to help you with that.
>
>
Hi Anthony,

*Sigh* I was afraid you would say that. Honestly, there's nothing I'd like
more than to _not_ have to spend days doing all these techno-acrobatics just
trying to compile and/or run the software that's going to be the most
important piece in building our service. Unfortunately, we're not a funded
company. Which is why we're stuck with the old Sun equipment that we've had,
relics of our old service that didn't take off. They're still pretty good
servers and they hardly got used even if they're 6 years old running on
64-bit 600Mhz UltraSparcII :(

So, we need to find a happy medium here, ideally. I don't mind talking about
the support contract however and get a sense of what it will actually cost
before I just reject the offer without even knowing the cost.

An alternative/counter-offer could be , how about I donate one of these v100
Sparc servers to the project to build/test against since you said that the
support for Sparc is limited? :)
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From anthony.minessale at gmail.com  Tue Apr 12 00:38:20 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 11 Apr 2011 15:38:20 -0500
Subject: [Freeswitch-users] Core Dump after compiling on Debian for Sparc
In-Reply-To: <BANLkTinDMzuf9TPXpJpQew2un4e5cbuNsw@mail.gmail.com>
References: <BANLkTin3moUaFNVY-wtFOxX5TW1cKWGpMA@mail.gmail.com>
	<BANLkTinY3sCWAHuyfeGQsuTJA7=gwYV=ZA@mail.gmail.com>
	<BANLkTinDMzuf9TPXpJpQew2un4e5cbuNsw@mail.gmail.com>
Message-ID: <BANLkTinvzXTmsiDohVW7t-xsWNx2Ht-eXw@mail.gmail.com>

please contact me offlist.

On Mon, Apr 11, 2011 at 3:35 PM, A E [Gmail] <all.eforums at gmail.com> wrote:
> On Mon, Apr 11, 2011 at 10:45 AM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> We have limited support of sparc because we don't have any machines to
>> work on.
>> You should consider finalizing your support contract and maybe let us
>> find you a consultant to help you with that.
>>
>
> Hi Anthony,
> *Sigh* I was afraid you would say that. Honestly, there's nothing I'd like
> more than to _not_ have to spend days doing all these techno-acrobatics just
> trying to compile and/or run the software that's going to be the most
> important piece in building our service. Unfortunately, we're not a funded
> company. Which is why we're stuck with the old Sun equipment that we've had,
> relics of our old service that didn't take off. They're still pretty good
> servers and they hardly got used even if they're 6 years old running on
> 64-bit 600Mhz UltraSparcII :(
> So, we need to find a happy medium here, ideally. I don't mind talking about
> the support contract however and get a sense of what it will actually cost
> before I just reject the offer without even knowing the cost.
> An alternative/counter-offer could be , how about I donate one of these v100
> Sparc servers to the project to build/test against?since you said that the
> support for Sparc is limited? :)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900


From fvillarroel at yahoo.com  Tue Apr 12 01:23:51 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Mon, 11 Apr 2011 14:23:51 -0700 (PDT)
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <BANLkTimLGyyv6AtSaS2k3YKjRzhtyZHcxA@mail.gmail.com>
Message-ID: <372832.58244.qm@web34302.mail.mud.yahoo.com>

Dear.

Yes i need setup b leg with a accountcode different to a leg, like this

a leg = accountcode = foo
b leg = accountcode = foo1

I trying with 

?<action application="export" data="accountcode=foo"/>
But in my CDR for inbound and outbound call is equal to foo.

I need setup inbound call with accountcode = foo and outbound call with accountcode = foo1

Regards.




--- On Sun, 4/10/11, Michael Collins <msc at freeswitch.org> wrote:

From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Account ACL
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Sunday, April 10, 2011, 10:57 PM



On Sun, Apr 10, 2011 at 3:27 PM, FERNANDO VILLARROEL <fvillarroel at yahoo.com> wrote:

Dear Pablo.

Thank you for you help.

My problem is setup accountcode for outbound gateway (sip_profiles/external ). I will try and inform to you my tests results.


For inbound gateway the acoountcode is setup fine the  like this:

?????? <action application="set" data="accountcode=foo"/>
?????? <action application="bridge" data="sofia/gateway/ms6/$1$2"/>


I need setup CDR accountcode for my sip_profiles/external or in this example for gateway named ms6.

So, you need the variable "accountcode" to be "foo" on the B leg of this call? If so, just use "export" instead of "set" and it will be done
?<action application="export" data="accountcode=foo"/>
If that's not what you want then I recommend that you try explaining from the beginning (again) your problem because we are having a difficult time understanding what you are asking for.

Thanks,
MC

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From msc at freeswitch.org  Tue Apr 12 01:31:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Apr 2011 14:31:05 -0700
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <372832.58244.qm@web34302.mail.mud.yahoo.com>
References: <BANLkTimLGyyv6AtSaS2k3YKjRzhtyZHcxA@mail.gmail.com>
	<372832.58244.qm@web34302.mail.mud.yahoo.com>
Message-ID: <BANLkTi=0EKx0u_m7XvEFLv42NPNNa342BQ@mail.gmail.com>

On Mon, Apr 11, 2011 at 2:23 PM, FERNANDO VILLARROEL
<fvillarroel at yahoo.com>wrote:

> Dear.
>
> Yes i need setup b leg with a accountcode different to a leg, like this
>
> a leg = accountcode = foo
> b leg = accountcode = foo1
>
> I trying with
>
>  <action application="export" data="accountcode=foo"/>
>
> But in my CDR for inbound and outbound call is equal to foo.
>
> I need setup inbound call with accountcode = foo and outbound call with
> accountcode = foo1
>
>
How do you know that the outbound leg needs to be "foo1"? In other words,
where do you look to get the value of "foo1"?
-MC
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From fvillarroel at yahoo.com  Tue Apr 12 02:32:52 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Mon, 11 Apr 2011 15:32:52 -0700 (PDT)
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <BANLkTi=0EKx0u_m7XvEFLv42NPNNa342BQ@mail.gmail.com>
Message-ID: <860454.94851.qm@web34302.mail.mud.yahoo.com>

Dear Michael.

Thank you for you help. I will try to explain but excuse my bad english.

In my FS i received traffic from a gateway A and this traffic i? forward to another Gateway B.

Gateway A ----> My FS -----> Gateway B.

Ok in my CDR i have both calls, inbound and outbound calls. So i need setup inbound call with accountcode = foo and outbound call i need setup   with another accountcode variable like accountcode = foo1.

So like this i will can inform to my customer (Gateway A) how much traffic i receive from him and i will inform to my provider (Gateway B ) how much traffic i send to him.

It's possible or let me know another idea how i can do.

Regards.

--- On Mon, 4/11/11, Michael Collins <msc at freeswitch.org> wrote:

From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Account ACL
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Monday, April 11, 2011, 6:31 PM



On Mon, Apr 11, 2011 at 2:23 PM, FERNANDO VILLARROEL <fvillarroel at yahoo.com> wrote:

Dear.

Yes i need setup b leg with a accountcode different to a leg, like this

a leg = accountcode = foo
b leg = accountcode = foo1


I trying with 

?<action application="export" data="accountcode=foo"/>
But in my CDR for inbound and outbound call is equal to foo.


I need setup inbound call with accountcode = foo and outbound call with accountcode = foo1


How do you know that the outbound leg needs to be "foo1"? In other words, where do you look to get the value of "foo1"?
-MC


-----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
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From frankie.k.yiu at gmail.com  Tue Apr 12 03:36:11 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Mon, 11 Apr 2011 16:36:11 -0700
Subject: [Freeswitch-users] No DTMF event generated during a session when
	calling switch_core_session_send_dtmf()
Message-ID: <BANLkTim7rrHh0nCFLHRsVesu-4g4Tk=DDA@mail.gmail.com>

Hi there,

I am sending a DTMF key * through switch_core_session_send_dtmf() <in c++
code> during a session while an audio is playing through PlayAndGetDigits()
in Mod_managed, but to my surprised I do not receive any DTMF event
generated.  Is there something else that I need to do?  The status returns
success.

This is the actual code in my C++

switch_dtmf_t dtmf = { '*', switch_core_default_dtmf_duration(0), 0};
switch_status_t dtmfStatus = switch_core_session_send_dtmf(curSession,
&dtmf);

Please let me know.

Thanks,

Frankie
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From msc at freeswitch.org  Tue Apr 12 04:20:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 11 Apr 2011 17:20:45 -0700
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <860454.94851.qm@web34302.mail.mud.yahoo.com>
References: <BANLkTi=0EKx0u_m7XvEFLv42NPNNa342BQ@mail.gmail.com>
	<860454.94851.qm@web34302.mail.mud.yahoo.com>
Message-ID: <BANLkTikNc4qAXk-2M2vRwSvYJskB4gK=Cg@mail.gmail.com>

Fernando,

What is your native language? We have users who are fluent in Spanish,
Brazilian Portugese, and many others. Ask your question in your native
language and someone will be able to help those of us who speak only English
to understand.

That being say, I don't see what you can't set the account code on the B
leg. You can do this:
<action application="export" data="nolocal:accountcode=foo1"/>

However, you still have not told use how you know what value to use. If you
literally need to append "1" to the accountcode you can do something like
this:
<action application="export" data="nolocal:accountcode=${accountcode}1"/>

Be sure to use "nolocal:" so that your A leg accountcode value is not
affected. Try the above and let us know if that does what you need.

-MC

On Mon, Apr 11, 2011 at 3:32 PM, FERNANDO VILLARROEL
<fvillarroel at yahoo.com>wrote:

> Dear Michael.
>
> Thank you for you help. I will try to explain but excuse my bad english.
>
> In my FS i received traffic from a gateway A and this traffic i  forward to
> another Gateway B.
>
> Gateway A ----> My FS -----> Gateway B.
>
> Ok in my CDR i have both calls, inbound and outbound calls. So i need setup
> inbound call with accountcode = foo and outbound call i need setup with
> another accountcode variable like accountcode = foo1.
>
> So like this i will can inform to my customer (Gateway A) how much traffic
> i receive from him and i will inform to my provider (Gateway B ) how much
> traffic i send to him.
>
> It's possible or let me know another idea how i can do.
>
> Regards.
>
> --- On *Mon, 4/11/11, Michael Collins <msc at freeswitch.org>* wrote:
>
>
> From: Michael Collins <msc at freeswitch.org>
> Subject: Re: [Freeswitch-users] Account ACL
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Monday, April 11, 2011, 6:31 PM
>
>
>
> On Mon, Apr 11, 2011 at 2:23 PM, FERNANDO VILLARROEL <
> fvillarroel at yahoo.com <http://mc/compose?to=fvillarroel at yahoo.com>> wrote:
>
> Dear.
>
> Yes i need setup b leg with a accountcode different to a leg, like this
>
> a leg = accountcode = foo
> b leg = accountcode = foo1
>
> I trying with
>
>  <action application="export" data="accountcode=foo"/>
>
> But in my CDR for inbound and outbound call is equal to foo.
>
> I need setup inbound call with accountcode = foo and outbound call with
> accountcode = foo1
>
>
> How do you know that the outbound leg needs to be "foo1"? In other words,
> where do you look to get the value of "foo1"?
> -MC
>
>
> -----Inline Attachment Follows-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jason at jasonjgw.net  Tue Apr 12 04:23:41 2011
From: jason at jasonjgw.net (Jason White)
Date: Tue, 12 Apr 2011 00:23:41 +0000 (UTC)
Subject: [Freeswitch-users] FreeSWITCH sometimes binds to
	loopback	interface during boot
References: <intmi8$e9b$1@dough.gmane.org>
	<CB1CEFA1-0F43-451C-8E6F-A5AC5EC95728@freeswitch.org>
Message-ID: <io062d$882$1@dough.gmane.org>

Brian West  <freeswitch-users at lists.freeswitch.org> wrote:
>have you tried "sofia profile external restart"

Yes, I thought it would work, but it didn't change the address binding,
whereas restarting FreeSWITCH did.




From fvillarroel at yahoo.com  Tue Apr 12 05:44:28 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Mon, 11 Apr 2011 18:44:28 -0700 (PDT)
Subject: [Freeswitch-users] Account ACL SOLVED
In-Reply-To: <BANLkTikNc4qAXk-2M2vRwSvYJskB4gK=Cg@mail.gmail.com>
Message-ID: <647039.54386.qm@web34301.mail.mud.yahoo.com>

Dear Michael.

My native language is spanish.

Thank you very much for help me. I solved my problem with the script:

<action application="export" data="nolocal:accountcode=foo1"/>

Thank you.


--- On Mon, 4/11/11, Michael Collins <msc at freeswitch.org> wrote:

From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Account ACL
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Monday, April 11, 2011, 9:20 PM

Fernando,
What is your native language? We have users who are fluent in Spanish, Brazilian Portugese, and many others. Ask your question in your native language and someone will be able to help those of us who speak only English to understand.

That being say, I don't see what you can't set the account code on the B leg. You can do this:<action application="export" data="nolocal:accountcode=foo1"/>

However, you still have not told use how you know what value to use. If you literally need to append "1" to the accountcode you can do something like this:<action application="export" data="nolocal:accountcode=${accountcode}1"/>

Be sure to use "nolocal:" so that your A leg accountcode value is not affected. Try the above and let us know if that does what you need.
-MC


On Mon, Apr 11, 2011 at 3:32 PM, FERNANDO VILLARROEL <fvillarroel at yahoo.com> wrote:

Dear Michael.

Thank you for you help. I will try to explain but excuse my bad english.

In my FS i received traffic from a gateway A and this traffic i? forward to another Gateway B.


Gateway A ----> My FS -----> Gateway B.

Ok in my CDR i have both calls, inbound and outbound calls. So i need setup inbound call with accountcode = foo and outbound call i need setup   with another accountcode variable like accountcode = foo1.


So like this i will can inform to my customer (Gateway A) how much traffic i receive from him and i will inform to my provider (Gateway B ) how much traffic i send to him.

It's possible or let me know another idea how i can do.


Regards.

--- On Mon, 4/11/11, Michael Collins <msc at freeswitch.org> wrote:


From: Michael Collins <msc at freeswitch.org>
Subject: Re: [Freeswitch-users] Account ACL
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>

Date: Monday, April 11, 2011, 6:31 PM



On Mon, Apr 11, 2011 at 2:23 PM, FERNANDO VILLARROEL <fvillarroel at yahoo.com> wrote:



Dear.

Yes i need setup b leg with a accountcode different to a leg, like this

a leg = accountcode = foo
b leg = accountcode = foo1


I trying with 

?<action application="export" data="accountcode=foo"/>
But in my CDR for inbound and outbound call is equal to foo.



I need setup inbound call with accountcode = foo and outbound call with accountcode = foo1


How do you know that the outbound leg needs to be "foo1"? In other words, where do you look to get the value of "foo1"?

-MC


-----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





-----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
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From pablosaro at gmail.com  Tue Apr 12 07:20:01 2011
From: pablosaro at gmail.com (Pablo Hernan Saro)
Date: Tue, 12 Apr 2011 00:20:01 -0300
Subject: [Freeswitch-users] Account ACL
In-Reply-To: <BANLkTikNc4qAXk-2M2vRwSvYJskB4gK=Cg@mail.gmail.com>
References: <BANLkTi=0EKx0u_m7XvEFLv42NPNNa342BQ@mail.gmail.com>
	<860454.94851.qm@web34302.mail.mud.yahoo.com>
	<BANLkTikNc4qAXk-2M2vRwSvYJskB4gK=Cg@mail.gmail.com>
Message-ID: <BANLkTimGCKh6enA3+C27jM54OK=K038CjA@mail.gmail.com>

Hi Fernando, try the solution proposed by MC. If you still having problems,
please drop an email in your native language and someone will be glad to
help you (my native language is spanish by the way).


