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, skipping
2011-04-08 10:09:33.381769 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/log/cdr-csv/2995.csv
2011-04-08 10:09:33.381769 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/log/cdr-csv/Master.csv




From eric at loopfx.com  Tue Apr 12 21:59:46 2011
From: eric at loopfx.com (Eric Beard)
Date: Tue, 12 Apr 2011 13:59:46 -0400
Subject: [Freeswitch-users] Lua session originate
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DCA92D3D@doctorbob.loop.local>

When I try to use lua to originate a call, I get this:

freeswitch at internal> lua test_originate.lua
-ERR encountered

freeswitch at internal> 2011-04-12 13:03:17.806648 [ERR] mod_lua.cpp:191 Error in originate expected 4.
.4 args, got 3
stack traceback:
        [C]: in function 'originate'
        /usr/local/freeswitch/scripts/test_originate.lua:3: in main chunk

I'm using the following line of code:

local new_session = freeswitch.Session();
new_session.originate(session, "sofia/gateway/mygatewayname/mynumber", 60);

This is in the Wiki:
session:originate
local new_session = freeswitch.Session();
new_session.originate(session, dest[, timeout]);
dest - quoted dialplan destanation. For example: "sofia/internal/1000 at 10.0.0.1" or "sofia/gateway/my_sip_provider/my_dest_number"
timeout - origination timeout in seconds
Note: in recent git versions, session.originate expects 4 arguments; the example above does not work (anymore?).
What is the 4th arg?


-----------------------
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 elijah at crankenstein.com  Tue Apr 12 22:25:15 2011
From: elijah at crankenstein.com (elijah)
Date: Tue, 12 Apr 2011 11:25:15 -0700
Subject: [Freeswitch-users] error loading mod_file_string
Message-ID: <BANLkTikZh1v5uHyOaMDpOtNH_FHNSZG2Kg@mail.gmail.com>

After updating a FreeSwitch installation against the repository today I see
this error in the console on startup:

switch_loadable_module.c:928 Error Loading module
/usr/local/freeswitch/mod/mod_file_string.so
**/usr/local/freeswitch/mod/mod_file_string.so: cannot open shared object
file: No such file or directory**

There are no other apparent symptoms. Please let me know if I'm an idiot or
if you prefer I enter this in Jira or just think I should comment out
mod_file_string in my modules.conf.

thanks,
elijah
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From nick.rosier at gmail.com  Tue Apr 12 22:26:09 2011
From: nick.rosier at gmail.com (Nick Rosier)
Date: Tue, 12 Apr 2011 20:26:09 +0200
Subject: [Freeswitch-users] Gateway with dynamic IP address
In-Reply-To: <BANLkTi=8LfL-CMkczXgJg8FOF1_FTYtxdQ@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>
	<BANLkTi=8LfL-CMkczXgJg8FOF1_FTYtxdQ@mail.gmail.com>
Message-ID: <BANLkTims24tpVuEmq1r84XA1bZd02ebfsw@mail.gmail.com>

The problem is this isn't a user, it's a gateway that registers itself
as a user (so FS does not register to it but the other way around). I
need to dial to it as a user and specify the number it has to dial.

On 12 April 2011 12:20, Steven Ayre <steveayre at gmail.com> wrote:
> 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From dave at dchorton.com  Tue Apr 12 20:49:42 2011
From: dave at dchorton.com (Dave Horton)
Date: Tue, 12 Apr 2011 12:49:42 -0400
Subject: [Freeswitch-users] how to pass arbitrary headers from A leg to
	B leg when bridging
Message-ID: <047B4BA6-39DE-4BB4-BFED-7353A6EBBB8B@dchorton.com>

So I'm guessing this isn't possible without hacking the source code, which I've already done to solve my problem for now.

But I'd like to make sure there isn't a better way of doing things, and thus I'd like to revise and restate my question for clarity.  First, though, let me describe what I am doing, because I think it's a not-uncommon scenario that I think would be something that others may want to do.  I basically want to use FS as a simple transcoding server between two endpoints, call them A and B.  Calls coming in from A will be using speex codec and I want to send them out to B using PCMU; calls coming in from B will be PCMU and I want to send them to A using speex.  The FS server will be a B2BUA and will be doing transcoding only -- no authentication (and no registration).  Simple, right?  The only fly in the ointment is that A is authenticating calls with B by providing a Proxy-Authorization header.  So I need to take the Proxy-Authorization header received on the A leg and include it on the B leg.

So far, the only way I have found to do that is to hack the code to create a new channel variable.  I've done this, and it works.  However, this leads me to the following questions

1) Is there a better way to do this?  If there is no way to do it as a dialplan out of the box, can it be done as a script?

2) sofia has parsed all of the sip headers on the incoming invite for us, and they're all available from mod_sofia.  Shouldn't those all be available to us (i.e., application developers) by some means (i.e., channel variables)?

Dave



From m.sobkow at marketelsystems.com  Tue Apr 12 22:29:10 2011
From: m.sobkow at marketelsystems.com (Mark Sobkow)
Date: Tue, 12 Apr 2011 12:29:10 -0600
Subject: [Freeswitch-users] Can someone point me to examples of how to
 program message-based calls?
In-Reply-To: <4DA32EAA.8070104@marketelsystems.com>
References: <32121300.77951302392962541.JavaMail.root@julie.marketel>	<549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDB9@cooper>
	<4DA32EAA.8070104@marketelsystems.com>
Message-ID: <4DA499F6.408@marketelsystems.com>

Just to let you know, the event-based Erlang interface changes I was 
working on are now functional.  Our guinea pigs are testing the new code 
now.  I had some hairy issues in our application code to resolve once I 
started trapping the events, but it works.

-- 
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 brad at tritelcomm.com  Tue Apr 12 22:35:53 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Tue, 12 Apr 2011 11:35:53 -0700
Subject: [Freeswitch-users] error loading mod_file_string
In-Reply-To: <BANLkTikZh1v5uHyOaMDpOtNH_FHNSZG2Kg@mail.gmail.com>
References: <BANLkTikZh1v5uHyOaMDpOtNH_FHNSZG2Kg@mail.gmail.com>
Message-ID: <BANLkTikz-qyJFFuH5D_5G6Aa1G5-UB20+g@mail.gmail.com>

This sounds like it just wasn't compiled. Try manually compiling the module.
cd /usr/src/freeswitch && make mod_file_string-install

Make sure you edit modules.conf and uncomment the mod_file_string line to
enable this to be built upon every 'make current'

On Tue, Apr 12, 2011 at 11:25 AM, elijah <elijah at crankenstein.com> wrote:

> After updating a FreeSwitch installation against the repository today I see
> this error in the console on startup:
>
> switch_loadable_module.c:928 Error Loading module
> /usr/local/freeswitch/mod/mod_file_string.so
> **/usr/local/freeswitch/mod/mod_file_string.so: cannot open shared object
> file: No such file or directory**
>
> There are no other apparent symptoms. Please let me know if I'm an idiot or
> if you prefer I enter this in Jira or just think I should comment out
> mod_file_string in my modules.conf.
>
> thanks,
> elijah
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Apr 12 22:41:49 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 12 Apr 2011 11:41:49 -0700 (PDT)
Subject: [Freeswitch-users] Lua session originate
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DCA92D3D@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DCA92D3D@doctorbob.loop.local>
Message-ID: <1302633709973-6266239.post@n2.nabble.com>

The fourth parameter is the switch_state_handler_table reference and for most
uses(script language) you should just pass null  but I dont use lua so I
dont know the details on that.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Lua-session-originate-tp6266121p6266239.html
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From jeff at jefflenk.com  Tue Apr 12 22:42:36 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Tue, 12 Apr 2011 11:42:36 -0700 (PDT)
Subject: [Freeswitch-users] error loading mod_file_string
In-Reply-To: <BANLkTikz-qyJFFuH5D_5G6Aa1G5-UB20+g@mail.gmail.com>
References: <BANLkTikZh1v5uHyOaMDpOtNH_FHNSZG2Kg@mail.gmail.com>
	<BANLkTikz-qyJFFuH5D_5G6Aa1G5-UB20+g@mail.gmail.com>
Message-ID: <1302633756754-6266242.post@n2.nabble.com>

that module has been removed - no longer needed.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/error-loading-mod-file-string-tp6266207p6266242.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From frankie.k.yiu at gmail.com  Tue Apr 12 22:55:28 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Tue, 12 Apr 2011 11:55:28 -0700
Subject: [Freeswitch-users] Problem with switch_core_session_send_dtmf()--no
 event generated, no DTMF found in channel?
Message-ID: <BANLkTikJvjhZ6Gbq97oR9Q_EemYtCUMQAw@mail.gmail.com>

Hi there,

New to FreeSWITCH and hope someone here can answer my question.

I am calling the following function during a call:

1) switch_dtmf_t dtmf = { '*', switch_core_default_dtmf_duration(0), 0};
2) switch_status_t dtmfStatus = switch_core_session_send_dtmf(curSession,
&dtmf);     < return SUCCESS >
3) switch_size_t iDTMF =
switch_channel_has_dtmf(switch_core_session_get_channel(oDataFrameInfo.strSession));
< return 0 >

I am assuming after #2, I should get DTMF event and #3 should return 1,
however I do not get any of those

Could someone please tell me what is wrong with my code or if there is bug
in freeSWITCH code?

Thanks in avance.

Frankie
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From eric at loopfx.com  Tue Apr 12 23:14:57 2011
From: eric at loopfx.com (Eric Beard)
Date: Tue, 12 Apr 2011 15:14:57 -0400
Subject: [Freeswitch-users] Lua session originate
In-Reply-To: <1302633709973-6266239.post@n2.nabble.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DCA92D3D@doctorbob.loop.local>
	<1302633709973-6266239.post@n2.nabble.com>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DCA92E03@doctorbob.loop.local>

Thanks Jeff,

When I try NULL I get this:

freeswitch at internal> 2011-04-12 14:49:31.582049 [ERR] mod_lua.cpp:191 Error in originate (arg 2), ex
pected 'CoreSession *' got 'string'
stack traceback:
        [C]: in function 'originate'
        /usr/local/freeswitch/scripts/test_originate.lua:3: in main chunk

I've been digging through the source code, and I can't find where 4 args are required.  I see this in freeswitch_lua.cpp:

int Session::originate(CoreSession *a_leg_session, char *dest, int timeout)
{
        int x = CoreSession::originate(a_leg_session, dest, timeout);

        if (x) {
                setLUA(L);
        }

        return x;
}

-----------------------
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 Jeff Lenk
Sent: Tuesday, April 12, 2011 2:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Lua session originate

The fourth parameter is the switch_state_handler_table reference and for most
uses(script language) you should just pass null  but I dont use lua so I
dont know the details on that.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Lua-session-originate-tp6266121p6266239.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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


From msc at freeswitch.org  Tue Apr 12 23:19:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Apr 2011 12:19:10 -0700
Subject: [Freeswitch-users] write/read-codec variables
In-Reply-To: <6141302607644@web133.yandex.ru>
References: <6141302607644@web133.yandex.ru>
Message-ID: <BANLkTik1dB6AsgDN2aSYC-im-Ci=PsPB3g@mail.gmail.com>

Well, you will need to parse the b-leg CDR on some level. There is no
convention for logging the A leg CDR and only one specific variable from the
B leg. I suppose you can play around with this variable:

http://wiki.freeswitch.org/wiki/Variable_copy_xml_cdr

However, I doubt what you want to do is possible without getting your hands
a little dirty with parsing some XML.