On Mon, Apr 11, 2011 at 9:20 PM, Michael Collins <msc at freeswitch.org> wrote:

> Fernando,
>
> What is your native language? We have users who are fluent in Spanish,
> Brazilian Portugese, and many others. Ask your question in your native
> language and someone will be able to help those of us who speak only English
> to understand.
>
> That being say, I don't see what you can't set the account code on the B
> leg. You can do this:
> <action application="export" data="nolocal:accountcode=foo1"/>
>
> However, you still have not told use how you know what value to use. If you
> literally need to append "1" to the accountcode you can do something like
> this:
> <action application="export" data="nolocal:accountcode=${accountcode}1"/>
>
> Be sure to use "nolocal:" so that your A leg accountcode value is not
> affected. Try the above and let us know if that does what you need.
>
> -MC
>
>
> On Mon, Apr 11, 2011 at 3:32 PM, FERNANDO VILLARROEL <
> fvillarroel at yahoo.com> wrote:
>
>> Dear Michael.
>>
>> Thank you for you help. I will try to explain but excuse my bad english.
>>
>> In my FS i received traffic from a gateway A and this traffic i  forward
>> to another Gateway B.
>>
>> Gateway A ----> My FS -----> Gateway B.
>>
>> Ok in my CDR i have both calls, inbound and outbound calls. So i need
>> setup inbound call with accountcode = foo and outbound call i need setup
>> with another accountcode variable like accountcode = foo1.
>>
>> So like this i will can inform to my customer (Gateway A) how much traffic
>> i receive from him and i will inform to my provider (Gateway B ) how much
>> traffic i send to him.
>>
>> It's possible or let me know another idea how i can do.
>>
>> Regards.
>>
>> --- On *Mon, 4/11/11, Michael Collins <msc at freeswitch.org>* wrote:
>>
>>
>> From: Michael Collins <msc at freeswitch.org>
>> Subject: Re: [Freeswitch-users] Account ACL
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Date: Monday, April 11, 2011, 6:31 PM
>>
>>
>>
>> On Mon, Apr 11, 2011 at 2:23 PM, FERNANDO VILLARROEL <
>> fvillarroel at yahoo.com <http://mc/compose?to=fvillarroel at yahoo.com>>wrote:
>>
>>  Dear.
>>
>> Yes i need setup b leg with a accountcode different to a leg, like this
>>
>> a leg = accountcode = foo
>> b leg = accountcode = foo1
>>
>> I trying with
>>
>>  <action application="export" data="accountcode=foo"/>
>>
>> But in my CDR for inbound and outbound call is equal to foo.
>>
>> I need setup inbound call with accountcode = foo and outbound call with
>> accountcode = foo1
>>
>>
>> How do you know that the outbound leg needs to be "foo1"? In other words,
>> where do you look to get the value of "foo1"?
>> -MC
>>
>>
>> -----Inline Attachment Follows-----
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<http://mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From boris at tagnet.ru  Tue Apr 12 08:55:03 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 12 Apr 2011 10:55:03 +0600
Subject: [Freeswitch-users] Still can't understand gateways
Message-ID: <4DA3DB27.4010205@tagnet.ru>

Hello!

     I have profile named ipbx with gateway defined:
<include>
<gateway name="test.tagnet.hn">
<!--/// account username *required* ///-->
<param name="username" value="test"/>
<!--/// auth realm: *optional* same as gateway name, if blank ///-->
<param name="realm" value="public"/>
<!--/// username to use in from: *optional* same as  username, if blank 
///-->
<param name="from-user" value="gw_test"/>
<!--/// account password *required* ///-->
<param name="password" value="2007"/>
<!--/// extension for inbound calls: *optional* same as username, if 
blank ///
-->
<param name="extension" value="gw_test"/>
<!--/// proxy host: *optional* same as realm, if blank ///-->
<param name="proxy" value="192.168.3.253"/>
<!--/// send register to this proxy: *optional* same as proxy, if blank 
///-->
<!--<param name="register-proxy" value="mysbc.com"/>-->
<!--/// expire in seconds: *optional* 3600, if blank ///-->
<!--<param name="expire-seconds" value="60"/>-->
<!--/// do not register ///-->
<param name="register" value="false"/>
<!-- which transport to use for register -->
<!--<param name="register-transport" value="udp"/>-->
<!--How many seconds before a retry when a failure or timeout occurs -->
<!--<param name="retry-seconds" value="30"/>-->
<!--Use the callerid of an inbound call in the from field on outbound 
calls vi
a this gateway -->
<!--<param name="caller-id-in-from" value="false"/>-->
<!--extra sip params to send in the contact-->
<!--<param name="contact-params" value="tport=tcp"/>-->
<!--send an options ping every x seconds, failure will unregister and/or 
mark
it down-->
<!--<param name="ping" value="25"/>-->
</gateway>
</include>


Gateway is present with running profile:
60      RUNNING (1)
      ipbx::test.tagnet.hn       gateway                     
sip:test at 192.168.3.253      NOREG


There is an extension in context public:
<extension name="gw_test">
<condition field="destination_number" expression="^gw_test$">
<action application="info" data="ALERT"/>
<action application="hangup" data="NO_ROUTE_DESTINATION"/>
</condition>
</extension>


So, my inbound calls from this gateway should go to extension gw_test? 
But they don't...
What is wrong with my config? FreeSWITCH Version 1.0.head (git-1c95ad9 
2011-01-20 22-43-50 -0300)

-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From rebel.pappas at gmail.com  Tue Apr 12 11:31:10 2011
From: rebel.pappas at gmail.com (alex pappas)
Date: Tue, 12 Apr 2011 10:31:10 +0300
Subject: [Freeswitch-users] Calling card test
In-Reply-To: <BANLkTinwW15pLdjACgCWc+wnxV+Zr9eqag@mail.gmail.com>
References: <BANLkTikMO8Jg_Yx9HtTxC7OXWFPjbJh4Lg@mail.gmail.com>
	<8436C9B0-A124-4492-ABEC-D83FE1CCAB62@gmail.com>
	<BANLkTimiYN7ta3-T4q_ZTx8Y13b=awDjig@mail.gmail.com>
	<BANLkTinr+P=1H9WW5LpmQQFRkMkrYf-LWg@mail.gmail.com>
	<B6DE24D8-DB8E-487B-AFEF-405B56F509B7@gmail.com>
	<BANLkTi=FPfiixcyvmn94Oz0PU=SW5NsBsg@mail.gmail.com>
	<BANLkTinwW15pLdjACgCWc+wnxV+Zr9eqag@mail.gmail.com>
Message-ID: <BANLkTinq3_9Mn84fm54TmuYSo5U0m7d73Q@mail.gmail.com>

Michael,

Thank you for all the info!

Cheers
\Alx


On Mon, Apr 11, 2011 at 7:21 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Apr 11, 2011 at 7:10 AM, alex pappas <rebel.pappas at gmail.com>wrote:
>
>> Nothing wrong with nibblebill. I'm trying to simulate an existing system
>> and I need to start everything from scratch. That's why I'm asking about how
>> I can have performance with Freeswitch.
>>
>
> Scripting right from the dialplan has a lower barrier to entry but will not
> scale as well as using outbound event socket. If I were in your shoes I
> would roll up my sleeves and learn ESL. Pick your favorite language. ESL has
> bindings for:
> C/C++
> Perl
> PHP
> Python
> Ruby
> TCL
>
> I highly recommend getting the FreeSWITCH book and reading chapter 9. Of
> course, chapters 1 through 6 are also important for getting a foundation for
> using FreeSWITCH, but chapter 9 has a lot of solid information about using
> the event socket and ESL.
>
> Note: you asked about using inbound vs. outbound event socket. If I
> understand your situation correctly you need to handle the case where an
> outside party calls in to your FS server. This means you need outbound event
> socket. (See the 'socket' dp tool on the wiki.)
>
> Enjoy!
> -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
>
>
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From u2nsam at gmail.com  Tue Apr 12 13:33:42 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 12 Apr 2011 15:03:42 +0530
Subject: [Freeswitch-users] proxy SDP
Message-ID: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>

Hi all,


Is there method to just proxy SDP through Freeswitch through sip profile ?



Regards
Sam
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From Richard.Smith at streetcar.co.uk  Tue Apr 12 13:40:23 2011
From: Richard.Smith at streetcar.co.uk (Richard Smith)
Date: Tue, 12 Apr 2011 10:40:23 +0100
Subject: [Freeswitch-users] Freeswitch deployment examples
Message-ID: <DB1E3945816C0244960CA0D0F6D2747903B3E642@streetcarsrv03.streetcar.priv>

Hi,

 

We're currently undergoing an exercise to replace our current aging
Asterisk system with something else and Freeswitch is one of the options
we're considering.

 

Our current Asterisk deployment has done us well, however the hardware
is aging, and the configuration hasn't changed much except for some
basic additions and changes. We're looking to replace it and add in
capability to extend and embrace new channels of communication.

 

We have some reservations over the implementation and would ideally like
to talk to some people who are running it in anger.

 

Specifically we're looking to discuss mod_callcentre or OpenACD
functionality.

 

If someone is in and around London (UK) and  would like to discuss their
experiences of running a callcentre on Freeswitch with us, possibly over
some beers (on us of course) we'd be really grateful.

 

Kind regards

 

Richard Smith

Systems Administrator
t 0203 004 7890



 

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From me at nevian.org  Tue Apr 12 13:52:12 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Tue, 12 Apr 2011 13:52:12 +0400
Subject: [Freeswitch-users] Still can't understand gateways
In-Reply-To: <4DA3DB27.4010205@tagnet.ru>
References: <4DA3DB27.4010205@tagnet.ru>
Message-ID: <1091151302601933@web154.yandex.ru>

Hello,

12.04.2011, 08:55, "Boris Kovalenko" <boris at tagnet.ru>:
> Hello!
>
> ?????I have profile named ipbx with gateway defined:
> <include>
> <gateway name="test.tagnet.hn">
> <!--/// account username *required* ///-->
> <param name="username" value="test"/>
> <!--/// auth realm: *optional* same as gateway name, if blank ///-->
> <param name="realm" value="public"/>
> <!--/// username to use in from: *optional* same as ?username, if blank
> ///-->
> <param name="from-user" value="gw_test"/>
> <!--/// account password *required* ///-->
> <param name="password" value="2007"/>
> <!--/// extension for inbound calls: *optional* same as username, if
> blank ///
> -->
> <param name="extension" value="gw_test"/>
> There is an extension in context public:
> <extension name="gw_test">
> <condition field="destination_number" expression="^gw_test$">
> <action application="info" data="ALERT"/>
> <action application="hangup" data="NO_ROUTE_DESTINATION"/>
> </condition>
> </extension>
>
> So, my inbound calls from this gateway should go to extension gw_test?
> But they don't...
> What is wrong with my config? FreeSWITCH Version 1.0.head (git-1c95ad9
> 2011-01-20 22-43-50 -0300)


Is this GW defined in public context?
Hint: realm != context!
Pls, look at example 'incoming.xml' in public profile in default config set

-- 
wbr,
Serge


From boris at tagnet.ru  Tue Apr 12 13:57:10 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 12 Apr 2011 15:57:10 +0600
Subject: [Freeswitch-users] Still can't understand gateways
In-Reply-To: <1091151302601933@web154.yandex.ru>
References: <4DA3DB27.4010205@tagnet.ru> <1091151302601933@web154.yandex.ru>
Message-ID: <4DA421F6.5070504@tagnet.ru>

Hello!

     Hmm... I thinked GW is defined within profile not within context???
> Hello,
>
> 12.04.2011, 08:55, "Boris Kovalenko"<boris at tagnet.ru>:
>> Hello!
>>
>>       I have profile named ipbx with gateway defined:
>> <include>
>> <gateway name="test.tagnet.hn">
>> <!--/// account username *required* ///-->
>> <param name="username" value="test"/>
>> <!--/// auth realm: *optional* same as gateway name, if blank ///-->
>> <param name="realm" value="public"/>
>> <!--/// username to use in from: *optional* same as  username, if blank
>> ///-->
>> <param name="from-user" value="gw_test"/>
>> <!--/// account password *required* ///-->
>> <param name="password" value="2007"/>
>> <!--/// extension for inbound calls: *optional* same as username, if
>> blank ///
>> -->
>> <param name="extension" value="gw_test"/>
>> There is an extension in context public:
>> <extension name="gw_test">
>> <condition field="destination_number" expression="^gw_test$">
>> <action application="info" data="ALERT"/>
>> <action application="hangup" data="NO_ROUTE_DESTINATION"/>
>> </condition>
>> </extension>
>>
>> So, my inbound calls from this gateway should go to extension gw_test?
>> But they don't...
>> What is wrong with my config? FreeSWITCH Version 1.0.head (git-1c95ad9
>> 2011-01-20 22-43-50 -0300)
>
> Is this GW defined in public context?
> Hint: realm != context!
> Pls, look at example 'incoming.xml' in public profile in default config set
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From steveayre at gmail.com  Tue Apr 12 14:20:06 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 12 Apr 2011 11:20:06 +0100
Subject: [Freeswitch-users] Gateway with dynamic IP address
In-Reply-To: <BANLkTi=4wX_cc9hPU0S1-O0UT0RE_Tom0Q@mail.gmail.com>
References: <C9C8E1CC-4621-44FF-8D19-D8DB35D41530@gmail.com>
	<AANLkTinQNJpc0UP4hSjnSaa5zqcqgVoLhV0ScrPDvwNN@mail.gmail.com>
	<828493E7-A5E7-4896-844F-271AB72AD38B@gmail.com>
	<AANLkTinZWWijJBBWr0ArbtSGjHuv1cpHDDxKiZqj4Agg@mail.gmail.com>
	<BANLkTik1B7XhyCnDe2zs3Pqob4MD4evaPQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58C43A41E5@cooper>
	<BANLkTi=4wX_cc9hPU0S1-O0UT0RE_Tom0Q@mail.gmail.com>
Message-ID: <BANLkTi=8LfL-CMkczXgJg8FOF1_FTYtxdQ@mail.gmail.com>

To dial a user you use <action application="bridge" data="user/username"/>,
FS then figures out the Sofia URI for you from the registration.

-Steve

On 11 April 2011 20:53, Nick Rosier <nick.rosier at gmail.com> wrote:

> On 5 April 2011 22:45, Peter Olsson <peter.olsson at visionutveckling.se>
> wrote:
> > What you wan't to do is to add a user. Then you dial this user, which by
> then is registered in FreeSWITCH, and it will find the path.
> >
> > So no gateway in this case, it's when you want to register to an external
> server, a user is when someone registers to you, and you wan't to be able to
> dial outside through this.
> >
> > /Peter
>
> Can someone help me with the URI. It's driving me crazy.
> This is what I've got but it's not working:
>
> <action application="bridge"
> data="sofia/sipinterface_1/trunk1 at pbx.domain.com/$1"/>
>
> What am I doing wrong?
>
> N.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From steveayre at gmail.com  Tue Apr 12 14:30:34 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 12 Apr 2011 11:30:34 +0100
Subject: [Freeswitch-users] Freeswitch Server Down
In-Reply-To: <BANLkTikLGEqSnL7yYnvVmBE6AvYgCBTXmg@mail.gmail.com>
References: <BANLkTimerjRCutH8LFaRjvCfSju6byO1ow@mail.gmail.com>
	<BANLkTikLGEqSnL7yYnvVmBE6AvYgCBTXmg@mail.gmail.com>
Message-ID: <637F58A1-8368-4E26-A192-3629D9BDAF8F@gmail.com>

Are you on Linux? If so run dmesg and see if there are any messages indicating freeswitch had a segmentation fault or general protection fault. If there is it's a bug and there will hopefully be a coredump file that will contain useful information for tracking the problem down.