-MC

On Tue, Apr 12, 2011 at 4:27 AM, Serge S. Yuriev <me at nevian.org> wrote:

>
> 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gchen00 at insightbb.com  Wed Apr 13 00:00:22 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Tue, 12 Apr 2011 16:00:22 -0400
Subject: [Freeswitch-users] FS does not repose to SIP OK message.
Message-ID: <fc6f8b8d15ce3.4da47716@insightbb.com>

Just update my test FS to newest snapshot: FreeSWITCH Version 1.0.head (git-5310735 2011-04-07 15-47-30 -0500)
I am using SJphone softphone to call into my test FS1. This FS1 then forward the call to another FS2.
FS2 will answer the call and start Music On Hold.? Basically I am using SJphone to initiate a SIP call? and let FS2 to answer the call with Music On Hold.
It is working with older version of FS. Now after update to this newest version. The call sometime will not go through. The SJPhone just keep ringing until timeout. This happens maybe on half of the calls. I also tried to use Asterisk to replace FS2 for Music On Hold and it did the same thing. The following is the part of console sofia log info:
2011-04-12 15:29:06.285484 [DEBUG] mod_sofia.c:84 sofia/internal/5596 at 226.59.129.221:5060 SOFIA INIT
nua: nh_create_handle: entering
nua: nua_handle_bind: entering
nua: nua_invite: entering
nua(0x18cc2c80): sent signal r_invite
2011-04-12 15:29:06.285484 [DEBUG] mod_sofia.c:124 (sofia/internal/5596 at 226.59.129.221:5060) State Change CS_INIT -> CS_ROUTING
2011-04-12 15:29:06.285484 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/5596 at 226.59.129.221:5060 [BREAK]
2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/5596 at 226.59.129.221:5060) State INIT going to sleep
2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/5596 at 226.59.129.221:5060) Running State Change CS_ROUTING
nua(0x18cc2c80): recv signal r_invite
2011-04-12 15:29:06.285484 [DEBUG] switch_channel.c:1668 (sofia/internal/5596 at 226.59.129.221:5060) Callstate Change DOWN -> RINGING
nua: nua_stack_set_params: entering
soa_clone(static::0x18c66f60, 0x18c474e0, 0x18cc2c80) called
soa_set_params(static::0x2aaabc072a60, ...) called
soa_set_params(static::0x2aaabc072a60, ...) called
soa_set_user_sdp(static::0x2aaabc072a60, (nil), 0x18ccb7b7, -1) called
soa_set_capability_sdp(static::0x2aaabc072a60, (nil), 0x18ccb7b7, -1) called
2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/5596 at 226.59.129.221:5060) State ROUTING
2011-04-12 15:29:06.285484 [DEBUG] mod_sofia.c:147 sofia/internal/5596 at 226.59.129.221:5060 SOFIA ROUTING
2011-04-12 15:29:06.285484 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/5596 at 226.59.129.221:5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA
nua(0x18cc2c80): adding session usage
2011-04-12 15:29:06.285484 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/5596 at 226.59.129.221:5060 [BREAK]
2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/5596 at 226.59.129.221:5060) State ROUTING going to sleep
2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/5596 at 226.59.129.221:5060) Running State Change CS_CONSUME_MEDIA
nta_leg_tcreate(0x2aaaac05e220)
soa_init_offer_answer(static::0x2aaabc072a60) called
soa_generate_offer(static::0x2aaabc072a60, 0) called
soa_static_offer_answer_action(0x2aaabc072a60, soa_generate_offer): called
soa_static(0x2aaabc072a60, soa_generate_offer): generating local description
soa_static(0x2aaabc072a60, soa_generate_offer): upgrade with local description
2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/5596 at 226.59.129.221:5060) State CONSUME_MEDIA
soa_sdp_mode_set(0x406f6c60, (nil), ""): called
2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/5596 at 226.59.129.221:5060) State CONSUME_MEDIA going to sleep
soa_static(0x2aaabc072a60, soa_generate_offer): storing local description
soa_get_local_sdp(static::0x2aaabc072a60, [(nil)], [0x406f6dc8], [0x406f6dd4]) called
nta: selecting scheme sip
tport_tsend(0x18c69750) tpn = */226.59.129.221:5060
tport_resolve addrinfo = 226.59.129.221:5060
tport_by_addrinfo(0x18c69750): not found by name */226.59.129.221:5060
tport_vsend(0x18c69750): 1245 bytes of 1245 to udp/226.59.129.221:5060
tport_vsend returned 1245
nta: sent INVITE (10982209) to */226.59.129.221:5060
tport_pend(0x18c69750): pending 0x2aaaac059c30 for udp/226.59.129.223:5060 (already 0)
nta: timer set to 32000 ms
nta: timer shortened to 1000 ms
nua(0x18cc2c80): call state changed: init -> calling, sent offer
soa_get_local_sdp(static::0x2aaabc072a60, [0x406f6db8], [0x406f6db0], [(nil)]) called
nua(0x18cc2c80): event i_state INVITE sent
nua: nua_application_event: entering
2011-04-12 15:29:06.285484 [DEBUG] sofia.c:4761 Channel sofia/internal/5596 at 226.59.129.221:5060 entering state [calling][0]
nua: nua_handle_magic: entering
tport_wakeup_pri(0x18c69750): events IN
tport_recv_event(0x18c69750)
tport_recv_iovec(0x18c69750) msg 0x18c87620 from (udp/226.59.129.223:5060) has 344 bytes, veclen = 1
tport_deliver(0x18c69750): msg 0x18c87620 (344 bytes) from udp/226.59.129.221:5060/sip next=(nil)
nta: received 100 Trying for INVITE (10982209)
nta: 100 Trying is going to a transaction
nta_outgoing: RTT is 0.765 ms
tport_release(0x18c69750): 0x2aaaac059c30 by 0x2aaabc05da60 with 0x18c87620 (preliminary)
tport_wakeup_pri(0x18c69750): events IN
tport_recv_event(0x18c69750)
tport_recv_iovec(0x18c69750) msg 0x2aaabc070b20 from (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
tport_deliver(0x18c69750): msg 0x2aaabc070b20 (1235 bytes) from udp/226.59.129.221:5060/sip next=(nil)
nta: received 200 OK for INVITE (10982209)
nta: 200 OK is going to a transaction
tport_release(0x18c69750): 0x2aaaac059c30 by 0x2aaabc05da60 with 0x2aaabc070b20
soa_set_remote_sdp(static::0x2aaabc072a60, (nil), 0x2aaabc0712ac, 247) called
soa_process_answer(static::0x2aaabc072a60) called
soa_static_offer_answer_action(0x2aaabc072a60, soa_process_answer): called
soa_sdp_mode_set(0x2aaabc070240, 0x2aaabc073500, ""): called
soa_static(0x2aaabc072a60, soa_process_answer): upgrade codecs with remote description
soa_static(0x2aaabc072a60, soa_process_answer): storing local description
soa_activate(static::0x2aaabc072a60, (nil)) called
nua(0x18cc2c80): INVITE: processed SDP answer in 200 OK
nua(0x18cc2c80): event r_invite 200 OK
nua(0x18cc2c80): call state changed: calling -> completing, received answer
soa_get_remote_sdp(static::0x2aaabc072a60, [0x406f6828], [0x406f6820], [(nil)]) called
soa_get_params(static::0x2aaabc072a60, ...) called
nua: nua_application_event: entering
nua(0x18cc2c80): event i_state 200 OK
2011-04-12 15:29:06.294311 [INFO] sofia.c:740 sofia/internal/5596 at 226.59.129.221:5060 Update Callee ID to "5596" <sip:5596 at 226.59.129.221>
nta: timer not set
tport_wakeup_pri(0x18c69750): events IN
tport_recv_event(0x18c69750)
tport_recv_iovec(0x18c69750) msg 0x2aaab4022620 from (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
tport_deliver(0x18c69750): msg 0x2aaab4022620 (1235 bytes) from udp/226.59.129.221:5060/sip next=(nil)
nta: received 200 OK for INVITE (10982209)
nta: 200 OK is going to a transaction
nta: 200 OK is duplicate response to 10982209 INVITE
??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
nta: timer set next to 31009 ms
tport_wakeup_pri(0x18c69750): events IN
tport_recv_event(0x18c69750)
tport_recv_iovec(0x18c69750) msg 0x2aaab4022620 from (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
tport_deliver(0x18c69750): msg 0x2aaab4022620 (1235 bytes) from udp/226.59.129.221:5060/sip next=(nil)
nta: received 200 OK for INVITE (10982209)
nta: 200 OK is going to a transaction
nta: 200 OK is duplicate response to 10982209 INVITE
??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
tport_wakeup_pri(0x18c69750): events IN
tport_recv_event(0x18c69750)
tport_recv_iovec(0x18c69750) msg 0x2aaaac060220 from (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
tport_deliver(0x18c69750): msg 0x2aaaac060220 (1235 bytes) from udp/226.59.129.221:5060/sip next=(nil)
nta: received 200 OK for INVITE (10982209)
nta: 200 OK is going to a transaction
nta: 200 OK is duplicate response to 10982209 INVITE
??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
tport_wakeup_pri(0x18c69750): events IN
tport_recv_event(0x18c69750)
tport_recv_iovec(0x18c69750) msg 0x2aaaac060220 from (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
tport_deliver(0x18c69750): msg 0x2aaaac060220 (1235 bytes) from udp/226.59.129.221:5060/sip next=(nil)
nta: received 200 OK for INVITE (10982209)
nta: 200 OK is going to a transaction
nta: 200 OK is duplicate response to 10982209 INVITE
??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
tport_wakeup_pri(0x18c69750): events IN
tport_recv_event(0x18c69750)
tport_recv_iovec(0x18c69750) msg 0x2aaaac060220 from (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
tport_deliver(0x18c69750): msg 0x2aaaac060220 (1235 bytes) from udp/226.59.129.221:5060/sip next=(nil)
nta: received 200 OK for INVITE (10982209)
nta: 200 OK is going to a transaction
nta: 200 OK is duplicate response to 10982209 INVITE
??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
tport_wakeup_pri(0x18c69750): events IN
tport_recv_event(0x18c69750)
tport_recv_iovec(0x18c69750) msg 0x18c87620 from (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
tport_deliver(0x18c69750): msg 0x18c87620 (1235 bytes) from udp/226.59.129.221:5060/sip next=(nil)
nta: received 200 OK for INVITE (10982209)
nta: 200 OK is going to a transaction
nta: 200 OK is duplicate response to 10982209 INVITE
??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
tport_wakeup_pri(0x18c69750): events IN
tport_recv_event(0x18c69750)
tport_recv_iovec(0x18c69750) msg 0x18c87620 from (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
tport_deliver(0x18c69750): msg 0x18c87620 (1235 bytes) from udp/226.59.129.221:5060/sip next=(nil)
nta: received 200 OK for INVITE (10982209)
nta: 200 OK is going to a transaction
nta: 200 OK is duplicate response to 10982209 INVITE
??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
tport_wakeup_pri(0x18c69750): events IN
tport_recv_event(0x18c69750)
tport_recv_iovec(0x18c69750) msg 0x18c87620 from (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
tport_deliver(0x18c69750): msg 0x18c87620 (1235 bytes) from udp/226.59.129.221:5060/sip next=(nil)
nta: received 200 OK for INVITE (10982209)
nta: 200 OK is going to a transaction
nta: 200 OK is duplicate response to 10982209 INVITE
??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
tport_wakeup_pri(0x18c4dcc0): events IN
tport_recv_event(0x18c4dcc0)
tport_recv_iovec(0x18c4dcc0) msg 0x2aaaac060220 from (udp/226.59.129.223:5080) has 876 bytes, veclen = 1
tport_deliver(0x18c4dcc0): msg 0x2aaaac060220 (876 bytes) from udp/226.59.139.61:5080/sip next=(nil)
nta: received INVITE sip:5025155596 at fs2000.lightyear.net SIP/2.0 (CSeq 1)
nta: INVITE (1) going to existing INVITE transaction
nta: re-received INVITE request, retransmitting 100 reply
tport_tsend(0x18c4dcc0) tpn = UDP/226.59.139.61:5060
tport_resolve addrinfo = 226.59.139.61:5060
tport_by_addrinfo(0x18c4dcc0): not found by name UDP/226.59.139.61:5060
tport_vsend(0x18c4dcc0): 397 bytes of 397 to udp/226.59.139.61:5060
tport_vsend returned 397
tport_wakeup_pri(0x18c69750): events IN
tport_recv_event(0x18c69750)
tport_recv_iovec(0x18c69750) msg 0x18c87620 from (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
tport_deliver(0x18c69750): msg 0x18c87620 (1235 bytes) from udp/226.59.129.221:5060/sip next=(nil)
nta: received 200 OK for INVITE (10982209)
nta: 200 OK is going to a transaction
nta: 200 OK is duplicate response to 10982209 INVITE
??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
tport_wakeup_pri(0x18c69750): events IN
tport_recv_event(0x18c69750)
tport_recv_iovec(0x18c69750) msg 0x18c87620 from (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
tport_deliver(0x18c69750): msg 0x18c87620 (1235 bytes) from udp/226.59.129.221:5060/sip next=(nil)
nta: received 200 OK for INVITE (10982209)
nta: 200 OK is going to a transaction
nta: 200 OK is duplicate response to 10982209 INVITE
??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
2011-04-12 15:29:36.002806 [DEBUG] switch_channel.c:2563 (sofia/internal/5596 at 226.59.129.221:5060) Callstate Change RINGING -> HANGUP
2011-04-12 15:29:36.002806 [NOTICE] switch_ivr_originate.c:3329 Hangup sofia/internal/5596 at 226.59.129.221:5060 [CS_CONSUME_MEDIA] [NO_ANSWER]
2011-04-12 15:29:36.002806 [DEBUG] switch_channel.c:2579 Send signal sofia/internal/5596 at 226.59.129.221:5060 [KILL]
2011-04-12 15:29:36.002806 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/5596 at 226.59.129.221:5060 [BREAK]
2011-04-12 15:29:36.002806 [INFO] mod_dptools.c:2647 Originate Failed.? Cause: NO_ANSWER
2011-04-12 15:29:36.002806 [DEBUG] switch_cpp.cpp:988 sofia/external/1009 at fs2000.lightyear.net destroy/unlink session from object
2011-04-12 15:29:36.002806 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/5596 at 226.59.129.221:5060) Running State Change CS_HANGUP
2011-04-12 15:29:36.002806 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/5596 at 226.59.129.221:5060) State HANGUP
EXECUTE sofia/external/1009 at fs2000.lightyear.net answer()
2011-04-12 15:29:36.003897 [DEBUG] sofia_glue.c:3014 AUDIO RTP [sofia/external/1009 at fs2000.lightyear.net] 226.59.129.223 port 27272 -> 226.59.139.61 port 49420 codec: 3 ms: 20
nua: nua_handle_magic: entering
2011-04-12 15:29:36.003897 [DEBUG] switch_rtp.c:1623 Starting timer [soft] 160 bytes per 20ms
nua: nua_application_event: entering
2011-04-12 15:29:36.003897 [DEBUG] mod_sofia.c:457 Channel sofia/internal/5596 at 226.59.129.221:5060 hanging up, cause: NO_ANSWER
2011-04-12 15:29:36.004809 [DEBUG] sofia_glue.c:3276 Set 2833 dtmf send payload to 101
2011-04-12 15:29:36.004809 [DEBUG] sofia_glue.c:3281 Set 2833 dtmf receive payload to 101
2011-04-12 15:29:36.004809 [DEBUG] mod_sofia.c:681 Local SDP sofia/external/1009 at fs2000.lightyear.net:
v=0
o=FreeSWITCH 1302609304 1302609305 IN IP4 226.59.129.223
s=FreeSWITCH
c=IN IP4 226.59.129.223





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From eric at loopfx.com  Wed Apr 13 00:02:29 2011
From: eric at loopfx.com (Eric Beard)
Date: Tue, 12 Apr 2011 16:02:29 -0400
Subject: [Freeswitch-users] Lua session originate - freeswitch crash
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DCA92E03@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DCA92D3D@doctorbob.loop.local>
	<1302633709973-6266239.post@n2.nabble.com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92E03@doctorbob.loop.local>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DCA92EA1@doctorbob.loop.local>

Ok, so I got a little further by trial and error, and realized the 4th arg is due to lua function calling syntax.  If you use the dot notation on a function declared with a colon, it expects "this" as the first arg.