Steve on iPhone

On 11 Apr 2011, at 18:49, Michael Collins <msc at freeswitch.org> wrote:

> 
> 
> On Mon, Apr 11, 2011 at 10:32 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
> Hi to all, 
> 
> Today, All my IVR are stopped working. 
> When I check the freeSwitch it was down. 
> I dont know the reason why Its Down. 
> How I can find the reason,  why my freeSwitch went down? 
> 
> You'll need to check logs to track down what happened. I know that sometimes you'll see lots of log lines in freeswitch.log and then all of a sudden nothing, so that will help you pinpoint when things went wrong. Possibly you'll see some log lines with errors or warnings. Or you might see that the "fsctl shutdown" command was sent to it. (Then you'll need to go hunt down whoever did that. :)
> 
> -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
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From jgallartm at gmail.com  Tue Apr 12 15:02:55 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Tue, 12 Apr 2011 13:02:55 +0200
Subject: [Freeswitch-users] Passing SIP headers from b-leg to a-leg
Message-ID: <BANLkTimZjq1EffupZKV1ais=zkx9R+D_WQ@mail.gmail.com>

Hi all

I'm trying to pass some custom X-headers in final failed responses (>400)
 from B-leg to A-leg...is there a way to accomplish this?

Thanks in advance
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From ce at kapper.net  Tue Apr 12 15:18:46 2011
From: ce at kapper.net (Clemens Ebentheuer)
Date: Tue, 12 Apr 2011 13:18:46 +0200
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
Message-ID: <1B19ABD72889C245AE8EEE08AC24103A28C423231C@exmachina.office.kapper.net>

http://wiki.freeswitch.org/wiki/Proxy_media#How_to_enable_it

ce
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sam
Sent: Tuesday, April 12, 2011 11:34 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] proxy SDP

Hi all,


Is there method to just proxy SDP through Freeswitch through sip profile ?



Regards
Sam
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From me at nevian.org  Tue Apr 12 15:27:24 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Tue, 12 Apr 2011 15:27:24 +0400
Subject: [Freeswitch-users] write/read-codec variables
Message-ID: <6141302607644@web133.yandex.ru>


Hi

I have transcoded session from iLBC to g711 and want to see this in CDRs via mod_cdr_csv.
I'm writing only legA to CDR, can I get these variables from legB in this situation?
Any other solution w/o parsing xml_cdr?


-- 
wbr,
Serge


From me at nevian.org  Tue Apr 12 15:27:44 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Tue, 12 Apr 2011 15:27:44 +0400
Subject: [Freeswitch-users] Still can't understand gateways
In-Reply-To: <4DA421F6.5070504@tagnet.ru>
References: <4DA3DB27.4010205@tagnet.ru> <1091151302601933@web154.yandex.ru>
	<4DA421F6.5070504@tagnet.ru>
Message-ID: <6481302607664@web133.yandex.ru>

Hello,

12.04.2011, 13:57, "Boris Kovalenko" <boris at tagnet.ru>;:

> ?Hello!
>
> ??????Hmm... I thinked GW is defined within profile not within context???

Pardon, of course you are right.. in general.
I meant that context public linked to external profile by default if you didn't set it explicitly elsewhere.
So if you defined your GW in internal profile it will search ext in default context, not public.

btw, you can declare gw in directory and even in user record:
http://wiki.freeswitch.org/wiki/Clarification:gateways

>> ??Hello,
>>
>> ??12.04.2011, 08:55, "Boris Kovalenko"<boris at tagnet.ru>;;:
>>> ??Hello!
>>>
>>> ????????I have profile named ipbx with gateway defined:
>>> ??<include>
>>> ??<gateway name="test.tagnet.hn">
>>> ??<!--/// account username *required* ///-->
>>> ??<param name="username" value="test"/>
>>> ??<!--/// auth realm: *optional* same as gateway name, if blank ///-->
>>> ??<param name="realm" value="public"/>
>>> ??<!--/// username to use in from: *optional* same as ?username, if blank
>>> ??///-->
>>> ??<param name="from-user" value="gw_test"/>
>>> ??<!--/// account password *required* ///-->
>>> ??<param name="password" value="2007"/>
>>> ??<!--/// extension for inbound calls: *optional* same as username, if
>>> ??blank ///
>>> ??-->
>>> ??<param name="extension" value="gw_test"/>
>>> ??There is an extension in context public:
>>> ??<extension name="gw_test">
>>> ??<condition field="destination_number" expression="^gw_test$">
>>> ??<action application="info" data="ALERT"/>
>>> ??<action application="hangup" data="NO_ROUTE_DESTINATION"/>
>>> ??</condition>
>>> ??</extension>
>>>
>>> ??So, my inbound calls from this gateway should go to extension gw_test?
>>> ??But they don't...
>>> ??What is wrong with my config? FreeSWITCH Version 1.0.head (git-1c95ad9
>>> ??2011-01-20 22-43-50 -0300)
>> ??Is this GW defined in public context?
>> ??Hint: realm != context!
>> ??Pls, look at example 'incoming.xml' in public profile in default config set
> ?--
> ?? ?????????,
> ???????? ?????????
> ?????? "??????"
> ???(3435) 494991
>
> ?_______________________________________________
> ?FreeSWITCH-users mailing list
> ?FreeSWITCH-users at lists.freeswitch.org
> ?http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> ?UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> ?http://www.freeswitch.org
-- 
wbr,
Serge


From boris at tagnet.ru  Tue Apr 12 16:07:51 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 12 Apr 2011 18:07:51 +0600
Subject: [Freeswitch-users] Still can't understand gateways
In-Reply-To: <6481302607664@web133.yandex.ru>
References: <4DA3DB27.4010205@tagnet.ru>
	<1091151302601933@web154.yandex.ru>	<4DA421F6.5070504@tagnet.ru>
	<6481302607664@web133.yandex.ru>
Message-ID: <4DA44097.1000403@tagnet.ru>

Of course... my gw is looking in public context:
2011-04-12 18:06:52.529284 [INFO] mod_dialplan_xml.c:331 Processing test 
gw <12>->1234 in context public
2011-04-12 18:06:52.530329 [INFO] mod_dialplan_xml.c:331 Processing test 
gw <12>->ext_translate_extsrc in context features

But why the extension I configured does not work for it?

> Hello,
>
> 12.04.2011, 13:57, "Boris Kovalenko"<boris at tagnet.ru>;:
>
>>   Hello!
>>
>>        Hmm... I thinked GW is defined within profile not within context???
> Pardon, of course you are right.. in general.
> I meant that context public linked to external profile by default if you didn't set it explicitly elsewhere.
> So if you defined your GW in internal profile it will search ext in default context, not public.
>
> btw, you can declare gw in directory and even in user record:
> http://wiki.freeswitch.org/wiki/Clarification:gateways
>
>>>    Hello,
>>>
>>>    12.04.2011, 08:55, "Boris Kovalenko"<boris at tagnet.ru>;;:
>>>>    Hello!
>>>>
>>>>          I have profile named ipbx with gateway defined:
>>>>    <include>
>>>>    <gateway name="test.tagnet.hn">
>>>>    <!--/// account username *required* ///-->
>>>>    <param name="username" value="test"/>
>>>>    <!--/// auth realm: *optional* same as gateway name, if blank ///-->
>>>>    <param name="realm" value="public"/>
>>>>    <!--/// username to use in from: *optional* same as  username, if blank
>>>>    ///-->
>>>>    <param name="from-user" value="gw_test"/>
>>>>    <!--/// account password *required* ///-->
>>>>    <param name="password" value="2007"/>
>>>>    <!--/// extension for inbound calls: *optional* same as username, if
>>>>    blank ///
>>>>    -->
>>>>    <param name="extension" value="gw_test"/>
>>>>    There is an extension in context public:
>>>>    <extension name="gw_test">
>>>>    <condition field="destination_number" expression="^gw_test$">
>>>>    <action application="info" data="ALERT"/>
>>>>    <action application="hangup" data="NO_ROUTE_DESTINATION"/>
>>>>    </condition>
>>>>    </extension>
>>>>
>>>>    So, my inbound calls from this gateway should go to extension gw_test?
>>>>    But they don't...
>>>>    What is wrong with my config? FreeSWITCH Version 1.0.head (git-1c95ad9
>>>>    2011-01-20 22-43-50 -0300)
>>>    Is this GW defined in public context?
>>>    Hint: realm != context!
>>>    Pls, look at example 'incoming.xml' in public profile in default config set
>>   --
>>   ? ?????????,
>>     ????? ?????????
>>     ??? "??????"
>>     (3435) 494991
>>
>>   _______________________________________________
>>   FreeSWITCH-users mailing list
>>   FreeSWITCH-users at lists.freeswitch.org
>>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>   UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>   http://www.freeswitch.org


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From lakindia89 at gmail.com  Tue Apr 12 16:42:08 2011
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 12 Apr 2011 18:12:08 +0530
Subject: [Freeswitch-users] How to block CHAT done by using Twinkle
Message-ID: <BANLkTino-3RdE72W534WbWxXF_XGcwbKgQ@mail.gmail.com>

Hi all,
 Can any one please tell me how to block only a particular SIP packet,
especially MESSAGE packet.

The reason is,

The users are using Twinkle as softphone, and they are able to CHAT with
other users. Now I want to disable the CHAT message. When I looked into the
logs and sip trace, I found that the packet exchanged during CHAT is
"MESSAGE" packet. So is there any way to block this packet alone?? or is
there some other way to disable the CHAT??
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From ovvenkatesan at gmail.com  Tue Apr 12 15:04:33 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Tue, 12 Apr 2011 16:34:33 +0530
Subject: [Freeswitch-users] Freeswitch Server Down
In-Reply-To: <637F58A1-8368-4E26-A192-3629D9BDAF8F@gmail.com>
References: <BANLkTimerjRCutH8LFaRjvCfSju6byO1ow@mail.gmail.com>
	<BANLkTikLGEqSnL7yYnvVmBE6AvYgCBTXmg@mail.gmail.com>
	<637F58A1-8368-4E26-A192-3629D9BDAF8F@gmail.com>
Message-ID: <BANLkTikmZfQjM3wEiVYY+2xuQruCmmaUjQ@mail.gmail.com>

Hi Steven,

Thanks for your response.
Yes, Its on linux machine and
Since, I am new to linux platform I could
not able to understand the log file.
please find the attachment of *dmesg_messages*

I am getting error like

wp_tdmapi_read_msg:1296 User API Error: User Rx Len=1064 < Driver Rx
Len=1567 (hdr=64). User API must increase expected rx length!

I dont know, what is this mean.

Can you guide me please, what is went wrong and how to avoid in future?


Regards,
Venkat.




On Tue, Apr 12, 2011 at 4:00 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Are you on Linux? If so run dmesg and see if there are any messages
> indicating freeswitch had a segmentation fault or general protection fault.
> If there is it's a bug and there will hopefully be a coredump file that will
> contain useful information for tracking the problem down.
>
> Steve on iPhone
>
> On 11 Apr 2011, at 18:49, Michael Collins <msc at freeswitch.org> wrote:
>
>
>
> On Mon, Apr 11, 2011 at 10:32 AM, ovvenkat < <ovvenkatesan at gmail.com>
> ovvenkatesan at gmail.com> wrote:
>
>> Hi to all,
>>
>> Today, All my IVR are stopped working.
>> When I check the freeSwitch it was down.
>> I dont know the reason why Its Down.
>> How I can find the reason,  why my freeSwitch went down?
>>
>
> You'll need to check logs to track down what happened. I know that
> sometimes you'll see lots of log lines in freeswitch.log and then all of a
> sudden nothing, so that will help you pinpoint when things went wrong.
> Possibly you'll see some log lines with errors or warnings. Or you might see
> that the "fsctl shutdown" command was sent to it. (Then you'll need to go
> hunt down whoever did that. :)
>
> -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
>
>


-- 

If you have come to help me, you are wasting your time.
If you have come to because your liberation is bound up in mine, we can work
together.


Regards
Venkatesan OV.
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From Nabble at slickdeals.endjunk.com  Tue Apr 12 16:56:32 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 12 Apr 2011 05:56:32 -0700 (PDT)
Subject: [Freeswitch-users] Freeswitch deployment examples
In-Reply-To: <DB1E3945816C0244960CA0D0F6D2747903B3E642@streetcarsrv03.streetcar.priv>
References: <DB1E3945816C0244960CA0D0F6D2747903B3E642@streetcarsrv03.streetcar.priv>
Message-ID: <1302612992569-6264995.post@n2.nabble.com>

I am just curious what platform is your current Asterisk PBX System hosted on
and what is the average number of calls/second (CPS) on your current system?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-deployment-examples-tp6264484p6264995.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Tue Apr 12 17:07:11 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 12 Apr 2011 14:07:11 +0100
Subject: [Freeswitch-users] Still can't understand gateways
In-Reply-To: <4DA44097.1000403@tagnet.ru>
References: <4DA3DB27.4010205@tagnet.ru> <1091151302601933@web154.yandex.ru>
	<4DA421F6.5070504@tagnet.ru> <6481302607664@web133.yandex.ru>
	<4DA44097.1000403@tagnet.ru>
Message-ID: <BANLkTin4xOx3_5ymnt048wM2ygT2cp1pCg@mail.gmail.com>

Because you're matching the destination_number "gw_test". The logs show
you're dialing 1234 and then ext_translate_extsrc though, which don't match
that condition.

-Steve


2011/4/12 Boris Kovalenko <boris at tagnet.ru>

> Of course... my gw is looking in public context:
> 2011-04-12 18:06:52.529284 [INFO] mod_dialplan_xml.c:331 Processing test
> gw <12>->1234 in context public
> 2011-04-12 18:06:52.530329 [INFO] mod_dialplan_xml.c:331 Processing test
> gw <12>->ext_translate_extsrc in context features
>
> But why the extension I configured does not work for it?
>
> > Hello,
> >
> > 12.04.2011, 13:57, "Boris Kovalenko"<boris at tagnet.ru>;:
> >
> >>   Hello!
> >>
> >>        Hmm... I thinked GW is defined within profile not within
> context???
> > Pardon, of course you are right.. in general.
> > I meant that context public linked to external profile by default if you
> didn't set it explicitly elsewhere.
> > So if you defined your GW in internal profile it will search ext in
> default context, not public.
> >
> > btw, you can declare gw in directory and even in user record:
> > http://wiki.freeswitch.org/wiki/Clarification:gateways
> >
> >>>    Hello,
> >>>
> >>>    12.04.2011, 08:55, "Boris Kovalenko"<boris at tagnet.ru>;;:
> >>>>    Hello!
> >>>>
> >>>>          I have profile named ipbx with gateway defined:
> >>>>    <include>
> >>>>    <gateway name="test.tagnet.hn">
> >>>>    <!--/// account username *required* ///-->
> >>>>    <param name="username" value="test"/>
> >>>>    <!--/// auth realm: *optional* same as gateway name, if blank
> ///-->
> >>>>    <param name="realm" value="public"/>
> >>>>    <!--/// username to use in from: *optional* same as  username, if
> blank
> >>>>    ///-->
> >>>>    <param name="from-user" value="gw_test"/>
> >>>>    <!--/// account password *required* ///-->
> >>>>    <param name="password" value="2007"/>
> >>>>    <!--/// extension for inbound calls: *optional* same as username,
> if
> >>>>    blank ///
> >>>>    -->
> >>>>    <param name="extension" value="gw_test"/>
> >>>>    There is an extension in context public:
> >>>>    <extension name="gw_test">
> >>>>    <condition field="destination_number" expression="^gw_test$">
> >>>>    <action application="info" data="ALERT"/>
> >>>>    <action application="hangup" data="NO_ROUTE_DESTINATION"/>
> >>>>    </condition>
> >>>>    </extension>
> >>>>
> >>>>    So, my inbound calls from this gateway should go to extension
> gw_test?
> >>>>    But they don't...
> >>>>    What is wrong with my config? FreeSWITCH Version 1.0.head
> (git-1c95ad9
> >>>>    2011-01-20 22-43-50 -0300)
> >>>    Is this GW defined in public context?
> >>>    Hint: realm != context!
> >>>    Pls, look at example 'incoming.xml' in public profile in default
> config set
> >>   --
> >>   ? ?????????,
> >>     ????? ?????????
> >>     ??? "??????"
> >>     (3435) 494991
> >>
> >>   _______________________________________________
> >>   FreeSWITCH-users mailing list
> >>   FreeSWITCH-users at lists.freeswitch.org
> >>   http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>   UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>   http://www.freeswitch.org
>
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   (3435) 494991
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From boris at tagnet.ru  Tue Apr 12 17:38:05 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 12 Apr 2011 19:38:05 +0600
Subject: [Freeswitch-users] Still can't understand gateways
In-Reply-To: <BANLkTin4xOx3_5ymnt048wM2ygT2cp1pCg@mail.gmail.com>
References: <4DA3DB27.4010205@tagnet.ru>
	<1091151302601933@web154.yandex.ru>	<4DA421F6.5070504@tagnet.ru>
	<6481302607664@web133.yandex.ru>	<4DA44097.1000403@tagnet.ru>
	<BANLkTin4xOx3_5ymnt048wM2ygT2cp1pCg@mail.gmail.com>
Message-ID: <4DA455BD.8020206@tagnet.ru>

Hello!