My current code looks like this:

-- Originate an outbound call
local new_session = freeswitch.Session();
new_session:originate(session, "sofia/gateway/affinity/17277762768", 60);
new_session:waitForAnswer(session);
prompt = "/home/eric/test1.wav"
freeswitch.consoleLog("INFO", "About to play prompt file " .. prompt .."\n")
new_session:streamFile(prompt)
new_session:hangup()

The scary thing about this code is that it crashed freeSwitch when I ran it.

This is the last thing I saw in the logs:


-----------------------
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 Eric Beard
Sent: Tuesday, April 12, 2011 3:15 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Lua session originate

Thanks Jeff,

When I try NULL I get this:

freeswitch at internal> 2011-04-12 14:49:31.582049 [ERR] mod_lua.cpp:191 Error in originate (arg 2), ex
pected 'CoreSession *' got 'string'
stack traceback:
        [C]: in function 'originate'
        /usr/local/freeswitch/scripts/test_originate.lua:3: in main chunk

I've been digging through the source code, and I can't find where 4 args are required.  I see this in freeswitch_lua.cpp:

int Session::originate(CoreSession *a_leg_session, char *dest, int timeout)
{
        int x = CoreSession::originate(a_leg_session, dest, timeout);

        if (x) {
                setLUA(L);
        }

        return x;
}

-----------------------
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 Jeff Lenk
Sent: Tuesday, April 12, 2011 2:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Lua session originate

The fourth parameter is the switch_state_handler_table reference and for most
uses(script language) you should just pass null  but I dont use lua so I
dont know the details on that.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Lua-session-originate-tp6266121p6266239.html
Sent from the freeswitch-users mailing list archive at Nabble.com.

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From me at nevian.org  Wed Apr 13 00:07:04 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Wed, 13 Apr 2011 00:07:04 +0400
Subject: [Freeswitch-users] FS HA
In-Reply-To: <AANLkTin-sGbVsCT_GNiu3p_gDT_d7sBPLMTs=wGHJ515@mail.gmail.com>
References: <AANLkTi=_BPuE5KX4D48JDoOW9U3netsr9uDU9Lo5CgYO@mail.gmail.com>
	<FACE40F5-BB80-479A-BFFC-F07F6BA681D2@freeswitch.org>
	<350D44B3-A176-47A9-84F4-72EB62F08299@gmail.com>
	<AANLkTimJLMjPJwthN8AhDq=1_JnDbdmocbgd=-=usyTT@mail.gmail.com>
	<AANLkTi=ksT-ZkdOyXiVQkir7a9z2-58PcVirmv4xs5Kg@mail.gmail.com>
	<AANLkTikeU5FAgE-HU=0a5whf3JiD5VUCkYkCCRt4u=R0@mail.gmail.com>
	<AANLkTim3=HVjX5GZVfcVjtSCm-2kuWatyVS+YV0RpNCB@mail.gmail.com>
	<AANLkTimB-jxEjrSD7pp45DcAmCmZH_q-dycT25c1gn06@mail.gmail.com>
	<AANLkTimwD=847nJ6Lz0gDrUh=nmELW261ch3QBAhDhf-@mail.gmail.com>
	<AANLkTindsD6E7g=fqr0jr6sCtJrJg6M0Hb1P0LGKh6LR@mail.gmail.com>
	<AANLkTikjBm=XT4FrY1N5n_RDQ-TvjANuy149+f4Ai8sH@mail.gmail.com>
	<AANLkTin0Dxzz2wM5Vm2AKDAb2VetTiczq3xAfPhDabfo@mail.gmail.com>
	<AANLkTikoNS+seph5zuKWVom_5d=Xt1m7m0zw5CiiBaDH@mail.gmail.com>
	<AANLkTin-sGbVsCT_GNiu3p_gDT_d7sBPLMTs=wGHJ515@mail.gmail.com>
Message-ID: <254171302638824@web67.yandex.ru>

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From eric at loopfx.com  Wed Apr 13 00:07:07 2011
From: eric at loopfx.com (Eric Beard)
Date: Tue, 12 Apr 2011 16:07:07 -0400
Subject: [Freeswitch-users] Lua session originate - freeswitch crash
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DCA92EA1@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DCA92D3D@doctorbob.loop.local>
	<1302633709973-6266239.post@n2.nabble.com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92E03@doctorbob.loop.local>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92EA1@doctorbob.loop.local>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DCA92EB5@doctorbob.loop.local>

Here is the tail of the logs.

2011-04-12 15:56:07.594051 [DEBUG] switch_channel.c:2656 (sofia/external/17277762768) Callstate Chan
ge RINGING -> EARLY
2011-04-12 15:56:07.598555 [DEBUG] switch_ivr_originate.c:3412 Originate Resulted in Success: [sofia
/external/17277762768]
2011-04-12 15:56:07.598555 [DEBUG] switch_cpp.cpp:1064 (sofia/external/17277762768) State Change CS_
CONSUME_MEDIA -> CS_SOFT_EXECUTE

-----------------------
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 Eric Beard
Sent: Tuesday, April 12, 2011 4:02 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Lua session originate - freeswitch crash

Ok, so I got a little further by trial and error, and realized the 4th arg is due to lua function calling syntax.  If you use the dot notation on a function declared with a colon, it expects "this" as the first arg.

My current code looks like this:

-- Originate an outbound call
local new_session = freeswitch.Session();
new_session:originate(session, "sofia/gateway/affinity/17277762768", 60);
new_session:waitForAnswer(session);
prompt = "/home/eric/test1.wav"
freeswitch.consoleLog("INFO", "About to play prompt file " .. prompt .."\n")
new_session:streamFile(prompt)
new_session:hangup()

The scary thing about this code is that it crashed freeSwitch when I ran it.

This is the last thing I saw in the logs:


-----------------------
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 Eric Beard
Sent: Tuesday, April 12, 2011 3:15 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Lua session originate

Thanks Jeff,

When I try NULL I get this:

freeswitch at internal> 2011-04-12 14:49:31.582049 [ERR] mod_lua.cpp:191 Error in originate (arg 2), ex
pected 'CoreSession *' got 'string'
stack traceback:
        [C]: in function 'originate'
        /usr/local/freeswitch/scripts/test_originate.lua:3: in main chunk

I've been digging through the source code, and I can't find where 4 args are required.  I see this in freeswitch_lua.cpp:

int Session::originate(CoreSession *a_leg_session, char *dest, int timeout)
{
        int x = CoreSession::originate(a_leg_session, dest, timeout);

        if (x) {
                setLUA(L);
        }

        return x;
}

-----------------------
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 Jeff Lenk
Sent: Tuesday, April 12, 2011 2:42 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Lua session originate

The fourth parameter is the switch_state_handler_table reference and for most
uses(script language) you should just pass null  but I dont use lua so I
dont know the details on that.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Lua-session-originate-tp6266121p6266239.html
Sent from the freeswitch-users mailing list archive at Nabble.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

_______________________________________________
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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
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


From me at nevian.org  Wed Apr 13 00:23:26 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Wed, 13 Apr 2011 00:23:26 +0400
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: <267611302639806@web54.yandex.ru>


Hi

Wow, how interesting!
I'm thought the same way as Boris although didn't understand why do it this way and not as seen in examples..
Thanks a lot for explanation.

12.04.2011, 17:47, "Kristian Kielhofner" <kris at kriskinc.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
>
> ?_______________________________________________
> ?FreeSWITCH-users mailing list
> ?FreeSWITCH-users at 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  Wed Apr 13 00:23:41 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Wed, 13 Apr 2011 00:23:41 +0400
Subject: [Freeswitch-users] Full username in caller_profile->username
In-Reply-To: <1067231301666244@web109.yandex.ru>
References: <367151301558892@web113.yandex.ru>
	<AANLkTinYKSEo3jJwEtzduPJpj6zwKXT669baX_6PLxfd@mail.gmail.com>
	<1067231301666244@web109.yandex.ru>
Message-ID: <267691302639821@web54.yandex.ru>


Hi

Anyone?
Maybe there another way to get auth username and his domain?
I see some variables containing domain but is there authed one?

01.04.2011, 17:57, "Serge S. Yuriev" <me at nevian.org>;:

> ?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
>
> ?_______________________________________________
> ?FreeSWITCH-users mailing list
> ?FreeSWITCH-users at 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 peter.olsson at visionutveckling.se  Wed Apr 13 00:55:39 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 12 Apr 2011 22:55:39 +0200
Subject: [Freeswitch-users] Gateway with dynamic IP address
In-Reply-To: <BANLkTims24tpVuEmq1r84XA1bZd02ebfsw@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>
	<BANLkTi=8LfL-CMkczXgJg8FOF1_FTYtxdQ@mail.gmail.com>,
	<BANLkTims24tpVuEmq1r84XA1bZd02ebfsw@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDC7@cooper>

So, it's a user (from FS point of view).. A gateway (from somewhere) registers to your FS instance. Ypu want to dial to this gateway, just dial to it as a normal user. But just try to override the invite uri;

{sip_invite_to_uri=<sip:REAL_NUMBER@${domain_name}>}user/registered_user@${domain_name}

I just wrote this from my head, so it might have errors, but the big idea is to dial the user, but at the same time override the sip invite uri, to a different number.

At least this is what I do in this situation.. :)

/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 12 april 2011 20:26
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] Gateway with dynamic IP address

The problem is this isn't a user, it's a gateway that registers itself
as a user (so FS does not register to it but the other way around). I
need to dial to it as a user and specify the number it has to dial.

On 12 April 2011 12:20, Steven Ayre <steveayre at gmail.com> wrote:
> 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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:4da49a0e32765712911065!



From bwibowo at gmail.com  Wed Apr 13 03:23:37 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Wed, 13 Apr 2011 06:23:37 +0700
Subject: [Freeswitch-users] called number rewrite
Message-ID: <BANLkTikEMcBF-vK4QhTrb6ABw7fkDe=3Zw@mail.gmail.com>

hi
is it possible for freeswitch to change / rewrite called number.
say, i call 1234567 and FS will change the called number to 44444444


thx

budi
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From anthony.minessale at gmail.com  Wed Apr 13 03:27:57 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Apr 2011 18:27:57 -0500
Subject: [Freeswitch-users] Lua session originate - freeswitch crash
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DCA92EA1@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DCA92D3D@doctorbob.loop.local>
	<1302633709973-6266239.post@n2.nabble.com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92E03@doctorbob.loop.local>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92EA1@doctorbob.loop.local>
Message-ID: <BANLkTi=f8cRhf1QL2tb4C28mv3E0MSEY4g@mail.gmail.com>

1) You are not running the latest GIT.  Your scripts works fine on my
development box.
2) the originate method is deprecated.  you can just do
     local new_session =
freeswitch.Session("sofia/gateway/affinity/17277762768", session);
3) In either case you should not be doing what you do in this script
because the whole time you are playing the file to new_session the
original session is blocked not reading or writing any audio.