     But.... reading the docs:
||

<!--/// extension for inbound calls: *optional* same as username, if blank ///-->
<!--<param name="extension" value="cluecon"/>-->


What this parameter means? I thinked that if extension is specified all 
incoming calls are placed to this extension. Isn't?

> Because you're matching the destination_number "gw_test". The logs 
> show you're dialing 1234 and then ext_translate_extsrc though, which 
> don't match that condition.
>
> -Steve
>
>
> 2011/4/12 Boris Kovalenko <boris at tagnet.ru <mailto:boris at tagnet.ru>>
>
>     Of course... my gw is looking in public context:
>     2011-04-12 18:06:52.529284 [INFO] mod_dialplan_xml.c:331
>     Processing test
>     gw <12>->1234 in context public
>     2011-04-12 18:06:52.530329 [INFO] mod_dialplan_xml.c:331
>     Processing test
>     gw <12>->ext_translate_extsrc in context features
>
>     But why the extension I configured does not work for it?
>
>     > Hello,
>     >
>     > 12.04.2011, 13:57, "Boris Kovalenko"<boris at tagnet.ru
>     <mailto:boris at tagnet.ru>>;:
>     >
>     >>   Hello!
>     >>
>     >>        Hmm... I thinked GW is defined within profile not within
>     context???
>     > Pardon, of course you are right.. in general.
>     > I meant that context public linked to external profile by
>     default if you didn't set it explicitly elsewhere.
>     > So if you defined your GW in internal profile it will search ext
>     in default context, not public.
>     >
>     > btw, you can declare gw in directory and even in user record:
>     > http://wiki.freeswitch.org/wiki/Clarification:gateways
>     >
>     >>>    Hello,
>     >>>
>     >>>    12.04.2011, 08:55, "Boris Kovalenko"<boris at tagnet.ru
>     <mailto:boris at tagnet.ru>>;;:
>     >>>>    Hello!
>     >>>>
>     >>>>          I have profile named ipbx with gateway defined:
>     >>>> <include>
>     >>>> <gateway name="test.tagnet.hn <http://test.tagnet.hn>">
>     >>>> <!--/// account username *required* ///-->
>     >>>> <param name="username" value="test"/>
>     >>>> <!--/// auth realm: *optional* same as gateway name, if blank
>     ///-->
>     >>>> <param name="realm" value="public"/>
>     >>>> <!--/// username to use in from: *optional* same as
>      username, if blank
>     >>>>    ///-->
>     >>>> <param name="from-user" value="gw_test"/>
>     >>>> <!--/// account password *required* ///-->
>     >>>> <param name="password" value="2007"/>
>     >>>> <!--/// extension for inbound calls: *optional* same as
>     username, if
>     >>>>    blank ///
>     >>>>    -->
>     >>>> <param name="extension" value="gw_test"/>
>     >>>>    There is an extension in context public:
>     >>>> <extension name="gw_test">
>     >>>> <condition field="destination_number" expression="^gw_test$">
>     >>>> <action application="info" data="ALERT"/>
>     >>>> <action application="hangup" data="NO_ROUTE_DESTINATION"/>
>     >>>> </condition>
>     >>>> </extension>
>     >>>>
>     >>>>    So, my inbound calls from this gateway should go to
>     extension gw_test?
>     >>>>    But they don't...
>     >>>>    What is wrong with my config? FreeSWITCH Version 1.0.head
>     (git-1c95ad9
>     >>>>    2011-01-20 22-43-50 -0300)
>     >>>    Is this GW defined in public context?
>     >>>    Hint: realm != context!
>     >>>    Pls, look at example 'incoming.xml' in public profile in
>     default config set
>     >>   --
>     >>   ? ?????????,
>     >>     ????? ?????????
>     >>     ??? "??????"
>     >>     (3435) 494991
>     >>
>     >>   _______________________________________________
>     >>   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
>
>
>     --
>     ? ?????????,
>       ????? ?????????
>       ??? "??????"
>       (3435) 494991
>
>
>
>     _______________________________________________
>     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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991


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From kris at kriskinc.com  Tue Apr 12 17:47:28 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Tue, 12 Apr 2011 09:47:28 -0400
Subject: [Freeswitch-users] Still can't understand gateways
In-Reply-To: <4DA455BD.8020206@tagnet.ru>
References: <4DA3DB27.4010205@tagnet.ru> <1091151302601933@web154.yandex.ru>
	<4DA421F6.5070504@tagnet.ru> <6481302607664@web133.yandex.ru>
	<4DA44097.1000403@tagnet.ru>
	<BANLkTin4xOx3_5ymnt048wM2ygT2cp1pCg@mail.gmail.com>
	<4DA455BD.8020206@tagnet.ru>
Message-ID: <BANLkTimXWpBUVaUS4MzvjfV+3Z3SjGw5vg@mail.gmail.com>

This parameter specifies the value FreeSWITCH is going to use for the
username portion of the Contact URI when registering to your gateway:

extension = cluecon

FreeSWITCH sends a REGISTER packet with a contact address like this:

Contact: cluecon at your.ip.address

With a "standard" setup your provider is supposed to send calls to the
value provided in the Contact address with INVITEs to
cluecon at your.ip.address.  However, many providers (to implement DID
services) ignore the username portion and just use the host portion
(IP address) so what you end up with is an INVITE to
8005551212 at your.ip.address.

If I saw a SIP trace I'd know for sure but that's probably what's going on.

On Tue, Apr 12, 2011 at 9:38 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ??? But.... reading the docs:
>
> <!--/// extension for inbound calls: *optional* same as username, if blank
> ///-->
> <!--<param name="extension" value="cluecon"/>-->
>
> What this parameter means? I thinked that if extension is specified all
> incoming calls are placed to this extension. Isn't?
>
> Because you're matching the destination_number "gw_test". The logs show
> you're dialing 1234 and then ext_translate_extsrc though, which don't match
> that condition.
>
> -Steve
>
>
> 2011/4/12 Boris Kovalenko <boris at tagnet.ru>
>>
>> Of course... my gw is looking in public context:
>> 2011-04-12 18:06:52.529284 [INFO] mod_dialplan_xml.c:331 Processing test
>> gw <12>->1234 in context public
>> 2011-04-12 18:06:52.530329 [INFO] mod_dialplan_xml.c:331 Processing test
>> gw <12>->ext_translate_extsrc in context features
>>
>> But why the extension I configured does not work for it?
>>
>> > Hello,
>> >
>> > 12.04.2011, 13:57, "Boris Kovalenko"<boris at tagnet.ru>;:
>> >
>> >> ? Hello!
>> >>
>> >> ? ? ? ?Hmm... I thinked GW is defined within profile not within
>> >> context???
>> > Pardon, of course you are right.. in general.
>> > I meant that context public linked to external profile by default if you
>> > didn't set it explicitly elsewhere.
>> > So if you defined your GW in internal profile it will search ext in
>> > default context, not public.
>> >
>> > btw, you can declare gw in directory and even in user record:
>> > http://wiki.freeswitch.org/wiki/Clarification:gateways
>> >
>> >>> ? ?Hello,
>> >>>
>> >>> ? ?12.04.2011, 08:55, "Boris Kovalenko"<boris at tagnet.ru>;;:
>> >>>> ? ?Hello!
>> >>>>
>> >>>> ? ? ? ? ?I have profile named ipbx with gateway defined:
>> >>>> ? ?<include>
>> >>>> ? ?<gateway name="test.tagnet.hn">
>> >>>> ? ?<!--/// account username *required* ///-->
>> >>>> ? ?<param name="username" value="test"/>
>> >>>> ? ?<!--/// auth realm: *optional* same as gateway name, if blank
>> >>>> ///-->
>> >>>> ? ?<param name="realm" value="public"/>
>> >>>> ? ?<!--/// username to use in from: *optional* same as ?username, if
>> >>>> blank
>> >>>> ? ?///-->
>> >>>> ? ?<param name="from-user" value="gw_test"/>
>> >>>> ? ?<!--/// account password *required* ///-->
>> >>>> ? ?<param name="password" value="2007"/>
>> >>>> ? ?<!--/// extension for inbound calls: *optional* same as username,
>> >>>> if
>> >>>> ? ?blank ///
>> >>>> ? ?-->
>> >>>> ? ?<param name="extension" value="gw_test"/>
>> >>>> ? ?There is an extension in context public:
>> >>>> ? ?<extension name="gw_test">
>> >>>> ? ?<condition field="destination_number" expression="^gw_test$">
>> >>>> ? ?<action application="info" data="ALERT"/>
>> >>>> ? ?<action application="hangup" data="NO_ROUTE_DESTINATION"/>
>> >>>> ? ?</condition>
>> >>>> ? ?</extension>
>> >>>>
>> >>>> ? ?So, my inbound calls from this gateway should go to extension
>> >>>> gw_test?
>> >>>> ? ?But they don't...
>> >>>> ? ?What is wrong with my config? FreeSWITCH Version 1.0.head
>> >>>> (git-1c95ad9
>> >>>> ? ?2011-01-20 22-43-50 -0300)
>> >>> ? ?Is this GW defined in public context?
>> >>> ? ?Hint: realm != context!
>> >>> ? ?Pls, look at example 'incoming.xml' in public profile in default
>> >>> config set
>> >> ? --
>> >> ? ? ?????????,
>> >> ? ? ????? ?????????
>> >> ? ? ??? "??????"
>> >> ? ? (3435) 494991
>> >>
>> >> ? _______________________________________________
>> >> ? FreeSWITCH-users mailing list
>> >> ? FreeSWITCH-users at lists.freeswitch.org
>> >> ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> ? http://www.freeswitch.org
>>
>>
>> --
>> ? ?????????,
>> ? ????? ?????????
>> ? ??? "??????"
>> ? (3435) 494991
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   (3435) 494991
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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


From boris at tagnet.ru  Tue Apr 12 18:04:56 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 12 Apr 2011 20:04:56 +0600
Subject: [Freeswitch-users] Still can't understand gateways
In-Reply-To: <BANLkTimXWpBUVaUS4MzvjfV+3Z3SjGw5vg@mail.gmail.com>
References: <4DA3DB27.4010205@tagnet.ru>	<1091151302601933@web154.yandex.ru>	<4DA421F6.5070504@tagnet.ru>	<6481302607664@web133.yandex.ru>	<4DA44097.1000403@tagnet.ru>	<BANLkTin4xOx3_5ymnt048wM2ygT2cp1pCg@mail.gmail.com>	<4DA455BD.8020206@tagnet.ru>
	<BANLkTimXWpBUVaUS4MzvjfV+3Z3SjGw5vg@mail.gmail.com>
Message-ID: <4DA45C08.50303@tagnet.ru>

Hello!

     Ough... my misundestanding. I thinked this is extension in the 
dialplan where incoming calls are placed.

> This parameter specifies the value FreeSWITCH is going to use for the
> username portion of the Contact URI when registering to your gateway:
>
> extension = cluecon
>
> FreeSWITCH sends a REGISTER packet with a contact address like this:
>
> Contact: cluecon at your.ip.address
>
> With a "standard" setup your provider is supposed to send calls to the
> value provided in the Contact address with INVITEs to
> cluecon at your.ip.address.  However, many providers (to implement DID
> services) ignore the username portion and just use the host portion
> (IP address) so what you end up with is an INVITE to
> 8005551212 at your.ip.address.
>
> If I saw a SIP trace I'd know for sure but that's probably what's going on.
>
> On Tue, Apr 12, 2011 at 9:38 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>>      But.... reading the docs:
>>
>> <!--/// extension for inbound calls: *optional* same as username, if blank
>> ///-->
>> <!--<param name="extension" value="cluecon"/>-->
>>
>> What this parameter means? I thinked that if extension is specified all
>> incoming calls are placed to this extension. Isn't?
>>
>> Because you're matching the destination_number "gw_test". The logs show
>> you're dialing 1234 and then ext_translate_extsrc though, which don't match
>> that condition.
>>
>> -Steve
>>
>>
>> 2011/4/12 Boris Kovalenko<boris at tagnet.ru>
>>> Of course... my gw is looking in public context:
>>> 2011-04-12 18:06:52.529284 [INFO] mod_dialplan_xml.c:331 Processing test
>>> gw<12>->1234 in context public
>>> 2011-04-12 18:06:52.530329 [INFO] mod_dialplan_xml.c:331 Processing test
>>> gw<12>->ext_translate_extsrc in context features
>>>
>>> But why the extension I configured does not work for it?
>>>
>>>> Hello,
>>>>
>>>> 12.04.2011, 13:57, "Boris Kovalenko"<boris at tagnet.ru>;:
>>>>
>>>>>    Hello!
>>>>>
>>>>>         Hmm... I thinked GW is defined within profile not within
>>>>> context???
>>>> Pardon, of course you are right.. in general.
>>>> I meant that context public linked to external profile by default if you
>>>> didn't set it explicitly elsewhere.
>>>> So if you defined your GW in internal profile it will search ext in
>>>> default context, not public.
>>>>
>>>> btw, you can declare gw in directory and even in user record:
>>>> http://wiki.freeswitch.org/wiki/Clarification:gateways
>>>>
>>>>>>     Hello,
>>>>>>
>>>>>>     12.04.2011, 08:55, "Boris Kovalenko"<boris at tagnet.ru>;;:
>>>>>>>     Hello!
>>>>>>>
>>>>>>>           I have profile named ipbx with gateway defined:
>>>>>>>     <include>
>>>>>>>     <gateway name="test.tagnet.hn">
>>>>>>>     <!--/// account username *required* ///-->
>>>>>>>     <param name="username" value="test"/>
>>>>>>>     <!--/// auth realm: *optional* same as gateway name, if blank
>>>>>>> ///-->
>>>>>>>     <param name="realm" value="public"/>
>>>>>>>     <!--/// username to use in from: *optional* same as  username, if
>>>>>>> blank
>>>>>>>     ///-->
>>>>>>>     <param name="from-user" value="gw_test"/>
>>>>>>>     <!--/// account password *required* ///-->
>>>>>>>     <param name="password" value="2007"/>
>>>>>>>     <!--/// extension for inbound calls: *optional* same as username,
>>>>>>> if
>>>>>>>     blank ///
>>>>>>>     -->
>>>>>>>     <param name="extension" value="gw_test"/>
>>>>>>>     There is an extension in context public:
>>>>>>>     <extension name="gw_test">
>>>>>>>     <condition field="destination_number" expression="^gw_test$">
>>>>>>>     <action application="info" data="ALERT"/>
>>>>>>>     <action application="hangup" data="NO_ROUTE_DESTINATION"/>
>>>>>>>     </condition>
>>>>>>>     </extension>
>>>>>>>
>>>>>>>     So, my inbound calls from this gateway should go to extension
>>>>>>> gw_test?
>>>>>>>     But they don't...
>>>>>>>     What is wrong with my config? FreeSWITCH Version 1.0.head
>>>>>>> (git-1c95ad9
>>>>>>>     2011-01-20 22-43-50 -0300)
>>>>>>     Is this GW defined in public context?
>>>>>>     Hint: realm != context!
>>>>>>     Pls, look at example 'incoming.xml' in public profile in default
>>>>>> config set
>>>>>    --
>>>>>    ? ?????????,
>>>>>      ????? ?????????
>>>>>      ??? "??????"
>>>>>      (3435) 494991
>>>>>
>>>>>    _______________________________________________
>>>>>    FreeSWITCH-users mailing list
>>>>>    FreeSWITCH-users at lists.freeswitch.org
>>>>>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>    http://www.freeswitch.org
>>>
>>> --
>>> ? ?????????,
>>>    ????? ?????????
>>>    ??? "??????"
>>>    (3435) 494991
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> --
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    (3435) 494991
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   (3435) 494991