On Tue, Apr 12, 2011 at 3:02 PM, Eric Beard <eric at loopfx.com> wrote:
> Ok, so I got a little further by trial and error, and realized the 4th arg is due to lua function calling syntax. ?If you use the dot notation on a function declared with a colon, it expects "this" as the first arg.
>
> My current code looks like this:
>
> -- Originate an outbound call
> local new_session = freeswitch.Session();
> new_session:originate(session, "sofia/gateway/affinity/17277762768", 60);
> new_session:waitForAnswer(session);
> prompt = "/home/eric/test1.wav"
> freeswitch.consoleLog("INFO", "About to play prompt file " .. prompt .."\n")
> new_session:streamFile(prompt)
> new_session:hangup()
>
> The scary thing about this code is that it crashed freeSwitch when I ran it.
>
> This is the last thing I saw in the logs:
>
>
> -----------------------
> 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 Eric Beard
> Sent: Tuesday, April 12, 2011 3:15 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Lua session originate
>
> Thanks Jeff,
>
> When I try NULL I get this:
>
> freeswitch at internal> 2011-04-12 14:49:31.582049 [ERR] mod_lua.cpp:191 Error in originate (arg 2), ex
> pected 'CoreSession *' got 'string'
> stack traceback:
> ? ? ? ?[C]: in function 'originate'
> ? ? ? ?/usr/local/freeswitch/scripts/test_originate.lua:3: in main chunk
>
> I've been digging through the source code, and I can't find where 4 args are required. ?I see this in freeswitch_lua.cpp:
>
> int Session::originate(CoreSession *a_leg_session, char *dest, int timeout)
> {
> ? ? ? ?int x = CoreSession::originate(a_leg_session, dest, timeout);
>
> ? ? ? ?if (x) {
> ? ? ? ? ? ? ? ?setLUA(L);
> ? ? ? ?}
>
> ? ? ? ?return x;
> }
>
> -----------------------
> 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 Jeff Lenk
> Sent: Tuesday, April 12, 2011 2:42 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Lua session originate
>
> The fourth parameter is the switch_state_handler_table reference and for most
> uses(script language) you should just pass null ?but I dont use lua so I
> dont know the details on that.
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Lua-session-originate-tp6266121p6266239.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
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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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> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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From gourav at rentec.com  Wed Apr 13 03:55:56 2011
From: gourav at rentec.com (Gourav Vohra)
Date: Tue, 12 Apr 2011 19:55:56 -0400 (EDT)
Subject: [Freeswitch-users] Shared Call appearence, barging and presence
In-Reply-To: <1429335029.215077.1302626935413.JavaMail.root@zinnia1>
Message-ID: <72497602.220103.1302652556513.JavaMail.root@zinnia1>

It would be very helpful to know if this is how freeswitch works with SLA or if my configuration is broken.


Thanks.-

----- Original Message -----
From: "Gourav Vohra" <gourav at rentec.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Tuesday, April 12, 2011 12:48:55 PM
Subject: Shared Call appearence, barging and presence



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, skipping
2011-04-08 10:09:33.381769 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/log/cdr-csv/2995.csv
2011-04-08 10:09:33.381769 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/log/cdr-csv/Master.csv




From anthony.minessale at gmail.com  Wed Apr 13 04:12:26 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Apr 2011 19:12:26 -0500
Subject: [Freeswitch-users] Lua session originate
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DCA92E03@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DCA92D3D@doctorbob.loop.local>
	<1302633709973-6266239.post@n2.nabble.com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92E03@doctorbob.loop.local>
Message-ID: <BANLkTi=5BFK94kAZfN9ZTJWZ=s3Az38stw@mail.gmail.com>

1) You are not running the latest GIT.  Your scripts works fine on my
development box.
2) the originate method is deprecated.  you can just do
    local new_session =
freeswitch.Session("sofia/gateway/affinity/17277762768", session);
3) In either case you should not be doing what you do in this script
because the whole time you are playing the file to new_session the
original session is blocked not reading or writing any audio.

On Tue, Apr 12, 2011 at 2:14 PM, Eric Beard <eric at loopfx.com> wrote:
> Thanks Jeff,
>
> When I try NULL I get this:
>
> freeswitch at internal> 2011-04-12 14:49:31.582049 [ERR] mod_lua.cpp:191 Error in originate (arg 2), ex
> pected 'CoreSession *' got 'string'
> stack traceback:
> ? ? ? ?[C]: in function 'originate'
> ? ? ? ?/usr/local/freeswitch/scripts/test_originate.lua:3: in main chunk
>
> I've been digging through the source code, and I can't find where 4 args are required. ?I see this in freeswitch_lua.cpp:
>
> int Session::originate(CoreSession *a_leg_session, char *dest, int timeout)
> {
> ? ? ? ?int x = CoreSession::originate(a_leg_session, dest, timeout);
>
> ? ? ? ?if (x) {
> ? ? ? ? ? ? ? ?setLUA(L);
> ? ? ? ?}
>
> ? ? ? ?return x;
> }
>
> -----------------------
> 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 Jeff Lenk
> Sent: Tuesday, April 12, 2011 2:42 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Lua session originate
>
> The fourth parameter is the switch_state_handler_table reference and for most
> uses(script language) you should just pass null ?but I dont use lua so I
> dont know the details on that.
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Lua-session-originate-tp6266121p6266239.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
>



-- 
Anthony Minessale II

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 elijah at crankenstein.com  Wed Apr 13 04:16:46 2011
From: elijah at crankenstein.com (elijah)
Date: Tue, 12 Apr 2011 17:16:46 -0700
Subject: [Freeswitch-users] error loading mod_file_string
In-Reply-To: <1302633756754-6266242.post@n2.nabble.com>
References: <BANLkTikZh1v5uHyOaMDpOtNH_FHNSZG2Kg@mail.gmail.com>
	<BANLkTikz-qyJFFuH5D_5G6Aa1G5-UB20+g@mail.gmail.com>
	<1302633756754-6266242.post@n2.nabble.com>
Message-ID: <BANLkTikeqgqzXR4yDQZ2JGJDBfZtH3J32Q@mail.gmail.com>

I'll remove it from modules.conf

thank you

On Tue, Apr 12, 2011 at 11:42 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

> that module has been removed - no longer needed.
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/error-loading-mod-file-string-tp6266207p6266242.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 anthony.minessale at gmail.com  Wed Apr 13 04:21:18 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Apr 2011 19:21:18 -0500
Subject: [Freeswitch-users] FS does not repose to SIP OK message.
In-Reply-To: <fc6f8b8d15ce3.4da47716@insightbb.com>
References: <fc6f8b8d15ce3.4da47716@insightbb.com>
Message-ID: <BANLkTimmd8TDnH2D21Sim-Tz=h6yoTda=A@mail.gmail.com>

The ack is not being received.  Try your trace from the other side.
Try finding the misconfiguration and the NAT or SIP alg on your network.