From u2nsam at gmail.com  Tue Apr 12 19:04:09 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 12 Apr 2011 20:34:09 +0530
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <1B19ABD72889C245AE8EEE08AC24103A28C423231C@exmachina.office.kapper.net>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A28C423231C@exmachina.office.kapper.net>
Message-ID: <BANLkTimrvA=Lgva-FAdXxWRhS-_UDuTgTA@mail.gmail.com>

I have done that, but i want to pass the exact SDP what i get from leg A to
leg B

regards
Sam

On Tue, Apr 12, 2011 at 4:48 PM, Clemens Ebentheuer <ce at kapper.net> wrote:

> http://wiki.freeswitch.org/wiki/Proxy_media#How_to_enable_it
>
>
>
> ce
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sam
> *Sent:* Tuesday, April 12, 2011 11:34 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] proxy SDP
>
>
>
> Hi all,
>
>
> Is there method to just proxy SDP through Freeswitch through sip profile ?
>
>
>
> Regards
> Sam
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From steveayre at gmail.com  Tue Apr 12 19:11:26 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 12 Apr 2011 16:11:26 +0100
Subject: [Freeswitch-users] Freeswitch Server Down
In-Reply-To: <BANLkTikmZfQjM3wEiVYY+2xuQruCmmaUjQ@mail.gmail.com>
References: <BANLkTimerjRCutH8LFaRjvCfSju6byO1ow@mail.gmail.com>
	<BANLkTikLGEqSnL7yYnvVmBE6AvYgCBTXmg@mail.gmail.com>
	<637F58A1-8368-4E26-A192-3629D9BDAF8F@gmail.com>
	<BANLkTikmZfQjM3wEiVYY+2xuQruCmmaUjQ@mail.gmail.com>
Message-ID: <BANLkTi=C=cjknY7ckpxkxVrK-Ci0ZgAqrQ@mail.gmail.com>

wp_tdmapi_read_msg:1296 User API Error: User Rx Len=1064 < Driver Rx
Len=4150 (hdr=64). User API must increase expected rx length!
freeswitch[6072] general protection rip:2aaab4b3b894 rsp:419fac38 error:0

The 2nd line means freeswitch crashed because it tried to access a piece of
memory that wasn't its own. That is always a bug. It may be related to the
previous line, which suggests that it's a wanpipe problem.

Can you find a coredump anywhere? It'd be called core.6072 (i.e. core.PID)

What version are you running? You should try upgrading if you're on an old
version as it's possible it's something that's already fixed.

-Steve



On 12 April 2011 12:04, ovvenkat <ovvenkatesan at gmail.com> wrote:

> Hi Steven,
>
> Thanks for your response.
> Yes, Its on linux machine and
> Since, I am new to linux platform I could
> not able to understand the log file.
> please find the attachment of *dmesg_messages*
>
> I am getting error like
>
> wp_tdmapi_read_msg:1296 User API Error: User Rx Len=1064 < Driver Rx
> Len=1567 (hdr=64). User API must increase expected rx length!
>
> I dont know, what is this mean.
>
> Can you guide me please, what is went wrong and how to avoid in future?
>
>
> Regards,
> Venkat.
>
>
>
>
> On Tue, Apr 12, 2011 at 4:00 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Are you on Linux? If so run dmesg and see if there are any messages
>> indicating freeswitch had a segmentation fault or general protection fault.
>> If there is it's a bug and there will hopefully be a coredump file that will
>> contain useful information for tracking the problem down.
>>
>> Steve on iPhone
>>
>> On 11 Apr 2011, at 18:49, Michael Collins <msc at freeswitch.org> wrote:
>>
>>
>>
>> On Mon, Apr 11, 2011 at 10:32 AM, ovvenkat < <ovvenkatesan at gmail.com>
>> ovvenkatesan at gmail.com> wrote:
>>
>>> Hi to all,
>>>
>>> Today, All my IVR are stopped working.
>>> When I check the freeSwitch it was down.
>>> I dont know the reason why Its Down.
>>> How I can find the reason,  why my freeSwitch went down?
>>>
>>
>> You'll need to check logs to track down what happened. I know that
>> sometimes you'll see lots of log lines in freeswitch.log and then all of a
>> sudden nothing, so that will help you pinpoint when things went wrong.
>> Possibly you'll see some log lines with errors or warnings. Or you might see
>> that the "fsctl shutdown" command was sent to it. (Then you'll need to go
>> hunt down whoever did that. :)
>>
>> -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
>>
>>
>
>
> --
>
> If you have come to help me, you are wasting your time.
> If you have come to because your liberation is bound up in mine, we can
> work together.
>
>
> Regards
> Venkatesan OV.
>
>
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From anthony.minessale at gmail.com  Tue Apr 12 19:36:04 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Apr 2011 10:36:04 -0500
Subject: [Freeswitch-users] Passing SIP headers from b-leg to a-leg
In-Reply-To: <BANLkTimZjq1EffupZKV1ais=zkx9R+D_WQ@mail.gmail.com>
References: <BANLkTimZjq1EffupZKV1ais=zkx9R+D_WQ@mail.gmail.com>
Message-ID: <BANLkTinKY-=4U8UVnSBykcvkfAQ9huZ45g@mail.gmail.com>

assuming you are on a more modern release:

add {sip_copy_custom_headers=true} to the dial string of the b leg.




On Tue, Apr 12, 2011 at 6:02 AM, Javier Gallart <jgallartm at gmail.com> wrote:
> Hi all
> I'm trying to pass some custom X-headers in final failed responses (>400)
> ?from B-leg to A-leg...is there a way to accomplish this?
> 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
>
>



-- 
Anthony Minessale II

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FreeSWITCH Developer Conference
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pstn:+19193869900


From steveayre at gmail.com  Tue Apr 12 20:46:26 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 12 Apr 2011 17:46:26 +0100
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <BANLkTimrvA=Lgva-FAdXxWRhS-_UDuTgTA@mail.gmail.com>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A28C423231C@exmachina.office.kapper.net>
	<BANLkTimrvA=Lgva-FAdXxWRhS-_UDuTgTA@mail.gmail.com>
Message-ID: <BANLkTimFeP+r2uNk09BbVw7hAULf1MYvzA@mail.gmail.com>

Can you be more exact about what in the SDP you want to send across
directly?

If media is going through FS you can't - the SDP contains the IP and port
numbers for the RTP streams, so if FS is in the media path it must change
that part of the SDP.

Using bypass_media will probably keep the SDP completely intact, with the
media going directly between the endpoints. That can be a problem if the
endpoints can't see each other directly though.

-Steve


On 12 April 2011 16:04, Sam <u2nsam at gmail.com> wrote:

> I have done that, but i want to pass the exact SDP what i get from leg A to
> leg B
>
> regards
> Sam
>
> On Tue, Apr 12, 2011 at 4:48 PM, Clemens Ebentheuer <ce at kapper.net> wrote:
>
>> http://wiki.freeswitch.org/wiki/Proxy_media#How_to_enable_it
>>
>>
>>
>> ce
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sam
>> *Sent:* Tuesday, April 12, 2011 11:34 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] proxy SDP
>>
>>
>>
>> Hi all,
>>
>>
>> Is there method to just proxy SDP through Freeswitch through sip profile ?
>>
>>
>>
>> Regards
>> Sam
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gourav at rentec.com  Tue Apr 12 20:48:55 2011
From: gourav at rentec.com (Gourav Vohra)
Date: Tue, 12 Apr 2011 12:48:55 -0400 (EDT)
Subject: [Freeswitch-users] Shared Call appearence, barging and presence
In-Reply-To: <828065258.52145.1301433563225.JavaMail.root@zinnia1>
Message-ID: <1429335029.215077.1302626935413.JavaMail.root@zinnia1>



Thanks in advance with helping me with this.    

I am having some problems with sla.  

My setup includes polycom IP 650 phones (SIP version 3.3.1) and freeswitch downloaded on Apr 3 from the following link.
http://files.freeswitch.org/freeswitch-snapshot.tar.gz

Following is what my setup looks like:
phone1 - x2908
phone2 - x2995
phone3 - x2996, x2995

In my test I make a call from phone1 to x2995 and pick it up on phone2.  At this point I see the x2995's line in use on phone3.  Next I barge into the call from phone3.  At this point phone1, phone2 and phone3 are all on the call that was initiated from phone1.

Next I end the call on phone2.

The issue I am having is that after I barge in from phone3 and "End Call" on phone2 - The call remains established between phone 3 and phone1 but x2995 on phone2 does not show that the line is in use.  I believe that the call should remain established between phone1 and phone3 after phone2 drops out and the line appearance (x2995) on phone2 should look like it's still in use.  The led on the polycom 650 should change to red.  In my case it doesn't.   

On the polycom config x2995 is setup as a shared line with reg.x.bargeInEnabled set to "1".

Following is set on vars.xml.
  <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/>
  <X-PRE-PROCESS cmd="set" data="domain_name=$${domain}"/>

Following is set on the sip profile.
    <!--enable to use presence -->
    <param name="manage-presence" value="true"/>
    <!-- send a presence probe on each register to query devices to send presence instead of sending presence with less info -->
    <param name="presence-probe-on-register" value="true"/>-->
    <param name="manage-shared-appearance" value="true"/>
    <!-- used to share presence info across sofia profiles -->
    <!-- Name of the db to use for this profile -->
    <param name="dbname" value="share_presence"/>
    <param name="presence-hosts" value="$${domain},$${local_ip_v4}"/>

Following is set on the user registration.
      <param name="manage-presence" value="passive"/>
      <param name="manage-shared-appearance" value="true"/

Following is set on the dialplan for x2995 and x2996.
      <action application="bridge" data="{sip_invite_domain=$${domain_name}}user/2995@${domain}"/>


Following logs are for the call getting barged in from phone3 and getting dropped from phone2.

2011-04-08 10:08:37.557009 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/log/cdr-csv/2908.csv
2011-04-08 10:08:37.557009 [NOTICE] mod_logfile.c:158 New log started.
2011-04-08 10:09:04.414661 [DEBUG] sofia.c:6539 IP 192.168.100.75 Rejected by acl "domains". Falling back to Digest auth.
2011-04-08 10:09:04.414661 [WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile 'internal' for [2995 at 192.168.100.33] from ip 192.168.100.75
2011-04-08 10:09:04.428752 [DEBUG] sofia.c:6539 IP 192.168.100.75 Rejected by acl "domains". Falling back to Digest auth.
2011-04-08 10:09:04.428752 [NOTICE] switch_channel.c:812 New Channel sofia/internal/2995 at 192.168.100.33 [818ddd7f-99e2-4d22-b7ae-e62b6fdbfda4]
2011-04-08 10:09:04.429760 [DEBUG] switch_ivr.c:1600 (sofia/internal/sip:2995 at 192.168.100.74) State Change CS_EXCHANGE_MEDIA -> CS_ROUTING
2011-04-08 10:09:04.429760 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:2995 at 192.168.100.74 [BREAK]
2011-04-08 10:09:04.429760 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/sip:2995 at 192.168.100.74 [BREAK]
2011-04-08 10:09:04.429760 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/sip:2995 at 192.168.100.74 to inline[answer,conference:6c33a351-1842-4e45-9bfb-89249c44a8c6 at sla+flags{mintwo}@default]
2011-04-08 10:09:04.429760 [DEBUG] switch_ivr.c:1600 (sofia/internal/2908 at 192.168.100.33) State Change CS_EXECUTE -> CS_ROUTING
2011-04-08 10:09:04.429760 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/2908 at 192.168.100.33 [BREAK]
2011-04-08 10:09:04.429760 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/2908 at 192.168.100.33 [BREAK]
2011-04-08 10:09:04.429760 [NOTICE] switch_ivr.c:1606 Transfer sofia/internal/2908 at 192.168.100.33 to inline[answer,conference:6c33a351-1842-4e45-9bfb-89249c44a8c6 at sla+flags{mintwo}@default]
2011-04-08 10:09:04.429760 [DEBUG] sofia.c:4760 Channel sofia/internal/2995 at 192.168.100.33 entering state [received][100]
2011-04-08 10:09:04.429760 [DEBUG] sofia.c:4771 Remote SDP:
v=0
o=- 1302271492 1302271492 IN IP4 192.168.100.75
s=Polycom IP Phone
c=IN IP4 192.168.100.75
t=0 0
a=sendrecv
m=audio 2234 RTP/AVP 9 0 8 18 127
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:127 telephone-event/8000