On Tue, Apr 12, 2011 at 3:00 PM, Gary Chen <gchen00 at insightbb.com> wrote:
> Just update my test FS to newest snapshot: FreeSWITCH Version 1.0.head
> (git-5310735 2011-04-07 15-47-30 -0500)
> I am using SJphone softphone to call into my test FS1. This FS1 then forward
> the call to another FS2.
> FS2 will answer the call and start Music On Hold.? Basically I am using
> SJphone to initiate a SIP call? and let FS2 to answer the call with Music On
> Hold.
> It is working with older version of FS. Now after update to this newest
> version. The call sometime will not go through. The SJPhone just keep
> ringing until timeout. This happens maybe on half of the calls. I also tried
> to use Asterisk to replace FS2 for Music On Hold and it did the same thing.
> The following is the part of console sofia log info:
> 2011-04-12 15:29:06.285484 [DEBUG] mod_sofia.c:84
> sofia/internal/5596 at 226.59.129.221:5060 SOFIA INIT
> nua: nh_create_handle: entering
> nua: nua_handle_bind: entering
> nua: nua_invite: entering
> nua(0x18cc2c80): sent signal r_invite
> 2011-04-12 15:29:06.285484 [DEBUG] mod_sofia.c:124
> (sofia/internal/5596 at 226.59.129.221:5060) State Change CS_INIT -> CS_ROUTING
> 2011-04-12 15:29:06.285484 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/5596 at 226.59.129.221:5060 [BREAK]
> 2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:361
> (sofia/internal/5596 at 226.59.129.221:5060) State INIT going to sleep
> 2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/5596 at 226.59.129.221:5060) Running State Change CS_ROUTING
> nua(0x18cc2c80): recv signal r_invite
> 2011-04-12 15:29:06.285484 [DEBUG] switch_channel.c:1668
> (sofia/internal/5596 at 226.59.129.221:5060) Callstate Change DOWN -> RINGING
> nua: nua_stack_set_params: entering
> soa_clone(static::0x18c66f60, 0x18c474e0, 0x18cc2c80) called
> soa_set_params(static::0x2aaabc072a60, ...) called
> soa_set_params(static::0x2aaabc072a60, ...) called
> soa_set_user_sdp(static::0x2aaabc072a60, (nil), 0x18ccb7b7, -1) called
> soa_set_capability_sdp(static::0x2aaabc072a60, (nil), 0x18ccb7b7, -1) called
> 2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:364
> (sofia/internal/5596 at 226.59.129.221:5060) State ROUTING
> 2011-04-12 15:29:06.285484 [DEBUG] mod_sofia.c:147
> sofia/internal/5596 at 226.59.129.221:5060 SOFIA ROUTING
> 2011-04-12 15:29:06.285484 [DEBUG] switch_ivr_originate.c:66
> (sofia/internal/5596 at 226.59.129.221:5060) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> nua(0x18cc2c80): adding session usage
> 2011-04-12 15:29:06.285484 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/5596 at 226.59.129.221:5060 [BREAK]
> 2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:364
> (sofia/internal/5596 at 226.59.129.221:5060) State ROUTING going to sleep
> 2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/5596 at 226.59.129.221:5060) Running State Change
> CS_CONSUME_MEDIA
> nta_leg_tcreate(0x2aaaac05e220)
> soa_init_offer_answer(static::0x2aaabc072a60) called
> soa_generate_offer(static::0x2aaabc072a60, 0) called
> soa_static_offer_answer_action(0x2aaabc072a60, soa_generate_offer): called
> soa_static(0x2aaabc072a60, soa_generate_offer): generating local description
> soa_static(0x2aaabc072a60, soa_generate_offer): upgrade with local
> description
> 2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:383
> (sofia/internal/5596 at 226.59.129.221:5060) State CONSUME_MEDIA
> soa_sdp_mode_set(0x406f6c60, (nil), ""): called
> 2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:383
> (sofia/internal/5596 at 226.59.129.221:5060) State CONSUME_MEDIA going to sleep
> soa_static(0x2aaabc072a60, soa_generate_offer): storing local description
> soa_get_local_sdp(static::0x2aaabc072a60, [(nil)], [0x406f6dc8],
> [0x406f6dd4]) called
> nta: selecting scheme sip
> tport_tsend(0x18c69750) tpn = */226.59.129.221:5060
> tport_resolve addrinfo = 226.59.129.221:5060
> tport_by_addrinfo(0x18c69750): not found by name */226.59.129.221:5060
> tport_vsend(0x18c69750): 1245 bytes of 1245 to udp/226.59.129.221:5060
> tport_vsend returned 1245
> nta: sent INVITE (10982209) to */226.59.129.221:5060
> tport_pend(0x18c69750): pending 0x2aaaac059c30 for udp/226.59.129.223:5060
> (already 0)
> nta: timer set to 32000 ms
> nta: timer shortened to 1000 ms
> nua(0x18cc2c80): call state changed: init -> calling, sent offer
> soa_get_local_sdp(static::0x2aaabc072a60, [0x406f6db8], [0x406f6db0],
> [(nil)]) called
> nua(0x18cc2c80): event i_state INVITE sent
> nua: nua_application_event: entering
> 2011-04-12 15:29:06.285484 [DEBUG] sofia.c:4761 Channel
> sofia/internal/5596 at 226.59.129.221:5060 entering state [calling][0]
> nua: nua_handle_magic: entering
> tport_wakeup_pri(0x18c69750): events IN
> tport_recv_event(0x18c69750)
> tport_recv_iovec(0x18c69750) msg 0x18c87620 from (udp/226.59.129.223:5060)
> has 344 bytes, veclen = 1
> tport_deliver(0x18c69750): msg 0x18c87620 (344 bytes) from
> udp/226.59.129.221:5060/sip next=(nil)
> nta: received 100 Trying for INVITE (10982209)
> nta: 100 Trying is going to a transaction
> nta_outgoing: RTT is 0.765 ms
> tport_release(0x18c69750): 0x2aaaac059c30 by 0x2aaabc05da60 with 0x18c87620
> (preliminary)
> tport_wakeup_pri(0x18c69750): events IN
> tport_recv_event(0x18c69750)
> tport_recv_iovec(0x18c69750) msg 0x2aaabc070b20 from
> (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
> tport_deliver(0x18c69750): msg 0x2aaabc070b20 (1235 bytes) from
> udp/226.59.129.221:5060/sip next=(nil)
> nta: received 200 OK for INVITE (10982209)
> nta: 200 OK is going to a transaction
> tport_release(0x18c69750): 0x2aaaac059c30 by 0x2aaabc05da60 with
> 0x2aaabc070b20
> soa_set_remote_sdp(static::0x2aaabc072a60, (nil), 0x2aaabc0712ac, 247)
> called
> soa_process_answer(static::0x2aaabc072a60) called
> soa_static_offer_answer_action(0x2aaabc072a60, soa_process_answer): called
> soa_sdp_mode_set(0x2aaabc070240, 0x2aaabc073500, ""): called
> soa_static(0x2aaabc072a60, soa_process_answer): upgrade codecs with remote
> description
> soa_static(0x2aaabc072a60, soa_process_answer): storing local description
> soa_activate(static::0x2aaabc072a60, (nil)) called
> nua(0x18cc2c80): INVITE: processed SDP answer in 200 OK
> nua(0x18cc2c80): event r_invite 200 OK
> nua(0x18cc2c80): call state changed: calling -> completing, received answer
> soa_get_remote_sdp(static::0x2aaabc072a60, [0x406f6828], [0x406f6820],
> [(nil)]) called
> soa_get_params(static::0x2aaabc072a60, ...) called
> nua: nua_application_event: entering
> nua(0x18cc2c80): event i_state 200 OK
> 2011-04-12 15:29:06.294311 [INFO] sofia.c:740
> sofia/internal/5596 at 226.59.129.221:5060 Update Callee ID to "5596"
> <sip:5596 at 226.59.129.221>
> nta: timer not set
> tport_wakeup_pri(0x18c69750): events IN
> tport_recv_event(0x18c69750)
> tport_recv_iovec(0x18c69750) msg 0x2aaab4022620 from
> (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
> tport_deliver(0x18c69750): msg 0x2aaab4022620 (1235 bytes) from
> udp/226.59.129.221:5060/sip next=(nil)
> nta: received 200 OK for INVITE (10982209)
> nta: 200 OK is going to a transaction
> nta: 200 OK is duplicate response to 10982209 INVITE
> ??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
> nta: timer set next to 31009 ms
> tport_wakeup_pri(0x18c69750): events IN
> tport_recv_event(0x18c69750)
> tport_recv_iovec(0x18c69750) msg 0x2aaab4022620 from
> (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
> tport_deliver(0x18c69750): msg 0x2aaab4022620 (1235 bytes) from
> udp/226.59.129.221:5060/sip next=(nil)
> nta: received 200 OK for INVITE (10982209)
> nta: 200 OK is going to a transaction
> nta: 200 OK is duplicate response to 10982209 INVITE
> ??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
> tport_wakeup_pri(0x18c69750): events IN
> tport_recv_event(0x18c69750)
> tport_recv_iovec(0x18c69750) msg 0x2aaaac060220 from
> (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
> tport_deliver(0x18c69750): msg 0x2aaaac060220 (1235 bytes) from
> udp/226.59.129.221:5060/sip next=(nil)
> nta: received 200 OK for INVITE (10982209)
> nta: 200 OK is going to a transaction
> nta: 200 OK is duplicate response to 10982209 INVITE
> ??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
> tport_wakeup_pri(0x18c69750): events IN
> tport_recv_event(0x18c69750)
> tport_recv_iovec(0x18c69750) msg 0x2aaaac060220 from
> (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
> tport_deliver(0x18c69750): msg 0x2aaaac060220 (1235 bytes) from
> udp/226.59.129.221:5060/sip next=(nil)
> nta: received 200 OK for INVITE (10982209)
> nta: 200 OK is going to a transaction
> nta: 200 OK is duplicate response to 10982209 INVITE
> ??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
> tport_wakeup_pri(0x18c69750): events IN
> tport_recv_event(0x18c69750)
> tport_recv_iovec(0x18c69750) msg 0x2aaaac060220 from
> (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1
> tport_deliver(0x18c69750): msg 0x2aaaac060220 (1235 bytes) from
> udp/226.59.129.221:5060/sip next=(nil)
> nta: received 200 OK for INVITE (10982209)
> nta: 200 OK is going to a transaction
> nta: 200 OK is duplicate response to 10982209 INVITE
> ??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
> tport_wakeup_pri(0x18c69750): events IN
> tport_recv_event(0x18c69750)
> tport_recv_iovec(0x18c69750) msg 0x18c87620 from (udp/226.59.129.223:5060)
> has 1235 bytes, veclen = 1
> tport_deliver(0x18c69750): msg 0x18c87620 (1235 bytes) from
> udp/226.59.129.221:5060/sip next=(nil)
> nta: received 200 OK for INVITE (10982209)
> nta: 200 OK is going to a transaction
> nta: 200 OK is duplicate response to 10982209 INVITE
> ??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
> tport_wakeup_pri(0x18c69750): events IN
> tport_recv_event(0x18c69750)
> tport_recv_iovec(0x18c69750) msg 0x18c87620 from (udp/226.59.129.223:5060)
> has 1235 bytes, veclen = 1
> tport_deliver(0x18c69750): msg 0x18c87620 (1235 bytes) from
> udp/226.59.129.221:5060/sip next=(nil)
> nta: received 200 OK for INVITE (10982209)
> nta: 200 OK is going to a transaction
> nta: 200 OK is duplicate response to 10982209 INVITE
> ??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
> tport_wakeup_pri(0x18c69750): events IN
> tport_recv_event(0x18c69750)
> tport_recv_iovec(0x18c69750) msg 0x18c87620 from (udp/226.59.129.223:5060)
> has 1235 bytes, veclen = 1
> tport_deliver(0x18c69750): msg 0x18c87620 (1235 bytes) from
> udp/226.59.129.221:5060/sip next=(nil)
> nta: received 200 OK for INVITE (10982209)
> nta: 200 OK is going to a transaction
> nta: 200 OK is duplicate response to 10982209 INVITE
> ??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
> tport_wakeup_pri(0x18c4dcc0): events IN
> tport_recv_event(0x18c4dcc0)
> tport_recv_iovec(0x18c4dcc0) msg 0x2aaaac060220 from
> (udp/226.59.129.223:5080) has 876 bytes, veclen = 1
> tport_deliver(0x18c4dcc0): msg 0x2aaaac060220 (876 bytes) from
> udp/226.59.139.61:5080/sip next=(nil)
> nta: received INVITE sip:5025155596 at fs2000.lightyear.net SIP/2.0 (CSeq 1)
> nta: INVITE (1) going to existing INVITE transaction
> nta: re-received INVITE request, retransmitting 100 reply
> tport_tsend(0x18c4dcc0) tpn = UDP/226.59.139.61:5060
> tport_resolve addrinfo = 226.59.139.61:5060
> tport_by_addrinfo(0x18c4dcc0): not found by name UDP/226.59.139.61:5060
> tport_vsend(0x18c4dcc0): 397 bytes of 397 to udp/226.59.139.61:5060
> tport_vsend returned 397
> tport_wakeup_pri(0x18c69750): events IN
> tport_recv_event(0x18c69750)
> tport_recv_iovec(0x18c69750) msg 0x18c87620 from (udp/226.59.129.223:5060)
> has 1235 bytes, veclen = 1
> tport_deliver(0x18c69750): msg 0x18c87620 (1235 bytes) from
> udp/226.59.129.221:5060/sip next=(nil)
> nta: received 200 OK for INVITE (10982209)
> nta: 200 OK is going to a transaction
> nta: 200 OK is duplicate response to 10982209 INVITE
> ??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
> tport_wakeup_pri(0x18c69750): events IN
> tport_recv_event(0x18c69750)
> tport_recv_iovec(0x18c69750) msg 0x18c87620 from (udp/226.59.129.223:5060)
> has 1235 bytes, veclen = 1
> tport_deliver(0x18c69750): msg 0x18c87620 (1235 bytes) from
> udp/226.59.129.221:5060/sip next=(nil)
> nta: received 200 OK for INVITE (10982209)
> nta: 200 OK is going to a transaction
> nta: 200 OK is duplicate response to 10982209 INVITE
> ??????? Via: SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg
> 2011-04-12 15:29:36.002806 [DEBUG] switch_channel.c:2563
> (sofia/internal/5596 at 226.59.129.221:5060) Callstate Change RINGING -> HANGUP
> 2011-04-12 15:29:36.002806 [NOTICE] switch_ivr_originate.c:3329 Hangup
> sofia/internal/5596 at 226.59.129.221:5060 [CS_CONSUME_MEDIA] [NO_ANSWER]
> 2011-04-12 15:29:36.002806 [DEBUG] switch_channel.c:2579 Send signal
> sofia/internal/5596 at 226.59.129.221:5060 [KILL]
> 2011-04-12 15:29:36.002806 [DEBUG] switch_core_session.c:1116 Send signal
> sofia/internal/5596 at 226.59.129.221:5060 [BREAK]
> 2011-04-12 15:29:36.002806 [INFO] mod_dptools.c:2647 Originate Failed.
> Cause: NO_ANSWER
> 2011-04-12 15:29:36.002806 [DEBUG] switch_cpp.cpp:988
> sofia/external/1009 at fs2000.lightyear.net destroy/unlink session from object
> 2011-04-12 15:29:36.002806 [DEBUG] switch_core_state_machine.c:325
> (sofia/internal/5596 at 226.59.129.221:5060) Running State Change CS_HANGUP
> 2011-04-12 15:29:36.002806 [DEBUG] switch_core_state_machine.c:565
> (sofia/internal/5596 at 226.59.129.221:5060) State HANGUP
> EXECUTE sofia/external/1009 at fs2000.lightyear.net answer()
> 2011-04-12 15:29:36.003897 [DEBUG] sofia_glue.c:3014 AUDIO RTP
> [sofia/external/1009 at fs2000.lightyear.net] 226.59.129.223 port 27272 ->
> 226.59.139.61 port 49420 codec: 3 ms: 20
> nua: nua_handle_magic: entering
> 2011-04-12 15:29:36.003897 [DEBUG] switch_rtp.c:1623 Starting timer [soft]
> 160 bytes per 20ms
> nua: nua_application_event: entering
> 2011-04-12 15:29:36.003897 [DEBUG] mod_sofia.c:457 Channel
> sofia/internal/5596 at 226.59.129.221:5060 hanging up, cause: NO_ANSWER
> 2011-04-12 15:29:36.004809 [DEBUG] sofia_glue.c:3276 Set 2833 dtmf send
> payload to 101
> 2011-04-12 15:29:36.004809 [DEBUG] sofia_glue.c:3281 Set 2833 dtmf receive
> payload to 101
> 2011-04-12 15:29:36.004809 [DEBUG] mod_sofia.c:681 Local SDP
> sofia/external/1009 at fs2000.lightyear.net:
> v=0
> o=FreeSWITCH 1302609304 1302609305 IN IP4 226.59.129.223
> s=FreeSWITCH
> c=IN IP4 226.59.129.223
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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 elijah at crankenstein.com  Wed Apr 13 04:33:30 2011
From: elijah at crankenstein.com (elijah)
Date: Tue, 12 Apr 2011 17:33:30 -0700
Subject: [Freeswitch-users] occasional ~5s delay during bind_meta_app
	execute_extenstion
Message-ID: <BANLkTikxT9VuejCLOUsbKCncLXv6HiEWiw@mail.gmail.com>

I have setup a feature where users who have dialed outbound have the option
to transfer the b-leg to a separate outbound number, like so:

    <extension name="outbound_9">
      <condition field="destination_number" expression="^9(\d{10})$">
        <action application="set" data="ringback=${us-ring}"/>
        <action application="set" data="transfer_ringback=$${hold_music}"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="bind_meta_key=*"/>
        <action application="bind_meta_app" data="1 a s
execute_extension::dx XML mycontext"/>
        <action application="transfer" data="to_outbound XML mycontext"/>
      </condition>
    </extension>

    <extension name="dx">
      <condition field="destination_number" expression="^dx$">
        <action application="log" data="INFO transferring the b-leg..."/>
        <action application="answer"/>
        <action application="sleep" data="1000"/>
        <action application="playback"
data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/transfer2.wav"/>
        <action application="read" data="11 11
'tone_stream://%(10000,0,350,440)' digits 5000 #"/>
        <action application="execute_extension" data="is_transfer XML
mycontext"/>
      </condition>
    </extension>

A delay sometimes occurs after users have pressed *1 and before the message
"transferring the b-leg..." appears in the console. This delay happens ~1/3
of the time and lasts for ~5s. Do you know how I should execute this
functionality in a way that will not cause such a delay?

thanks,
elijah
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From anthony.minessale at gmail.com  Wed Apr 13 04:44:42 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 12 Apr 2011 19:44:42 -0500
Subject: [Freeswitch-users] occasional ~5s delay during bind_meta_app
	execute_extenstion
In-Reply-To: <BANLkTikxT9VuejCLOUsbKCncLXv6HiEWiw@mail.gmail.com>
References: <BANLkTikxT9VuejCLOUsbKCncLXv6HiEWiw@mail.gmail.com>
Message-ID: <BANLkTin+HLFD6dnc=zWs_P-gftsgmpP+vw@mail.gmail.com>

your digit timeout in your read happens to be 5 seconds so maybe you
are not dialing the #

On Tue, Apr 12, 2011 at 7:33 PM, elijah <elijah at crankenstein.com> wrote:
> I have setup a feature where users who have dialed outbound have the option
> to transfer the b-leg to a?separate?outbound number, like so:
> ?? ?<extension name="outbound_9">
> ?? ? ?<condition field="destination_number" expression="^9(\d{10})$">
> ?? ? ? ?<action application="set" data="ringback=${us-ring}"/>
> ?? ? ? ?<action application="set" data="transfer_ringback=$${hold_music}"/>
> ?? ? ? ?<action application="set" data="continue_on_fail=true"/>
> ?? ? ? ?<action application="set" data="bind_meta_key=*"/>
> ?? ? ? ?<action application="bind_meta_app" data="1 a s
> execute_extension::dx XML mycontext"/>
> ?? ? ? ?<action application="transfer" data="to_outbound XML?mycontext"/>
> ?? ? ?</condition>
> ?? ?</extension>
> ?? ?<extension name="dx">
> ?? ? ?<condition field="destination_number" expression="^dx$">
> ?? ? ? ?<action application="log" data="INFO transferring the b-leg..."/>
> ?? ? ? ?<action application="answer"/>
> ?? ? ? ?<action application="sleep" data="1000"/>
> ?? ? ? ?<action application="playback"
> data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/transfer2.wav"/>
> ?? ? ? ?<action application="read" data="11 11
> 'tone_stream://%(10000,0,350,440)' digits 5000 #"/>
> ?? ? ? ?<action application="execute_extension" data="is_transfer XML
> mycontext"/>
> ?? ? ?</condition>
> ?? ?</extension>
> A delay sometimes occurs after users have pressed *1 and before the message
> "transferring the b-leg..." appears in the console. This delay happens ~1/3
> of the time and lasts for ~5s. Do you know how I should execute this
> functionality in a way that will not cause such a delay?
> thanks,
> elijah
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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 krice at freeswitch.org  Wed Apr 13 04:52:46 2011
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 12 Apr 2011 19:52:46 -0500
Subject: [Freeswitch-users] called number rewrite
In-Reply-To: <BANLkTikEMcBF-vK4QhTrb6ABw7fkDe=3Zw@mail.gmail.com>
Message-ID: <C9CA5E0E.135DE%krice@freeswitch.org>

Sure... Just have freeswitch change it

<extension name=?foo?>
    <condition field=?destination_number? expression=?^1234567$?>
        <action application=?transfer? data=?987654 XML default?/>
                <!-- OR -->
       <action application=?bridge?
data=?sofia/default/10203040 at somehost.com?/>
  </condition>
</extension>

Depending on what you are trying to do these will rewrite the destination
number

On 4/12/11 6:23 PM, "budi wibowo" <bwibowo at gmail.com> wrote:

> hi?
> is it possible for freeswitch to change / rewrite called number.?
> say, i call 1234567 and FS will change the called number to 44444444?
> 
> 
> thx
> 
> budi?
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 brian at freeswitch.org  Wed Apr 13 04:53:30 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Apr 2011 19:53:30 -0500
Subject: [Freeswitch-users] Shared Call appearence, barging and presence
In-Reply-To: <72497602.220103.1302652556513.JavaMail.root@zinnia1>
References: <72497602.220103.1302652556513.JavaMail.root@zinnia1>
Message-ID: <CBC89A18-0BC1-4014-88F8-33A393D1BF4C@freeswitch.org>

I would need to see your polycom config options I have tested with 3.3.1 and it works fine... so please provide me some info on how you setup the phones... also please don't hijack threads.  Please click new message and input the list address.  You have clicked reply on the CALL_REJECT thread and changed the subject.

Thanks,
Brian

On Apr 12, 2011, at 6:55 PM, Gourav Vohra wrote:

> It would be very helpful to know if this is how freeswitch works with SLA or if my configuration is broken.
> 
> 
> Thanks.-
> 
> ----- Original Message -----
> From: "Gourav Vohra" <gourav at rentec.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Tuesday, April 12, 2011 12:48:55 PM
> Subject: Shared Call appearence, barging and presence
> 
> 
> 
> 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, skipping
> 2011-04-08 10:09:33.381769 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/log/cdr-csv/2995.csv
> 2011-04-08 10:09:33.381769 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/log/cdr-csv/Master.csv
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Wed Apr 13 04:55:14 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 12 Apr 2011 19:55:14 -0500
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: <1BDDB8AD-4CBD-4515-A7AD-693A5E875523@freeswitch.org>

We aren't a proxy... we have transcended into this quasi proxy in some scenarios which mostly involve t.38... as for proxy media DO NOT USE IT.  Just saying it might go away since the purpose of it is now not needed since we have full t.38.

Thanks,
Brian

On Apr 12, 2011, at 10:04 AM, Sam wrote:

> I have done that, but i want to pass the exact SDP what i get from leg A to
> leg B
> 
> regards
> Sam



From elijah at crankenstein.com  Wed Apr 13 04:57:26 2011
From: elijah at crankenstein.com (elijah)
Date: Tue, 12 Apr 2011 17:57:26 -0700
Subject: [Freeswitch-users] occasional ~5s delay during bind_meta_app
	execute_extenstion
In-Reply-To: <BANLkTin+HLFD6dnc=zWs_P-gftsgmpP+vw@mail.gmail.com>
References: <BANLkTikxT9VuejCLOUsbKCncLXv6HiEWiw@mail.gmail.com>
	<BANLkTin+HLFD6dnc=zWs_P-gftsgmpP+vw@mail.gmail.com>
Message-ID: <BANLkTinnpJrLAg+XWs1izQ0MKE_ym+PVcQ@mail.gmail.com>

I'm afraid that cannot be the cause. I'm experiencing the delay before the
read is executed, immediately after a user presses *1 but before the
extension named 'dx' executes anything.


On Tue, Apr 12, 2011 at 5:44 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> your digit timeout in your read happens to be 5 seconds so maybe you
> are not dialing the #
>
> On Tue, Apr 12, 2011 at 7:33 PM, elijah <elijah at crankenstein.com> wrote:
> > I have setup a feature where users who have dialed outbound have the
> option
> > to transfer the b-leg to a separate outbound number, like so:
> >     <extension name="outbound_9">
> >       <condition field="destination_number" expression="^9(\d{10})$">
> >         <action application="set" data="ringback=${us-ring}"/>
> >         <action application="set"
> data="transfer_ringback=$${hold_music}"/>
> >         <action application="set" data="continue_on_fail=true"/>
> >         <action application="set" data="bind_meta_key=*"/>
> >         <action application="bind_meta_app" data="1 a s
> > execute_extension::dx XML mycontext"/>
> >         <action application="transfer" data="to_outbound XML mycontext"/>
> >       </condition>
> >     </extension>
> >     <extension name="dx">
> >       <condition field="destination_number" expression="^dx$">
> >         <action application="log" data="INFO transferring the b-leg..."/>
> >         <action application="answer"/>
> >         <action application="sleep" data="1000"/>
> >         <action application="playback"
> >
> data="/usr/local/freeswitch/sounds/en/us/callie/misc/8000/transfer2.wav"/>
> >         <action application="read" data="11 11
> > 'tone_stream://%(10000,0,350,440)' digits 5000 #"/>
> >         <action application="execute_extension" data="is_transfer XML
> > mycontext"/>
> >       </condition>
> >     </extension>
> > A delay sometimes occurs after users have pressed *1 and before the
> message
> > "transferring the b-leg..." appears in the console. This delay happens
> ~1/3
> > of the time and lasts for ~5s. Do you know how I should execute this
> > functionality in a way that will not cause such a delay?
> > thanks,
> > elijah
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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 bwibowo at gmail.com  Wed Apr 13 05:09:26 2011
From: bwibowo at gmail.com (Budi wibowo)
Date: Wed, 13 Apr 2011 01:09:26 +0000
Subject: [Freeswitch-users] called number rewrite
In-Reply-To: <C9CA5E0E.135DE%krice@freeswitch.org>
References: <BANLkTikEMcBF-vK4QhTrb6ABw7fkDe=3Zw@mail.gmail.com><C9CA5E0E.135DE%krice@freeswitch.org>
Message-ID: <171211778-1302657000-cardhu_decombobulator_blackberry.rim.net-377026527-@b4.c2.bise3.blackberry>

Thx anybody know how many mapping/rewrite can be performed by fs?
(Based on experience)

-----Original Message-----
From: Ken Rice <krice at freeswitch.org>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Tue, 12 Apr 2011 19:52:46 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] called number rewrite

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




From krice at freeswitch.org  Wed Apr 13 05:23:23 2011
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 12 Apr 2011 20:23:23 -0500
Subject: [Freeswitch-users] called number rewrite
In-Reply-To: <171211778-1302657000-cardhu_decombobulator_blackberry.rim.net-377026527-@b4.c2.bise3.blackberry>
Message-ID: <C9CA653B.135E6%krice@freeswitch.org>

Freeswitch is not a proxy, its a B2BUA... So there is speed penalties
there... 

Many people are hesitant to quote numbers as performance is directly tied to
your specific configuration...


On 4/12/11 8:09 PM, "Budi wibowo" <bwibowo at gmail.com> wrote:

> Thx anybody know how many mapping/rewrite can be performed by fs?
> (Based on experience)
> 
> -----Original Message-----
> From: Ken Rice <krice at freeswitch.org>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Tue, 12 Apr 2011 19:52:46
> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] called number rewrite
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
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> http://www.freeswitch.org




From infos at madovsky.org  Wed Apr 13 06:31:12 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 12 Apr 2011 22:31:12 -0400
Subject: [Freeswitch-users] how to pass arbitrary headers from A leg toB
	leg when bridging
References: <047B4BA6-39DE-4BB4-BFED-7353A6EBBB8B@dchorton.com>
Message-ID: <8924541742684F17A2A226B05C9F1D97@e1705>

maybe opensips would be more adapted for your needs

----- Original Message ----- 
From: "Dave Horton" <dave at dchorton.com>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, April 12, 2011 12:49 PM
Subject: Re: [Freeswitch-users] how to pass arbitrary headers from A leg toB 
leg when bridging


> So I'm guessing this isn't possible without hacking the source code, which 
> I've already done to solve my problem for now.
>
> But I'd like to make sure there isn't a better way of doing things, and 
> thus I'd like to revise and restate my question for clarity.  First, 
> though, let me describe what I am doing, because I think it's a 
> not-uncommon scenario that I think would be something that others may want 
> to do.  I basically want to use FS as a simple transcoding server between 
> two endpoints, call them A and B.  Calls coming in from A will be using 
> speex codec and I want to send them out to B using PCMU; calls coming in 
> from B will be PCMU and I want to send them to A using speex.  The FS 
> server will be a B2BUA and will be doing transcoding only -- no 
> authentication (and no registration).  Simple, right?  The only fly in the 
> ointment is that A is authenticating calls with B by providing a 
> Proxy-Authorization header.  So I need to take the Proxy-Authorization 
> header received on the A leg and include it on the B leg.
>
> So far, the only way I have found to do that is to hack the code to create 
> a new channel variable.  I've done this, and it works.  However, this 
> leads me to the following questions
>
> 1) Is there a better way to do this?  If there is no way to do it as a 
> dialplan out of the box, can it be done as a script?
>
> 2) sofia has parsed all of the sip headers on the incoming invite for us, 
> and they're all available from mod_sofia.  Shouldn't those all be 
> available to us (i.e., application developers) by some means (i.e., 
> channel variables)?
>
> Dave
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From gourav at rentec.com  Wed Apr 13 06:59:15 2011
From: gourav at rentec.com (Gourav Vohra)
Date: Tue, 12 Apr 2011 22:59:15 -0400 (EDT)
Subject: [Freeswitch-users] Shared Call appearence, barging and presence
In-Reply-To: <CBC89A18-0BC1-4014-88F8-33A393D1BF4C@freeswitch.org>
Message-ID: <1663096009.221390.1302663555843.JavaMail.root@zinnia1>


Brian,

Thanks for your help.

Following configs on phone2 and phone3.  
 
Following is the phone2 polycom config. x2995 is on this phone.