2011-04-08 10:09:04.429760 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [G722:9:8000:20:64000]/[G7221:115:32000:20:48000]
2011-04-08 10:09:04.429760 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [G722:9:8000:20:64000]/[G7221:107:16000:20:32000]
2011-04-08 10:09:04.429760 [DEBUG] sofia_glue.c:4637 Audio Codec Compare [G722:9:8000:20:64000]/[G722:9:8000:20:64000]
2011-04-08 10:09:04.429760 [DEBUG] sofia_glue.c:2760 Set Codec sofia/internal/2995 at 192.168.100.33 G722/8000 20 ms 160 samples 64000 bits
2011-04-08 10:09:04.429760 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/2995 at 192.168.100.33) Running State Change CS_NEW
2011-04-08 10:09:04.429760 [DEBUG] switch_core_state_machine.c:343 (sofia/internal/2995 at 192.168.100.33) State NEW
2011-04-08 10:09:04.430787 [DEBUG] sofia_glue.c:4751 Set 2833 dtmf send/recv payload to 127
2011-04-08 10:09:04.430787 [DEBUG] sofia.c:4942 (sofia/internal/2995 at 192.168.100.33) State Change CS_NEW -> CS_INIT
2011-04-08 10:09:04.430787 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/2995 at 192.168.100.33 [BREAK]
2011-04-08 10:09:04.430787 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/2995 at 192.168.100.33) Running State Change CS_INIT
2011-04-08 10:09:04.430787 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/2995 at 192.168.100.33) State INIT
2011-04-08 10:09:04.430787 [DEBUG] mod_sofia.c:84 sofia/internal/2995 at 192.168.100.33 SOFIA INIT
2011-04-08 10:09:04.430787 [DEBUG] mod_sofia.c:124 (sofia/internal/2995 at 192.168.100.33) State Change CS_INIT -> CS_ROUTING
2011-04-08 10:09:04.430787 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/2995 at 192.168.100.33 [BREAK]
2011-04-08 10:09:04.430787 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/2995 at 192.168.100.33) State INIT going to sleep
2011-04-08 10:09:04.430787 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/2995 at 192.168.100.33) Running State Change CS_ROUTING
2011-04-08 10:09:04.431879 [DEBUG] switch_channel.c:1668 (sofia/internal/2995 at 192.168.100.33) Callstate Change DOWN -> RINGING
2011-04-08 10:09:04.431879 [DEBUG] switch_ivr_bridge.c:582 BRIDGE THREAD DONE [sofia/internal/2908 at 192.168.100.33]
2011-04-08 10:09:04.431879 [DEBUG] switch_ivr_bridge.c:602 Send signal sofia/internal/sip:2995 at 192.168.100.74 [BREAK]
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/2995 at 192.168.100.33) State ROUTING
2011-04-08 10:09:04.431879 [DEBUG] mod_sofia.c:147 sofia/internal/2995 at 192.168.100.33 SOFIA ROUTING
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:77 sofia/internal/2995 at 192.168.100.33 Standard ROUTING
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/2995 at 192.168.100.33) State Change CS_ROUTING -> CS_EXECUTE
2011-04-08 10:09:04.431879 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/2995 at 192.168.100.33 [BREAK]
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/2995 at 192.168.100.33) State ROUTING going to sleep
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/2995 at 192.168.100.33) Running State Change CS_EXECUTE
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/2995 at 192.168.100.33) State EXECUTE
2011-04-08 10:09:04.431879 [DEBUG] mod_sofia.c:240 sofia/internal/2995 at 192.168.100.33 SOFIA EXECUTE
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:157 sofia/internal/2995 at 192.168.100.33 Standard EXECUTE
EXECUTE sofia/internal/2995 at 192.168.100.33 answer()
2011-04-08 10:09:04.431879 [DEBUG] switch_ivr_bridge.c:582 BRIDGE THREAD DONE [sofia/internal/sip:2995 at 192.168.100.74]
2011-04-08 10:09:04.431879 [DEBUG] switch_ivr_bridge.c:602 Send signal sofia/internal/2908 at 192.168.100.33 [BREAK]
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:374 (sofia/internal/sip:2995 at 192.168.100.74) State EXCHANGE_MEDIA going to sleep
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:2995 at 192.168.100.74) Running State Change CS_ROUTING
2011-04-08 10:09:04.431879 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/sip:2995 at 192.168.100.74 [BREAK]
2011-04-08 10:09:04.431879 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/2908 at 192.168.100.33 [BREAK]
2011-04-08 10:09:04.431879 [DEBUG] switch_channel.c:1668 (sofia/internal/sip:2995 at 192.168.100.74) Callstate Change ACTIVE -> RINGING
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/2908 at 192.168.100.33) State EXECUTE going to sleep
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/2908 at 192.168.100.33) Running State Change CS_ROUTING
2011-04-08 10:09:04.431879 [DEBUG] switch_channel.c:1668 (sofia/internal/2908 at 192.168.100.33) Callstate Change ACTIVE -> RINGING
2011-04-08 10:09:04.431879 [DEBUG] sofia_glue.c:3001 AUDIO RTP [sofia/internal/2995 at 192.168.100.33] 192.168.100.33 port 29998 -> 192.168.100.75 port 2234 codec: 9 ms: 20
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/2908 at 192.168.100.33) State ROUTING
2011-04-08 10:09:04.431879 [DEBUG] mod_sofia.c:147 sofia/internal/2908 at 192.168.100.33 SOFIA ROUTING
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:77 sofia/internal/2908 at 192.168.100.33 Standard ROUTING
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/2908 at 192.168.100.33) State Change CS_ROUTING -> CS_EXECUTE
2011-04-08 10:09:04.431879 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/2908 at 192.168.100.33 [BREAK]
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/2908 at 192.168.100.33) State ROUTING going to sleep
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/2908 at 192.168.100.33) Running State Change CS_EXECUTE
2011-04-08 10:09:04.431879 [DEBUG] switch_channel.c:1670 (sofia/internal/2908 at 192.168.100.33) Callstate Change RINGING -> ACTIVE
2011-04-08 10:09:04.431879 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/2908 at 192.168.100.33) State EXECUTE
2011-04-08 10:09:04.431879 [DEBUG] mod_sofia.c:240 sofia/internal/2908 at 192.168.100.33 SOFIA EXECUTE
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:157 sofia/internal/2908 at 192.168.100.33 Standard EXECUTE
EXECUTE sofia/internal/2908 at 192.168.100.33 answer()
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/sip:2995 at 192.168.100.74) State ROUTING
2011-04-08 10:09:04.431879 [DEBUG] mod_sofia.c:147 sofia/internal/sip:2995 at 192.168.100.74 SOFIA ROUTING
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:77 sofia/internal/sip:2995 at 192.168.100.74 Standard ROUTING
2011-04-08 10:09:04.431879 [INFO] switch_channel.c:2457 sofia/internal/sip:2995 at 192.168.100.74 Flipping CID from "Gourav Vohra" <2908> to "Outbound Call" <2995>
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/sip:2995 at 192.168.100.74) State Change CS_ROUTING -> CS_EXECUTE
2011-04-08 10:09:04.431879 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/sip:2995 at 192.168.100.74 [BREAK]
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/sip:2995 at 192.168.100.74) State ROUTING going to sleep
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/sip:2995 at 192.168.100.74) Running State Change CS_EXECUTE
2011-04-08 10:09:04.431879 [DEBUG] switch_channel.c:1670 (sofia/internal/sip:2995 at 192.168.100.74) Callstate Change RINGING -> ACTIVE
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/sip:2995 at 192.168.100.74) State EXECUTE
2011-04-08 10:09:04.431879 [DEBUG] mod_sofia.c:240 sofia/internal/sip:2995 at 192.168.100.74 SOFIA EXECUTE
2011-04-08 10:09:04.431879 [DEBUG] switch_core_state_machine.c:157 sofia/internal/sip:2995 at 192.168.100.74 Standard EXECUTE
EXECUTE sofia/internal/sip:2995 at 192.168.100.74 answer()
EXECUTE sofia/internal/2908 at 192.168.100.33 conference(6c33a351-1842-4e45-9bfb-89249c44a8c6 at sla+flags{mintwo})
EXECUTE sofia/internal/sip:2995 at 192.168.100.74 conference(6c33a351-1842-4e45-9bfb-89249c44a8c6 at sla+flags{mintwo})
2011-04-08 10:09:04.433706 [INFO] mod_conference.c:6496 using channel sound prefix: /usr/local/freeswitch/sounds/en/us/callie
2011-04-08 10:09:04.433706 [DEBUG] mod_conference.c:5464 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
2011-04-08 10:09:04.433706 [DEBUG] mod_conference.c:5509 Raw Codec Activation Success L16 at 16000hz 1 channel 20ms
2011-04-08 10:09:04.433706 [DEBUG] switch_core_codec.c:116 sofia/internal/sip:2995 at 192.168.100.74 Push codec L16:70
2011-04-08 10:09:04.433706 [DEBUG] mod_conference.c:1069 Setup timer success interval: 20  samples: 320
2011-04-08 10:09:04.433706 [DEBUG] sofia_glue.c:3263 Set 2833 dtmf send payload to 127
2011-04-08 10:09:04.433706 [DEBUG] sofia_glue.c:3268 Set 2833 dtmf receive payload to 127
2011-04-08 10:09:04.433706 [DEBUG] mod_sofia.c:681 Local SDP sofia/internal/2995 at 192.168.100.33:
v=0
o=FreeSWITCH 1302241746 1302241747 IN IP4 192.168.100.33
s=FreeSWITCH
c=IN IP4 192.168.100.33
t=0 0
m=audio 29998 RTP/AVP 9 127
a=rtpmap:9 G722/8000
a=rtpmap:127 telephone-event/8000
a=fmtp:127 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2011-04-08 10:09:04.433706 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/2995 at 192.168.100.33 [BREAK]
2011-04-08 10:09:04.433706 [DEBUG] switch_channel.c:2821 (sofia/internal/2995 at 192.168.100.33) Callstate Change RINGING -> ACTIVE
2011-04-08 10:09:04.433706 [NOTICE] mod_dptools.c:930 Channel [sofia/internal/2995 at 192.168.100.33] has been answered
2011-04-08 10:09:04.433706 [DEBUG] mod_conference.c:5464 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms
2011-04-08 10:09:04.433706 [DEBUG] mod_conference.c:5509 Raw Codec Activation Success L16 at 16000hz 1 channel 20ms
2011-04-08 10:09:04.433706 [DEBUG] switch_core_codec.c:116 sofia/internal/2908 at 192.168.100.33 Push codec L16:70
2011-04-08 10:09:04.435298 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/sip:2995 at 192.168.100.74 [BREAK]
2011-04-08 10:09:04.435298 [DEBUG] mod_conference.c:2552 Setup timer soft success interval: 20  samples: 160
2011-04-08 10:09:04.435298 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/2908 at 192.168.100.33 [BREAK]
2011-04-08 10:09:04.435298 [DEBUG] mod_conference.c:2552 Setup timer soft success interval: 20  samples: 160
2011-04-08 10:09:04.435298 [DEBUG] sofia.c:4760 Channel sofia/internal/2995 at 192.168.100.33 entering state [completed][200]
EXECUTE sofia/internal/2995 at 192.168.100.33 conference(6c33a351-1842-4e45-9bfb-89249c44a8c6 at sla+flags{mintwo})
2011-04-08 10:09:04.435298 [DEBUG] mod_conference.c:5464 Raw Codec Activation Success L16 at 16000hz 1 channel 20ms
2011-04-08 10:09:04.435298 [DEBUG] mod_conference.c:5509 Raw Codec Activation Success L16 at 16000hz 1 channel 20ms
2011-04-08 10:09:04.436366 [DEBUG] switch_core_codec.c:116 sofia/internal/2995 at 192.168.100.33 Push codec L16:70
2011-04-08 10:09:04.436366 [DEBUG] switch_core_session.c:709 Send signal sofia/internal/2995 at 192.168.100.33 [BREAK]
2011-04-08 10:09:04.436366 [DEBUG] mod_conference.c:2552 Setup timer soft success interval: 20  samples: 160
2011-04-08 10:09:04.441402 [DEBUG] sofia.c:4760 Channel sofia/internal/2995 at 192.168.100.33 entering state [ready][200]
2011-04-08 10:09:04.511924 [DEBUG] switch_rtp.c:3082 Correct ip/port confirmed.
2011-04-08 10:09:04.526038 [WARNING] sofia_presence.c:781 external is passive, skipping
2011-04-08 10:09:04.527046 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:04.527046 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='2995' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal-ipv6' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:04]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:04 GMT]
Event-Date-Timestamp: [1302271744430787]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [585]
Channel-State: [CS_ROUTING]
Channel-Call-State: [DOWN]
Channel-State-Number: [2]
Channel-Name: [sofia/internal/2995 at 192.168.100.33]
Unique-ID: [818ddd7f-99e2-4d22-b7ae-e62b6fdbfda4]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-Presence-ID: [2995 at 192.168.100.33]
Answer-State: [ringing]
Channel-Read-Codec-Name: [G722]
Channel-Read-Codec-Rate: [16000]
Channel-Read-Codec-Bit-Rate: [64000]
Channel-Write-Codec-Name: [G722]
Channel-Write-Codec-Rate: [16000]
Channel-Write-Codec-Bit-Rate: [64000]
Caller-Direction: [inbound]
Caller-Username: [2995]
Caller-Dialplan: [inline]
Caller-Caller-ID-Name: [Gourav Vohra]
Caller-Caller-ID-Number: [2995]
Caller-Network-Addr: [192.168.100.75]
Caller-ANI: [2995]
Caller-Destination-Number: [answer,conference:6c33a351-1842-4e45-9bfb-89249c44a8c6 at sla+flags{mintwo}]
Caller-Unique-ID: [818ddd7f-99e2-4d22-b7ae-e62b6fdbfda4]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/2995 at 192.168.100.33]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1302271744429760]
Caller-Channel-Created-Time: [1302271744429760]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [src/switch_channel.c]
login: [src/switch_channel.c]
from: [2995 at 192.168.100.33]
rpid: [unknown]
status: [CS_ROUTING]
event_type: [presence]
alt_event_type: [dialog]
presence-call-info-state: [alerting]
presence-call-info: [appearance-index=1]
presence-call-direction: [inbound]
event_count: [0]
Presence-Calling-File: [src/switch_channel.c]
Presence-Calling-Function: [switch_channel_perform_set_running_state]
Presence-Calling-Line: [1660]


2011-04-08 10:09:04.527046 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:04.528054 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal)
2011-04-08 10:09:04.529062 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '2995 at 192.168.100.33' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='2995' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:04]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:04 GMT]
Event-Date-Timestamp: [1302271744430787]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [585]
Channel-State: [CS_ROUTING]
Channel-Call-State: [DOWN]
Channel-State-Number: [2]
Channel-Name: [sofia/internal/2995 at 192.168.100.33]
Unique-ID: [818ddd7f-99e2-4d22-b7ae-e62b6fdbfda4]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-Presence-ID: [2995 at 192.168.100.33]
Answer-State: [ringing]
Channel-Read-Codec-Name: [G722]
Channel-Read-Codec-Rate: [16000]
Channel-Read-Codec-Bit-Rate: [64000]
Channel-Write-Codec-Name: [G722]
Channel-Write-Codec-Rate: [16000]
Channel-Write-Codec-Bit-Rate: [64000]
Caller-Direction: [inbound]
Caller-Username: [2995]
Caller-Dialplan: [inline]
Caller-Caller-ID-Name: [Gourav Vohra]
Caller-Caller-ID-Number: [2995]
Caller-Network-Addr: [192.168.100.75]
Caller-ANI: [2995]
Caller-Destination-Number: [answer,conference:6c33a351-1842-4e45-9bfb-89249c44a8c6 at sla+flags{mintwo}]
Caller-Unique-ID: [818ddd7f-99e2-4d22-b7ae-e62b6fdbfda4]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/2995 at 192.168.100.33]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1302271744429760]
Caller-Channel-Created-Time: [1302271744429760]
Caller-Channel-Answered-Time: [0]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [src/switch_channel.c]
login: [src/switch_channel.c]
from: [2995 at 192.168.100.33]
rpid: [unknown]
status: [CS_ROUTING]
event_type: [presence]
alt_event_type: [dialog]
presence-call-info-state: [alerting]
presence-call-info: [appearance-index=1]
presence-call-direction: [inbound]
event_count: [0]
Presence-Calling-File: [src/switch_channel.c]
Presence-Calling-Function: [switch_channel_perform_set_running_state]
Presence-Calling-Line: [1660]


2011-04-08 10:09:04.529062 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal)
2011-04-08 10:09:04.529062 [WARNING] sofia_presence.c:774 192.168.100.33 is an alias, skipping
2011-04-08 10:09:04.529062 [WARNING] sofia_presence.c:781 external is passive, skipping
2011-04-08 10:09:04.529062 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:04.529062 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='2995' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal-ipv6' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:04]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:04 GMT]
Event-Date-Timestamp: [1302271744431879]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [585]
Channel-State: [CS_ROUTING]
Channel-Call-State: [ACTIVE]
Channel-State-Number: [2]
Channel-Name: [sofia/internal/sip:2995 at 192.168.100.74]
Unique-ID: [e508f89d-e49b-49a7-ba5b-03c822ebe75f]
Call-Direction: [outbound]
Presence-Call-Direction: [outbound]
Channel-Presence-ID: [2995 at 192.168.100.33]
Channel-Call-UUID: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
Answer-State: [answered]
Channel-Read-Codec-Name: [PCMU]
Channel-Read-Codec-Rate: [8000]
Channel-Read-Codec-Bit-Rate: [64000]
Channel-Write-Codec-Name: [PCMU]
Channel-Write-Codec-Rate: [8000]
Channel-Write-Codec-Bit-Rate: [64000]
Caller-Direction: [outbound]
Caller-Username: [2908]
Caller-Dialplan: [inline]
Caller-Caller-ID-Name: [Gourav Vohra]
Caller-Caller-ID-Number: [2908]
Caller-Callee-ID-Name: [Outbound Call]
Caller-Callee-ID-Number: [2995]
Caller-Network-Addr: [192.168.100.74]
Caller-ANI: [2908]
Caller-Destination-Number: [answer,conference:6c33a351-1842-4e45-9bfb-89249c44a8c6 at sla+flags{mintwo}]
Caller-Unique-ID: [e508f89d-e49b-49a7-ba5b-03c822ebe75f]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-RDNIS: [2995]
Caller-Channel-Name: [sofia/internal/sip:2995 at 192.168.100.74]
Caller-Profile-Index: [2]
Caller-Profile-Created-Time: [1302271744429760]
Caller-Channel-Created-Time: [1302271711758979]
Caller-Channel-Answered-Time: [1302271714953388]
Caller-Channel-Progress-Time: [1302271711821261]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [src/switch_channel.c]
login: [src/switch_channel.c]
from: [2995 at 192.168.100.33]
rpid: [unknown]
status: [CS_ROUTING]
event_type: [presence]
alt_event_type: [dialog]
presence-call-direction: [outbound]
event_count: [2]
Presence-Calling-File: [src/switch_channel.c]
Presence-Calling-Function: [switch_channel_perform_set_running_state]
Presence-Calling-Line: [1660]