<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- Generated reg-advanced.cfg Configuration File -->
<polycomConfig xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="polycomConfig.xsd">
  <call call.transferOnConferenceEnd="1">
    <call.autoAnswer call.autoAnswer.H323="0" call.autoAnswer.SIP="0" call.autoAnswer.micMute="1" call.autoAnswer.ringClass="ringAutoAnswer" call.autoAnswer.videoMute="0">
    </call.autoAnswer>
    <call.autoOffHook call.autoOffHook.1.contact="" call.autoOffHook.1.enabled="0" call.autoOffHook.1.protocol="" call.autoOffHook.2.contact="" call.autoOffHook.2.enabled="0" call.autoOffHook.2.protocol="">
    </call.autoOffHook>
    <call.donotdisturb call.donotdisturb.perReg="0">
    </call.donotdisturb>
    <call.missedCallTracking call.missedCallTracking.1.enabled="1" call.missedCallTracking.2.enabled="1">
    </call.missedCallTracking>
    <call.serverMissedCall call.serverMissedCall.1.enabled="0" call.serverMissedCall.2.enabled="0">
    </call.serverMissedCall>
  </call>
  <device>
    <device.auth device.auth.localUserPassword="">
      <device.auth.localUserPassword device.auth.localUserPassword.set="0">
      </device.auth.localUserPassword>
    </device.auth>
    <device.em device.em.power="">
      <device.em.power device.em.power.set="0">
      </device.em.power>
    </device.em>
  </device>
<reg reg.1.displayName="test user1" 
        reg.1.address="2995" 
        reg.1.label="2995" 
        reg.1.type="shared" 
        reg.1.auth.userId="2995" 
        reg.1.auth.password="zebra" 
        reg.1.ringType="2" 
        reg.1.bargeInEnabled="1"
        reg.1.callsPerLineKey="4" 
        reg.1.csta="0" 
        reg.1.fwd.busy.contact="" 
        reg.1.fwd.busy.status="0" 
        reg.1.fwd.noanswer.contact="" 
        reg.1.fwd.noanswer.ringCount="0" 
        reg.1.fwd.noanswer.status="0" 
        reg.1.lineKeys="1" 
        reg.1.musicOnHold.uri="" 
        reg.1.outboundProxy.failOver.failBack.timeout="3600" 
        reg.1.outboundProxy.failOver.failRegistrationOn="1" 
        reg.1.outboundProxy.failOver.reRegisterOn="0" 
        reg.1.outboundProxy.address="192.168.100.33"
        reg.1.outboundProxy.port="5060" 
        reg.1.protocol.SIP="1" 
        reg.1.ringType="ringer2" 
        reg.1.serverFeatureControl.cf="1" 
        reg.1.serverFeatureControl.dnd="1" 
        reg.1.thirdPartyName="" 
        reg.1.type="private" 
        reg.1.useCompleteUriForRetrieve="1" 
        reg.1.server.1.expires="3600" 
        reg.1.server.1.expires.lineSeize="30" 
        reg.1.server.1.expires.overlap="60" 
        reg.1.server.1.lcs="0" 
        reg.1.server.1.retryMaxCount="3" 
        reg.1.server.1.retryTimeOut="0" 
        reg.1.server.2.expires="3600" 
        reg.1.server.2.expires.lineSeize="30" 
        reg.1.server.2.expires.overlap="60" 
        reg.1.server.2.lcs="0" 
        reg.1.server.2.retryMaxCount="3" 
        reg.1.server.2.retryTimeOut="0">
  </reg>
  <se se.destination="chassis">
    <se.rt>
      <se.rt.modification se.rt.modification.enabled="1">
      </se.rt.modification>
    </se.rt>
  </se>
  <up up.25mm="0" up.accessibilityFeatures="0" up.analogHeadsetOption="0" up.audioMode="0" up.callTypeOrderVoice="0" up.handsfreeMode="1" up.headsetMode="0" up.localClockEnabled="1" up.manualProtocolRouting="1" up.mwiVisible="0" up.onHookDialingEnabled="1" up.useDirectoryNames="0" up.welcomeSoundEnabled="1" up.welcomeSoundOnWarmBootEnabled="0">
    <up.2P5mmJack up.2P5mmJack.mode="0">
    </up.2P5mmJack>
    <up.audioSetup up.audioSetup.auxInput="2" up.audioSetup.auxOutput="2">
    </up.audioSetup>
    <up.backlight up.backlight.idleIntensity="1" up.backlight.onIntensity="3">
    </up.backlight>
    <up.idleBrowser up.idleBrowser.enabled="0">
    </up.idleBrowser>
    <up.manualProtocolRouting up.manualProtocolRouting.softKeys="1">
    </up.manualProtocolRouting>
    <up.offHookAction up.offHookAction.none="0">
    </up.offHookAction>
    <up.pictureFrame up.pictureFrame.folder="" up.pictureFrame.timePerImage="5">
    </up.pictureFrame>
    <up.screenSaver up.screenSaver.enabled="0" up.screenSaver.waitTime="15">
    </up.screenSaver>
    <up.toneControl up.toneControl.bass="0" up.toneControl.treble="0">
    </up.toneControl>
  </up>
  <voIpProt>
    <voIpProt.SIP>
      <voIpProt.SIP.assuredService>
        <voIpProt.SIP.assuredService.namespace>
          <voIpProt.SIP.assuredService.namespace.label>
            <voIpProt.SIP.assuredService.namespace.label.UCRdrsn voIpProt.SIP.assuredService.namespace.label.UCRdrsn.1="" voIpProt.SIP.assuredService.namespace.label.UCRdrsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.UCRdrsn>
            <voIpProt.SIP.assuredService.namespace.label.UCRdsn voIpProt.SIP.assuredService.namespace.label.UCRdsn.1="" voIpProt.SIP.assuredService.namespace.label.UCRdsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.UCRdsn>
            <voIpProt.SIP.assuredService.namespace.label.custom voIpProt.SIP.assuredService.namespace.label.custom.1="" voIpProt.SIP.assuredService.namespace.label.custom.2="">
            </voIpProt.SIP.assuredService.namespace.label.custom>
            <voIpProt.SIP.assuredService.namespace.label.drsn voIpProt.SIP.assuredService.namespace.label.drsn.1="" voIpProt.SIP.assuredService.namespace.label.drsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.drsn>
            <voIpProt.SIP.assuredService.namespace.label.dsn voIpProt.SIP.assuredService.namespace.label.dsn.1="" voIpProt.SIP.assuredService.namespace.label.dsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.dsn>
          </voIpProt.SIP.assuredService.namespace.label>
        </voIpProt.SIP.assuredService.namespace>
      </voIpProt.SIP.assuredService>
    </voIpProt.SIP>
  </voIpProt>
</polycomConfig>

Following is the phone3 polycom config. x2996 and x2995 are on this phone.

<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- Generated reg-advanced.cfg Configuration File -->
<polycomConfig xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="polycomConfig.xsd">
  <call call.transferOnConferenceEnd="1">
    <call.autoAnswer call.autoAnswer.H323="0" call.autoAnswer.SIP="0" call.autoAnswer.micMute="1" call.autoAnswer.ringClass="ringAutoAnswer" call.autoAnswer.videoMute="0">
    </call.autoAnswer>
    <call.autoOffHook call.autoOffHook.1.contact="" call.autoOffHook.1.enabled="0" call.autoOffHook.1.protocol="" call.autoOffHook.2.contact="" call.autoOffHook.2.enabled="0" call.autoOffHook.2.protocol="">
    </call.autoOffHook>
    <call.donotdisturb call.donotdisturb.perReg="0">
    </call.donotdisturb>
    <call.missedCallTracking call.missedCallTracking.1.enabled="1" call.missedCallTracking.2.enabled="1">
    </call.missedCallTracking>
    <call.serverMissedCall call.serverMissedCall.1.enabled="0" call.serverMissedCall.2.enabled="0">
    </call.serverMissedCall>
  </call>
  <device>
    <device.auth device.auth.localUserPassword="">
      <device.auth.localUserPassword device.auth.localUserPassword.set="0">
      </device.auth.localUserPassword>
    </device.auth>
    <device.em device.em.power="">
      <device.em.power device.em.power.set="0">
      </device.em.power>
    </device.em>
  </device>
<reg reg.1.displayName="test user2" 
        reg.1.address="2996" 
        reg.1.label="2996" 
        reg.1.type="private" 
        reg.1.auth.userId="2996" 
        reg.1.auth.password="zebra" 
        reg.1.ringType="2" 
        reg.1.bargeInEnabled="1"
        reg.1.callsPerLineKey="4" 
        reg.1.csta="0" 
        reg.1.fwd.busy.contact="" 
        reg.1.fwd.busy.status="0" 
        reg.1.fwd.noanswer.contact="" 
        reg.1.fwd.noanswer.ringCount="0" 
        reg.1.fwd.noanswer.status="0" 
        reg.1.lineKeys="1" 
        reg.1.musicOnHold.uri="" 
        reg.1.outboundProxy.failOver.failBack.timeout="3600" 
        reg.1.outboundProxy.failOver.failRegistrationOn="1" 
        reg.1.outboundProxy.failOver.reRegisterOn="0" 
        reg.1.outboundProxy.address="192.168.100.33"
        reg.1.outboundProxy.port="5060" 
        reg.1.protocol.SIP="1" 
        reg.1.ringType="ringer2" 
        reg.1.serverFeatureControl.cf="1" 
        reg.1.serverFeatureControl.dnd="1" 
        reg.1.thirdPartyName="" 
        reg.1.type="private" 
        reg.1.useCompleteUriForRetrieve="1" 
        reg.1.server.1.expires="3600" 
        reg.1.server.1.expires.lineSeize="30" 
        reg.1.server.1.expires.overlap="60" 
        reg.1.server.1.lcs="0" 
        reg.1.server.1.retryMaxCount="3" 
        reg.1.server.1.retryTimeOut="0" 
        reg.1.server.2.expires="3600" 
        reg.1.server.2.expires.lineSeize="30" 
        reg.1.server.2.expires.overlap="60" 
        reg.1.server.2.lcs="0" 
        reg.1.server.2.retryMaxCount="3" 
        reg.1.server.2.retryTimeOut="0"
        reg.2.displayName="test user1" 
        reg.2.address="2995" 
        reg.2.label="2995" 
        reg.2.type="shared" 
        reg.2.auth.userId="2995" 
        reg.2.auth.password="zebra" 
        reg.2.ringType="2" 
        reg.2.bargeInEnabled="1"
        reg.2.callsPerLineKey="4" 
        reg.2.csta="0" 
        reg.2.fwd.busy.contact="" 
        reg.2.fwd.busy.status="0" 
        reg.2.fwd.noanswer.contact="" 
        reg.2.fwd.noanswer.ringCount="0" 
        reg.2.fwd.noanswer.status="0" 
        reg.2.lineKeys="1" 
        reg.2.musicOnHold.uri="" 
        reg.2.outboundProxy.failOver.failBack.timeout="3600" 
        reg.2.outboundProxy.failOver.failRegistrationOn="1" 
        reg.2.outboundProxy.failOver.reRegisterOn="0" 
        reg.2.outboundProxy.address="192.168.100.33"
        reg.2.outboundProxy.port="5060" 
        reg.2.protocol.SIP="1" 
        reg.2.ringType="ringer2" 
        reg.2.serverFeatureControl.cf="1" 
        reg.2.serverFeatureControl.dnd="1" 
        reg.2.thirdPartyName="" 
        reg.2.type="private" 
        reg.2.useCompleteUriForRetrieve="1" 
        reg.2.server.1.expires="3600" 
        reg.2.server.1.expires.lineSeize="30" 
        reg.2.server.1.expires.overlap="60" 
        reg.2.server.1.lcs="0" 
        reg.2.server.1.retryMaxCount="3" 
        reg.2.server.1.retryTimeOut="0" 
        reg.2.server.2.expires="3600" 
        reg.2.server.2.expires.lineSeize="30" 
        reg.2.server.2.expires.overlap="60" 
        reg.2.server.2.lcs="0" 
        reg.2.server.2.retryMaxCount="3" 
        reg.2.server.2.retryTimeOut="0">
  </reg>
 <se se.destination="chassis">
    <se.rt>
      <se.rt.modification se.rt.modification.enabled="1">
      </se.rt.modification>
    </se.rt>
  </se>
  <up up.25mm="0" up.accessibilityFeatures="0" up.analogHeadsetOption="0" up.audioMode="0" up.callTypeOrderVoice="0" up.handsfreeMode="1" up.headsetMode="0" up.localClockEnabled="1" up.manualProtocolRouting="1" up.mwiVisible="0" up.onHookDialingEnabled="1" up.useDirectoryNames="0" up.welcomeSoundEnabled="1" up.welcomeSoundOnWarmBootEnabled="0">
    <up.2P5mmJack up.2P5mmJack.mode="0">
    </up.2P5mmJack>
    <up.audioSetup up.audioSetup.auxInput="2" up.audioSetup.auxOutput="2">
    </up.audioSetup>
    <up.backlight up.backlight.idleIntensity="1" up.backlight.onIntensity="3">
    </up.backlight>
    <up.idleBrowser up.idleBrowser.enabled="0">
    </up.idleBrowser>
    <up.manualProtocolRouting up.manualProtocolRouting.softKeys="1">
    </up.manualProtocolRouting>
    <up.offHookAction up.offHookAction.none="0">
    </up.offHookAction>
    <up.pictureFrame up.pictureFrame.folder="" up.pictureFrame.timePerImage="5">
    </up.pictureFrame>
    <up.screenSaver up.screenSaver.enabled="0" up.screenSaver.waitTime="15">
    </up.screenSaver>
    <up.toneControl up.toneControl.bass="0" up.toneControl.treble="0">
    </up.toneControl>
  </up>
  <voIpProt>
    <voIpProt.SIP>
      <voIpProt.SIP.assuredService>
        <voIpProt.SIP.assuredService.namespace>
          <voIpProt.SIP.assuredService.namespace.label>
            <voIpProt.SIP.assuredService.namespace.label.UCRdrsn voIpProt.SIP.assuredService.namespace.label.UCRdrsn.1="" voIpProt.SIP.assuredService.namespace.label.UCRdrsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.UCRdrsn>
            <voIpProt.SIP.assuredService.namespace.label.UCRdsn voIpProt.SIP.assuredService.namespace.label.UCRdsn.1="" voIpProt.SIP.assuredService.namespace.label.UCRdsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.UCRdsn>
            <voIpProt.SIP.assuredService.namespace.label.custom voIpProt.SIP.assuredService.namespace.label.custom.1="" voIpProt.SIP.assuredService.namespace.label.custom.2="">
            </voIpProt.SIP.assuredService.namespace.label.custom>
            <voIpProt.SIP.assuredService.namespace.label.drsn voIpProt.SIP.assuredService.namespace.label.drsn.1="" voIpProt.SIP.assuredService.namespace.label.drsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.drsn>
            <voIpProt.SIP.assuredService.namespace.label.dsn voIpProt.SIP.assuredService.namespace.label.dsn.1="" voIpProt.SIP.assuredService.namespace.label.dsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.dsn>
          </voIpProt.SIP.assuredService.namespace.label>
        </voIpProt.SIP.assuredService.namespace>
      </voIpProt.SIP.assuredService>
    </voIpProt.SIP>
  </voIpProt>
</polycomConfig>



Thanks.-


----- Original Message -----
From: "Brian West" <brian at freeswitch.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, April 12, 2011 8:53:30 PM
Subject: Re: [Freeswitch-users] Shared Call appearence, barging and presence

I would need to see your polycom config options I have tested with 3.3.1 and it works fine... so please provide me some info on how you setup the phones... also please don't hijack threads.  Please click new message and input the list address.  You have clicked reply on the CALL_REJECT thread and changed the subject.