2011-04-08 10:09:04.530069 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:04.530069 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal)
2011-04-08 10:09:04.530069 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '2995 at 192.168.100.33' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='2995' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:04]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:04 GMT]
Event-Date-Timestamp: [1302271744431879]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [585]
Channel-State: [CS_ROUTING]
Channel-Call-State: [ACTIVE]
Channel-State-Number: [2]
Channel-Name: [sofia/internal/sip:2995 at 192.168.100.74]
Unique-ID: [e508f89d-e49b-49a7-ba5b-03c822ebe75f]
Call-Direction: [outbound]
Presence-Call-Direction: [outbound]
Channel-Presence-ID: [2995 at 192.168.100.33]
Channel-Call-UUID: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
Answer-State: [answered]
Channel-Read-Codec-Name: [PCMU]
Channel-Read-Codec-Rate: [8000]
Channel-Read-Codec-Bit-Rate: [64000]
Channel-Write-Codec-Name: [PCMU]
Channel-Write-Codec-Rate: [8000]
Channel-Write-Codec-Bit-Rate: [64000]
Caller-Direction: [outbound]
Caller-Username: [2908]
Caller-Dialplan: [inline]
Caller-Caller-ID-Name: [Gourav Vohra]
Caller-Caller-ID-Number: [2908]
Caller-Callee-ID-Name: [Outbound Call]
Caller-Callee-ID-Number: [2995]
Caller-Network-Addr: [192.168.100.74]
Caller-ANI: [2908]
Caller-Destination-Number: [answer,conference:6c33a351-1842-4e45-9bfb-89249c44a8c6 at sla+flags{mintwo}]
Caller-Unique-ID: [e508f89d-e49b-49a7-ba5b-03c822ebe75f]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-RDNIS: [2995]
Caller-Channel-Name: [sofia/internal/sip:2995 at 192.168.100.74]
Caller-Profile-Index: [2]
Caller-Profile-Created-Time: [1302271744429760]
Caller-Channel-Created-Time: [1302271711758979]
Caller-Channel-Answered-Time: [1302271714953388]
Caller-Channel-Progress-Time: [1302271711821261]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [src/switch_channel.c]
login: [src/switch_channel.c]
from: [2995 at 192.168.100.33]
rpid: [unknown]
status: [CS_ROUTING]
event_type: [presence]
alt_event_type: [dialog]
presence-call-direction: [outbound]
event_count: [2]
Presence-Calling-File: [src/switch_channel.c]
Presence-Calling-Function: [switch_channel_perform_set_running_state]
Presence-Calling-Line: [1660]


2011-04-08 10:09:04.530069 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal)
2011-04-08 10:09:04.530069 [WARNING] sofia_presence.c:774 192.168.100.33 is an alias, skipping
2011-04-08 10:09:04.530069 [WARNING] sofia_presence.c:781 external is passive, skipping
2011-04-08 10:09:04.531077 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:04.531077 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='2908' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal-ipv6' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:04]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:04 GMT]
Event-Date-Timestamp: [1302271744431879]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [585]
Channel-State: [CS_ROUTING]
Channel-Call-State: [ACTIVE]
Channel-State-Number: [2]
Channel-Name: [sofia/internal/2908 at 192.168.100.33]
Unique-ID: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-Presence-ID: [2908 at 192.168.100.33]
Channel-Call-UUID: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
Answer-State: [answered]
Channel-Read-Codec-Name: [PCMU]
Channel-Read-Codec-Rate: [8000]
Channel-Read-Codec-Bit-Rate: [64000]
Channel-Write-Codec-Name: [PCMU]
Channel-Write-Codec-Rate: [8000]
Channel-Write-Codec-Bit-Rate: [64000]
Caller-Direction: [inbound]
Caller-Username: [2908]
Caller-Dialplan: [inline]
Caller-Caller-ID-Name: [Gourav Vohra]
Caller-Caller-ID-Number: [2908]
Caller-Callee-ID-Name: [Outbound Call]
Caller-Callee-ID-Number: [2995]
Caller-Network-Addr: [192.168.100.64]
Caller-ANI: [2908]
Caller-Destination-Number: [answer,conference:6c33a351-1842-4e45-9bfb-89249c44a8c6 at sla+flags{mintwo}]
Caller-Unique-ID: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-RDNIS: [2995]
Caller-Channel-Name: [sofia/internal/2908 at 192.168.100.33]
Caller-Profile-Index: [2]
Caller-Profile-Created-Time: [1302271744429760]
Caller-Channel-Created-Time: [1302271711753265]
Caller-Channel-Answered-Time: [1302271714972507]
Caller-Channel-Progress-Time: [1302271711821261]
Caller-Channel-Progress-Media-Time: [1302271711822270]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [src/switch_channel.c]
login: [src/switch_channel.c]
from: [2908 at 192.168.100.33]
rpid: [unknown]
status: [CS_ROUTING]
event_type: [presence]
alt_event_type: [dialog]
presence-call-info-state: [active]
presence-call-info: [appearance-index=1]
presence-call-direction: [inbound]
event_count: [2]
Presence-Calling-File: [src/switch_channel.c]
Presence-Calling-Function: [switch_channel_perform_set_running_state]
Presence-Calling-Line: [1660]


2011-04-08 10:09:04.532084 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:04.533099 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal)
2011-04-08 10:09:04.533099 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'CS_ROUTING','unknown','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '2908 at 192.168.100.33' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='2908' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:04]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:04 GMT]
Event-Date-Timestamp: [1302271744431879]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [585]
Channel-State: [CS_ROUTING]
Channel-Call-State: [ACTIVE]
Channel-State-Number: [2]
Channel-Name: [sofia/internal/2908 at 192.168.100.33]
Unique-ID: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-Presence-ID: [2908 at 192.168.100.33]
Channel-Call-UUID: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
Answer-State: [answered]
Channel-Read-Codec-Name: [PCMU]
Channel-Read-Codec-Rate: [8000]
Channel-Read-Codec-Bit-Rate: [64000]
Channel-Write-Codec-Name: [PCMU]
Channel-Write-Codec-Rate: [8000]
Channel-Write-Codec-Bit-Rate: [64000]
Caller-Direction: [inbound]
Caller-Username: [2908]
Caller-Dialplan: [inline]
Caller-Caller-ID-Name: [Gourav Vohra]
Caller-Caller-ID-Number: [2908]
Caller-Callee-ID-Name: [Outbound Call]
Caller-Callee-ID-Number: [2995]
Caller-Network-Addr: [192.168.100.64]
Caller-ANI: [2908]
Caller-Destination-Number: [answer,conference:6c33a351-1842-4e45-9bfb-89249c44a8c6 at sla+flags{mintwo}]
Caller-Unique-ID: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-RDNIS: [2995]
Caller-Channel-Name: [sofia/internal/2908 at 192.168.100.33]
Caller-Profile-Index: [2]
Caller-Profile-Created-Time: [1302271744429760]
Caller-Channel-Created-Time: [1302271711753265]
Caller-Channel-Answered-Time: [1302271714972507]
Caller-Channel-Progress-Time: [1302271711821261]
Caller-Channel-Progress-Media-Time: [1302271711822270]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [src/switch_channel.c]
login: [src/switch_channel.c]
from: [2908 at 192.168.100.33]
rpid: [unknown]
status: [CS_ROUTING]
event_type: [presence]
alt_event_type: [dialog]
presence-call-info-state: [active]
presence-call-info: [appearance-index=1]
presence-call-direction: [inbound]
event_count: [2]
Presence-Calling-File: [src/switch_channel.c]
Presence-Calling-Function: [switch_channel_perform_set_running_state]
Presence-Calling-Line: [1660]


2011-04-08 10:09:04.533099 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal)
2011-04-08 10:09:04.533099 [WARNING] sofia_presence.c:774 192.168.100.33 is an alias, skipping
2011-04-08 10:09:04.533099 [WARNING] sofia_presence.c:781 external is passive, skipping
2011-04-08 10:09:04.533099 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:04.533099 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Active (1 caller)','','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='6c33a351-1842-4e45-9bfb-89249c44a8c6' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal-ipv6' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:04]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:04 GMT]
Event-Date-Timestamp: [1302271744433706]
Event-Calling-File: [mod_conference.c]
Event-Calling-Function: [conference_add_member]
Event-Calling-Line-Number: [689]
proto: [conf]
login: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
from: [6c33a351-1842-4e45-9bfb-89249c44a8c6 at 192.168.100.33]
status: [Active (1 caller)]
event_type: [presence]
alt_event_type: [dialog]
event_count: [119]
unique-id: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
channel-state: [CS_ROUTING]
answer-state: [early]
presence-call-direction: [outbound]


2011-04-08 10:09:04.534119 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:04.534119 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal)
2011-04-08 10:09:04.534119 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Active (1 caller)','','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='6c33a351-1842-4e45-9bfb-89249c44a8c6' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:04]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:04 GMT]
Event-Date-Timestamp: [1302271744433706]
Event-Calling-File: [mod_conference.c]
Event-Calling-Function: [conference_add_member]
Event-Calling-Line-Number: [689]
proto: [conf]
login: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
from: [6c33a351-1842-4e45-9bfb-89249c44a8c6 at 192.168.100.33]
status: [Active (1 caller)]
event_type: [presence]
alt_event_type: [dialog]
event_count: [119]
unique-id: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
channel-state: [CS_ROUTING]
answer-state: [early]
presence-call-direction: [outbound]


2011-04-08 10:09:04.535125 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal)
2011-04-08 10:09:04.535125 [WARNING] sofia_presence.c:774 192.168.100.33 is an alias, skipping
2011-04-08 10:09:04.535125 [WARNING] sofia_presence.c:781 external is passive, skipping
2011-04-08 10:09:04.535125 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:04.535125 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Active (2 callers)','','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='6c33a351-1842-4e45-9bfb-89249c44a8c6' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal-ipv6' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:04]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:04 GMT]
Event-Date-Timestamp: [1302271744435298]
Event-Calling-File: [mod_conference.c]
Event-Calling-Function: [conference_add_member]
Event-Calling-Line-Number: [689]
proto: [conf]
login: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
from: [6c33a351-1842-4e45-9bfb-89249c44a8c6 at 192.168.100.33]
status: [Active (2 callers)]
event_type: [presence]
alt_event_type: [dialog]
event_count: [120]
unique-id: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
channel-state: [CS_ROUTING]
answer-state: [confirmed]
presence-call-direction: [inbound]


2011-04-08 10:09:04.535125 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:04.535125 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal)
2011-04-08 10:09:04.535125 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Active (2 callers)','','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='6c33a351-1842-4e45-9bfb-89249c44a8c6' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:04]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:04 GMT]
Event-Date-Timestamp: [1302271744435298]
Event-Calling-File: [mod_conference.c]
Event-Calling-Function: [conference_add_member]
Event-Calling-Line-Number: [689]
proto: [conf]
login: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
from: [6c33a351-1842-4e45-9bfb-89249c44a8c6 at 192.168.100.33]
status: [Active (2 callers)]
event_type: [presence]
alt_event_type: [dialog]
event_count: [120]
unique-id: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
channel-state: [CS_ROUTING]
answer-state: [confirmed]
presence-call-direction: [inbound]


2011-04-08 10:09:04.536132 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal)
2011-04-08 10:09:04.536132 [WARNING] sofia_presence.c:774 192.168.100.33 is an alias, skipping
2011-04-08 10:09:04.536132 [WARNING] sofia_presence.c:781 external is passive, skipping
2011-04-08 10:09:04.536132 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:04.537142 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'answered','unknown','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='2995' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal-ipv6' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:04]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:04 GMT]
Event-Date-Timestamp: [1302271744435298]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [585]
Channel-State: [CS_EXECUTE]
Channel-Call-State: [ACTIVE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/2995 at 192.168.100.33]
Unique-ID: [818ddd7f-99e2-4d22-b7ae-e62b6fdbfda4]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-Presence-ID: [2995 at 192.168.100.33]
Answer-State: [answered]
Channel-Read-Codec-Name: [G722]
Channel-Read-Codec-Rate: [16000]
Channel-Read-Codec-Bit-Rate: [64000]
Channel-Write-Codec-Name: [G722]
Channel-Write-Codec-Rate: [16000]
Channel-Write-Codec-Bit-Rate: [64000]
Caller-Direction: [inbound]
Caller-Username: [2995]
Caller-Dialplan: [inline]
Caller-Caller-ID-Name: [Gourav Vohra]
Caller-Caller-ID-Number: [2995]
Caller-Network-Addr: [192.168.100.75]
Caller-ANI: [2995]
Caller-Unique-ID: [818ddd7f-99e2-4d22-b7ae-e62b6fdbfda4]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/2995 at 192.168.100.33]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1302271744429760]
Caller-Channel-Created-Time: [1302271744429760]
Caller-Channel-Answered-Time: [1302271744433706]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [src/switch_channel.c]
login: [src/switch_channel.c]
from: [2995 at 192.168.100.33]
rpid: [unknown]
status: [answered]
event_type: [presence]
alt_event_type: [dialog]
presence-call-info-state: [active]
presence-call-info: [appearance-index=1]
presence-call-direction: [inbound]
event_count: [1]
Presence-Calling-File: [src/switch_channel.c]
Presence-Calling-Function: [switch_channel_perform_mark_answered]
Presence-Calling-Line: [2887]


2011-04-08 10:09:04.537142 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:04.538149 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal)
2011-04-08 10:09:04.538149 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'answered','unknown','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '2995 at 192.168.100.33' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='2995' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:04]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:04 GMT]
Event-Date-Timestamp: [1302271744435298]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [585]
Channel-State: [CS_EXECUTE]
Channel-Call-State: [ACTIVE]
Channel-State-Number: [4]
Channel-Name: [sofia/internal/2995 at 192.168.100.33]
Unique-ID: [818ddd7f-99e2-4d22-b7ae-e62b6fdbfda4]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-Presence-ID: [2995 at 192.168.100.33]
Answer-State: [answered]
Channel-Read-Codec-Name: [G722]
Channel-Read-Codec-Rate: [16000]
Channel-Read-Codec-Bit-Rate: [64000]
Channel-Write-Codec-Name: [G722]
Channel-Write-Codec-Rate: [16000]
Channel-Write-Codec-Bit-Rate: [64000]
Caller-Direction: [inbound]
Caller-Username: [2995]
Caller-Dialplan: [inline]
Caller-Caller-ID-Name: [Gourav Vohra]
Caller-Caller-ID-Number: [2995]
Caller-Network-Addr: [192.168.100.75]
Caller-ANI: [2995]
Caller-Unique-ID: [818ddd7f-99e2-4d22-b7ae-e62b6fdbfda4]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/2995 at 192.168.100.33]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1302271744429760]
Caller-Channel-Created-Time: [1302271744429760]
Caller-Channel-Answered-Time: [1302271744433706]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [src/switch_channel.c]
login: [src/switch_channel.c]
from: [2995 at 192.168.100.33]
rpid: [unknown]
status: [answered]
event_type: [presence]
alt_event_type: [dialog]
presence-call-info-state: [active]
presence-call-info: [appearance-index=1]
presence-call-direction: [inbound]
event_count: [1]
Presence-Calling-File: [src/switch_channel.c]
Presence-Calling-Function: [switch_channel_perform_mark_answered]
Presence-Calling-Line: [2887]


2011-04-08 10:09:04.539155 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal)
2011-04-08 10:09:04.539155 [WARNING] sofia_presence.c:774 192.168.100.33 is an alias, skipping
2011-04-08 10:09:04.539155 [WARNING] sofia_presence.c:781 external is passive, skipping
2011-04-08 10:09:04.539155 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:04.539155 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Active (3 callers)','','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='6c33a351-1842-4e45-9bfb-89249c44a8c6' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal-ipv6' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:04]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:04 GMT]
Event-Date-Timestamp: [1302271744436366]
Event-Calling-File: [mod_conference.c]
Event-Calling-Function: [conference_add_member]
Event-Calling-Line-Number: [689]
proto: [conf]
login: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
from: [6c33a351-1842-4e45-9bfb-89249c44a8c6 at 192.168.100.33]
status: [Active (3 callers)]
event_type: [presence]
alt_event_type: [dialog]
event_count: [121]
unique-id: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
channel-state: [CS_ROUTING]
answer-state: [confirmed]
presence-call-direction: [inbound]


2011-04-08 10:09:04.539155 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:04.539155 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal)
2011-04-08 10:09:04.539155 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Active (3 callers)','','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='6c33a351-1842-4e45-9bfb-89249c44a8c6' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:04]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:04 GMT]
Event-Date-Timestamp: [1302271744436366]
Event-Calling-File: [mod_conference.c]
Event-Calling-Function: [conference_add_member]
Event-Calling-Line-Number: [689]
proto: [conf]
login: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
from: [6c33a351-1842-4e45-9bfb-89249c44a8c6 at 192.168.100.33]
status: [Active (3 callers)]
event_type: [presence]
alt_event_type: [dialog]
event_count: [121]
unique-id: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
channel-state: [CS_ROUTING]
answer-state: [confirmed]
presence-call-direction: [inbound]


2011-04-08 10:09:04.540162 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal)
2011-04-08 10:09:04.540162 [WARNING] sofia_presence.c:774 192.168.100.33 is an alias, skipping
2011-04-08 10:09:13.943151 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/log/cdr-csv/2995.csv
2011-04-08 10:09:13.943151 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/log/cdr-csv/Master.csv

2011-04-08 10:09:13.944159 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/log/cdr-csv/2908.csv
2011-04-08 10:09:13.944159 [NOTICE] mod_logfile.c:158 New log started.
2011-04-08 10:09:27.679863 [DEBUG] switch_channel.c:2563 (sofia/internal/2995 at 192.168.100.33) Callstate Change ACTIVE -> HANGUP
2011-04-08 10:09:27.679863 [NOTICE] sofia.c:537 Hangup sofia/internal/2995 at 192.168.100.33 [CS_EXECUTE] [NORMAL_CLEARING]
2011-04-08 10:09:27.679863 [DEBUG] switch_channel.c:2579 Send signal sofia/internal/2995 at 192.168.100.33 [KILL]
2011-04-08 10:09:27.679863 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/2995 at 192.168.100.33 [BREAK]
2011-04-08 10:09:27.691976 [DEBUG] mod_conference.c:2810 Channel leaving conference, cause: NORMAL_CLEARING
2011-04-08 10:09:27.692990 [DEBUG] mod_conference.c:5986 sofia/internal/2995 at 192.168.100.33 skip receive message [UNBRIDGE] (channel is hungup already)
2011-04-08 10:09:27.692990 [DEBUG] switch_core_codec.c:141 sofia/internal/2995 at 192.168.100.33 Restore previous codec G722:9.
2011-04-08 10:09:27.692990 [DEBUG] switch_core_session.c:2060 sofia/internal/2995 at 192.168.100.33 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2011-04-08 10:09:27.692990 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/2995 at 192.168.100.33) State EXECUTE going to sleep
2011-04-08 10:09:27.692990 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/2995 at 192.168.100.33) Running State Change CS_HANGUP
2011-04-08 10:09:27.692990 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/2995 at 192.168.100.33) State HANGUP
2011-04-08 10:09:27.692990 [DEBUG] mod_sofia.c:451 sofia/internal/2995 at 192.168.100.33 Overriding SIP cause 480 with 200 from the other leg
2011-04-08 10:09:27.692990 [DEBUG] mod_sofia.c:457 Channel sofia/internal/2995 at 192.168.100.33 hanging up, cause: NORMAL_CLEARING
2011-04-08 10:09:27.693998 [DEBUG] switch_core_state_machine.c:46 sofia/internal/2995 at 192.168.100.33 Standard HANGUP, cause: NORMAL_CLEARING
2011-04-08 10:09:27.693998 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/2995 at 192.168.100.33) State HANGUP going to sleep
2011-04-08 10:09:27.693998 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/2995 at 192.168.100.33) State Change CS_HANGUP -> CS_REPORTING
2011-04-08 10:09:27.693998 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/2995 at 192.168.100.33 [BREAK]
2011-04-08 10:09:27.693998 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/2995 at 192.168.100.33) Running State Change CS_REPORTING
2011-04-08 10:09:27.693998 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/2995 at 192.168.100.33) State REPORTING
2011-04-08 10:09:27.693998 [DEBUG] switch_core_state_machine.c:53 sofia/internal/2995 at 192.168.100.33 Standard REPORTING, cause: NORMAL_CLEARING
2011-04-08 10:09:27.693998 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/2995 at 192.168.100.33) State REPORTING going to sleep
2011-04-08 10:09:27.693998 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/2995 at 192.168.100.33) State Change CS_REPORTING -> CS_DESTROY
2011-04-08 10:09:27.693998 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/2995 at 192.168.100.33 [BREAK]
2011-04-08 10:09:27.693998 [DEBUG] switch_core_session.c:1288 Session 111 (sofia/internal/2995 at 192.168.100.33) Locked, Waiting on external entities
2011-04-08 10:09:27.693998 [NOTICE] switch_core_session.c:1306 Session 111 (sofia/internal/2995 at 192.168.100.33) Ended
2011-04-08 10:09:27.693998 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/2995 at 192.168.100.33 [CS_DESTROY]
2011-04-08 10:09:27.695091 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/2995 at 192.168.100.33) Callstate Change HANGUP -> DOWN
2011-04-08 10:09:27.695091 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/2995 at 192.168.100.33) Running State Change CS_DESTROY
2011-04-08 10:09:27.695091 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/2995 at 192.168.100.33) State DESTROY
2011-04-08 10:09:27.695091 [DEBUG] mod_sofia.c:362 sofia/internal/2995 at 192.168.100.33 SOFIA DESTROY
2011-04-08 10:09:27.695091 [DEBUG] switch_core_state_machine.c:60 sofia/internal/2995 at 192.168.100.33 Standard DESTROY
2011-04-08 10:09:27.695091 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/2995 at 192.168.100.33) State DESTROY going to sleep
2011-04-08 10:09:27.741861 [WARNING] sofia_presence.c:781 external is passive, skipping
2011-04-08 10:09:27.741861 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:27.741861 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Active (2 callers)','','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='6c33a351-1842-4e45-9bfb-89249c44a8c6' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal-ipv6' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:27]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:27 GMT]
Event-Date-Timestamp: [1302271767692990]
Event-Calling-File: [mod_conference.c]
Event-Calling-Function: [conference_del_member]
Event-Calling-Line-Number: [890]
proto: [conf]
login: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
from: [6c33a351-1842-4e45-9bfb-89249c44a8c6 at 192.168.100.33]
status: [Active (2 callers)]
event_type: [presence]
alt_event_type: [dialog]
event_count: [122]
unique-id: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
channel-state: [CS_ROUTING]
answer-state: [confirmed]
call-direction: [inbound]


2011-04-08 10:09:27.742874 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:27.742874 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal)
2011-04-08 10:09:27.742874 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Active (2 callers)','','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='6c33a351-1842-4e45-9bfb-89249c44a8c6' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:27]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:27 GMT]
Event-Date-Timestamp: [1302271767692990]
Event-Calling-File: [mod_conference.c]
Event-Calling-Function: [conference_del_member]
Event-Calling-Line-Number: [890]
proto: [conf]
login: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
from: [6c33a351-1842-4e45-9bfb-89249c44a8c6 at 192.168.100.33]
status: [Active (2 callers)]
event_type: [presence]
alt_event_type: [dialog]
event_count: [122]
unique-id: [6c33a351-1842-4e45-9bfb-89249c44a8c6]
channel-state: [CS_ROUTING]
answer-state: [confirmed]
call-direction: [inbound]


2011-04-08 10:09:27.742874 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal)
2011-04-08 10:09:27.742874 [WARNING] sofia_presence.c:774 192.168.100.33 is an alias, skipping
2011-04-08 10:09:27.742874 [WARNING] sofia_presence.c:781 external is passive, skipping
2011-04-08 10:09:27.743881 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:27.743881 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Available','unknown','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='2995' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal-ipv6' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:27]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:27 GMT]
Event-Date-Timestamp: [1302271767692990]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [585]
Channel-State: [CS_HANGUP]
Channel-Call-State: [HANGUP]
Channel-State-Number: [10]
Channel-Name: [sofia/internal/2995 at 192.168.100.33]
Unique-ID: [818ddd7f-99e2-4d22-b7ae-e62b6fdbfda4]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-Presence-ID: [2995 at 192.168.100.33]
Answer-State: [hangup]
Channel-Read-Codec-Name: [G722]
Channel-Read-Codec-Rate: [16000]
Channel-Read-Codec-Bit-Rate: [64000]
Channel-Write-Codec-Name: [G722]
Channel-Write-Codec-Rate: [16000]
Channel-Write-Codec-Bit-Rate: [64000]
Caller-Direction: [inbound]
Caller-Username: [2995]
Caller-Dialplan: [inline]
Caller-Caller-ID-Name: [Gourav Vohra]
Caller-Caller-ID-Number: [2995]
Caller-Network-Addr: [192.168.100.75]
Caller-ANI: [2995]
Caller-Unique-ID: [818ddd7f-99e2-4d22-b7ae-e62b6fdbfda4]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/2995 at 192.168.100.33]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1302271744429760]
Caller-Channel-Created-Time: [1302271744429760]
Caller-Channel-Answered-Time: [1302271744433706]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [src/switch_channel.c]
login: [src/switch_channel.c]
from: [2995 at 192.168.100.33]
rpid: [unknown]
status: [CS_HANGUP]
event_type: [presence]
alt_event_type: [dialog]
presence-call-info-state: [idle]
presence-call-info: [appearance-index=1]
presence-call-direction: [inbound]
event_count: [2]
Presence-Calling-File: [src/switch_channel.c]
Presence-Calling-Function: [switch_channel_perform_set_running_state]
Presence-Calling-Line: [1660]


2011-04-08 10:09:27.744888 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal-ipv6)
2011-04-08 10:09:27.745897 [INFO] sofia_presence.c:862 IN START_PRESENCE_SQL (internal)
2011-04-08 10:09:27.745897 [ERR] sofia_presence.c:871 DUMP PRESENCE SQL:
select distinct sip_subscriptions.proto,sip_subscriptions.sip_user,sip_subscriptions.sip_host,sip_subscriptions.sub_to_user,sip_subscriptions.sub_to_host,sip_subscriptions.event,sip_subscriptions.contact,sip_subscriptions.call_id,sip_subscriptions.full_from,sip_subscriptions.full_via,sip_subscriptions.expires,sip_subscriptions.user_agent,sip_subscriptions.accept,sip_subscriptions.profile_name,'Available','unknown','192.168.100.33',sip_presence.status,sip_presence.rpid,sip_presence.open_closed,'','',sip_subscriptions.version, '2995 at 192.168.100.33' 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.version > -1 and sip_subscriptions.expires > -1 and (event='presence' or event='dialog') and sub_to_user='2995' and (sub_to_host='192.168.100.33' or presence_hosts like '%192.168.100.33%') and (sip_subscriptions.profile_name = 'internal' or sip_subscriptions.presence_hosts != sip_subscriptions.sub_to_host) 
EVENT DUMP:
Event-Name: [PRESENCE_IN]
Core-UUID: [4ff974b0-d64a-4319-8969-9781b791d838]
FreeSWITCH-Hostname: [testsrv1]
FreeSWITCH-IPv4: [192.168.100.33]
FreeSWITCH-IPv6: [::1]
Event-Date-Local: [2011-04-08 10:09:27]
Event-Date-GMT: [Fri, 08 Apr 2011 14:09:27 GMT]
Event-Date-Timestamp: [1302271767692990]
Event-Calling-File: [switch_channel.c]
Event-Calling-Function: [switch_channel_perform_presence]
Event-Calling-Line-Number: [585]
Channel-State: [CS_HANGUP]
Channel-Call-State: [HANGUP]
Channel-State-Number: [10]
Channel-Name: [sofia/internal/2995 at 192.168.100.33]
Unique-ID: [818ddd7f-99e2-4d22-b7ae-e62b6fdbfda4]
Call-Direction: [inbound]
Presence-Call-Direction: [inbound]
Channel-Presence-ID: [2995 at 192.168.100.33]
Answer-State: [hangup]
Channel-Read-Codec-Name: [G722]
Channel-Read-Codec-Rate: [16000]
Channel-Read-Codec-Bit-Rate: [64000]
Channel-Write-Codec-Name: [G722]
Channel-Write-Codec-Rate: [16000]
Channel-Write-Codec-Bit-Rate: [64000]
Caller-Direction: [inbound]
Caller-Username: [2995]
Caller-Dialplan: [inline]
Caller-Caller-ID-Name: [Gourav Vohra]
Caller-Caller-ID-Number: [2995]
Caller-Network-Addr: [192.168.100.75]
Caller-ANI: [2995]
Caller-Unique-ID: [818ddd7f-99e2-4d22-b7ae-e62b6fdbfda4]
Caller-Source: [mod_sofia]
Caller-Context: [default]
Caller-Channel-Name: [sofia/internal/2995 at 192.168.100.33]
Caller-Profile-Index: [1]
Caller-Profile-Created-Time: [1302271744429760]
Caller-Channel-Created-Time: [1302271744429760]
Caller-Channel-Answered-Time: [1302271744433706]
Caller-Channel-Progress-Time: [0]
Caller-Channel-Progress-Media-Time: [0]
Caller-Channel-Hangup-Time: [0]
Caller-Channel-Transfer-Time: [0]
Caller-Screen-Bit: [true]
Caller-Privacy-Hide-Name: [false]
Caller-Privacy-Hide-Number: [false]
proto: [src/switch_channel.c]
login: [src/switch_channel.c]
from: [2995 at 192.168.100.33]
rpid: [unknown]
status: [CS_HANGUP]
event_type: [presence]
alt_event_type: [dialog]
presence-call-info-state: [idle]
presence-call-info: [appearance-index=1]
presence-call-direction: [inbound]
event_count: [2]
Presence-Calling-File: [src/switch_channel.c]
Presence-Calling-Function: [switch_channel_perform_set_running_state]
Presence-Calling-Line: [1660]


2011-04-08 10:09:27.745897 [INFO] sofia_presence.c:890 IN END_PRESENCE_SQL (internal)
2011-04-08 10:09:27.745897 [WARNING] sofia_presence.c:774 192.168.100.33 is an alias