Thanks,
Brian

On Apr 12, 2011, at 6:55 PM, Gourav Vohra wrote:

> It would be very helpful to know if this is how freeswitch works with SLA or if my configuration is broken.
> 
> 
> Thanks.-
> 
> ----- Original Message -----
> From: "Gourav Vohra" <gourav at rentec.com>
> To: freeswitch-users at lists.freeswitch.org
> Sent: Tuesday, April 12, 2011 12:48:55 PM
> Subject: Shared Call appearence, barging and presence
> 
> 
> 
> 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, skipping
> 2011-04-08 10:09:33.381769 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/log/cdr-csv/2995.csv
> 2011-04-08 10:09:33.381769 [NOTICE] mod_cdr_csv.c:123 Rotated CDR logfile /usr/local/freeswitch/log/cdr-csv/Master.csv
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Wed Apr 13 07:00:06 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Tue, 12 Apr 2011 20:00:06 -0700 (PDT)
Subject: [Freeswitch-users] Incompatible destination
Message-ID: <34533.65062.qm@web34305.mail.mud.yahoo.com>

Hi All.

Today i updated my FS to version:

FreeSWITCH Version 1.0.head (git-54e5011 2011-04-12 13-35-39 -0500)

Before the do updated the calls are worked fine.

But now i can't made calls:

http://pastebin.ca/2045671

Anyone could help me?

Regards.


From msc at freeswitch.org  Wed Apr 13 08:05:15 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Apr 2011 21:05:15 -0700
Subject: [Freeswitch-users] FreeSWITCH: No Longer The Best Kept Secret In
	OSS VoIP Software
Message-ID: <BANLkTinDJxFHwxOdXGyK5whoBPhUxOEreg@mail.gmail.com>

As you know, the FreeSWITCH project and community are both growing rapidly.
We are always looking for people to step up and help out. This article talks
about some of the things that you can do to assist in dealing with the
growing pains that we all feel:

http://www.freeswitch.org/node/320

Thanks for your support and please keep up the good work!

-Michael
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From msc at freeswitch.org  Wed Apr 13 08:09:43 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Apr 2011 21:09:43 -0700
Subject: [Freeswitch-users] occasional ~5s delay during bind_meta_app
	execute_extenstion
In-Reply-To: <BANLkTinnpJrLAg+XWs1izQ0MKE_ym+PVcQ@mail.gmail.com>
References: <BANLkTikxT9VuejCLOUsbKCncLXv6HiEWiw@mail.gmail.com>
	<BANLkTin+HLFD6dnc=zWs_P-gftsgmpP+vw@mail.gmail.com>
	<BANLkTinnpJrLAg+XWs1izQ0MKE_ym+PVcQ@mail.gmail.com>
Message-ID: <BANLkTind1exnpMFgT-iS7sdbJqSi=QksOQ@mail.gmail.com>

On Tue, Apr 12, 2011 at 5:57 PM, elijah <elijah at crankenstein.com> wrote:

> I'm afraid that cannot be the cause. I'm experiencing the delay before the
> read is executed, immediately after a user presses *1 but before the
> extension named 'dx' executes anything.


Put these extensions into pb with a call log of this issue occurring. Use
console loglevel 7.
-MC
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From msc at freeswitch.org  Wed Apr 13 08:13:35 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 12 Apr 2011 21:13:35 -0700
Subject: [Freeswitch-users] Incompatible destination
In-Reply-To: <34533.65062.qm@web34305.mail.mud.yahoo.com>
References: <34533.65062.qm@web34305.mail.mud.yahoo.com>
Message-ID: <BANLkTikn9bhN0tejhWkr5nnJAnCA8fop=A@mail.gmail.com>

I recommend you use our pastebin and get a more complete log. If you go into
src/libs/esl/perl there is a logger.pl script that can assist with
collecting trace information and putting into pastebin. You need to make
sure that you are getting a sip trace as well as debug level output.

-MC

On Tue, Apr 12, 2011 at 8:00 PM, FERNANDO VILLARROEL
<fvillarroel at yahoo.com>wrote:

> Hi All.
>
> Today i updated my FS to version:
>
> FreeSWITCH Version 1.0.head (git-54e5011 2011-04-12 13-35-39 -0500)
>
> Before the do updated the calls are worked fine.
>
> But now i can't made calls:
>
> http://pastebin.ca/2045671
>
> 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
>
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From ovvenkatesan at gmail.com  Wed Apr 13 09:03:23 2011
From: ovvenkatesan at gmail.com (ovvenkat)
Date: Wed, 13 Apr 2011 10:33:23 +0530
Subject: [Freeswitch-users] Freeswitch Server Down
In-Reply-To: <BANLkTi=C=cjknY7ckpxkxVrK-Ci0ZgAqrQ@mail.gmail.com>
References: <BANLkTimerjRCutH8LFaRjvCfSju6byO1ow@mail.gmail.com>
	<BANLkTikLGEqSnL7yYnvVmBE6AvYgCBTXmg@mail.gmail.com>
	<637F58A1-8368-4E26-A192-3629D9BDAF8F@gmail.com>
	<BANLkTikmZfQjM3wEiVYY+2xuQruCmmaUjQ@mail.gmail.com>
	<BANLkTi=C=cjknY7ckpxkxVrK-Ci0ZgAqrQ@mail.gmail.com>
Message-ID: <BANLkTikPb6uR6ytUNtiGxr7NMypqA_JOYw@mail.gmail.com>

Hi Steven,

Could you please suggest me,
which one I need to upgrade?.

freeSwitch or wanpipe?

regards,
Venkat.


On Tue, Apr 12, 2011 at 8:41 PM, Steven Ayre <steveayre at gmail.com> wrote:

> 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.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 grsingh750 at gmail.com  Wed Apr 13 09:59:45 2011
From: grsingh750 at gmail.com (guru singh)
Date: Wed, 13 Apr 2011 11:29:45 +0530
Subject: [Freeswitch-users] Freeswitch Server Down
In-Reply-To: <BANLkTikPb6uR6ytUNtiGxr7NMypqA_JOYw@mail.gmail.com>
References: <BANLkTimerjRCutH8LFaRjvCfSju6byO1ow@mail.gmail.com>
	<BANLkTikLGEqSnL7yYnvVmBE6AvYgCBTXmg@mail.gmail.com>
	<637F58A1-8368-4E26-A192-3629D9BDAF8F@gmail.com>
	<BANLkTikmZfQjM3wEiVYY+2xuQruCmmaUjQ@mail.gmail.com>
	<BANLkTi=C=cjknY7ckpxkxVrK-Ci0ZgAqrQ@mail.gmail.com>
	<BANLkTikPb6uR6ytUNtiGxr7NMypqA_JOYw@mail.gmail.com>
Message-ID: <BANLkTimW=bMQfLVEcmC1jpMX2HCn4_7HUg@mail.gmail.com>

Hi,

If you are going to upgrade, it won't hurt to do it for both IMHO.

regards,
guru

On Wed, Apr 13, 2011 at 10:33 AM, ovvenkat <ovvenkatesan at gmail.com> wrote:
> Hi Steven,
>
> Could you please suggest me,
> which one I need to upgrade?.
>
> freeSwitch or wanpipe?
>
> regards,
> Venkat.
>
>
> On Tue, Apr 12, 2011 at 8:41 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> 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>
>>>> 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.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From u2nsam at gmail.com  Wed Apr 13 11:53:45 2011
From: u2nsam at gmail.com (Sam)
Date: Wed, 13 Apr 2011 13:23:45 +0530
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <BANLkTimFeP+r2uNk09BbVw7hAULf1MYvzA@mail.gmail.com>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A28C423231C@exmachina.office.kapper.net>
	<BANLkTimrvA=Lgva-FAdXxWRhS-_UDuTgTA@mail.gmail.com>
	<BANLkTimFeP+r2uNk09BbVw7hAULf1MYvzA@mail.gmail.com>
Message-ID: <BANLkTikyVkH_FyLsdhnkUBPOB-bZwjGcXQ@mail.gmail.com>

I want to pass the exact information what leg A is sending for Media
Description & Media Attributes and not connection information.


Regards
Sam



On Tue, Apr 12, 2011 at 10:16 PM, Steven Ayre <steveayre at gmail.com> wrote:

> 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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Apr 13 12:51:23 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Wed, 13 Apr 2011 10:51:23 +0200
Subject: [Freeswitch-users] Passing SIP headers from b-leg to a-leg
Message-ID: <BANLkTi=nxevwAXsVOtjwsAw8B4TeAr-yYQ@mail.gmail.com>

Hi Anthony,

I'm using verson FreeSWITCH Version 1.0.head (git-54e5011 2011-04-12
13-35-39 -0500), but I haven't been able to make it work. This is my bridge
application line:
<action application="bridge"
data="{sip_copy_custom_headers=true}sofia/internal/${
destination_number}@79.170.68.217"

This the the failed response I'm getting from the provider:
 SIP/2.0 503 Service Unavailable..Via: SIP/2.0/UDP
79.170.68.154;rport;branch=z9hG4bKN48DavB78KjBm..From: "34917019888" <
sip:34917019888 at 79.170.68.154>;tag=atFFtate53HcH..To:
  <sip:34661574758 at 79.170.68.217>;tag=1..Call-ID:
64911780-e04d-122e-ca81-001b7873d3a2..CSeq: 11006136 INVITE..Reason: Q.850
;cause=34 ;text="Unknown"..*X-Release-Source: Provider*..Content-Length:
0....

And this is the message sent by FS to the A_leg:

SIP/2.0 503 Service Unavailable..Via: SIP/2.0/UDP
79.170.68.144;branch=z9hG4bK805.e2f3bfa1.0..Via: SIP/2.0/UDP
79.170.64.151:5060;branch=z9hG4bK14faa4097ll7d213fINV26d034814d
  a55dbf..From:
<sip:34917019888 at 79.170.64.151;noa=international>;tag=26d03481-co7455-INS001..To:
<sip:34661574758 at 79.170.68.144
;noa=international>;tag=5cHKH35vKQ2eQ..Call-ID:
  4da55dbf000000050000001372519733 at ens.com..CSeq: 745501 INVITE..User-Agent:
FreeSWITCH-mod_sofia/1.0.head-git-54e5011 2011-04-12 13-35-39 -0500..Accept:
application/sdp..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..Reason:
Q.850;cause=41;text="NORMAL_TEMPORARY_FAILURE"..Content-Length:
0..Remote-Party-ID: "34661574758" <sip:34661574758 at 79.170.68.154
>;party=calling;privacy=off;screen=no....

Any idea of what am I doing wrong?

Regards

Javi

---------- Forwarded message ----------
> From: Anthony Minessale <anthony.minessale at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 12 Apr 2011 10:36:04 -0500
> Subject:
> 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
>
> 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.or<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:+19193869900
>
>
>
>
>
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From gcd at i.ph  Wed Apr 13 12:52:42 2011
From: gcd at i.ph (Nandy Dagondon)
Date: Wed, 13 Apr 2011 16:52:42 +0800
Subject: [Freeswitch-users] called number rewrite
In-Reply-To: <171211778-1302657000-cardhu_decombobulator_blackberry.rim.net-377026527-@b4.c2.bise3.blackberry>
References: <BANLkTikEMcBF-vK4QhTrb6ABw7fkDe=3Zw@mail.gmail.com>
	<C9CA5E0E.135DE%krice@freeswitch.org>
	<171211778-1302657000-cardhu_decombobulator_blackberry.rim.net-377026527-@b4.c2.bise3.blackberry>
Message-ID: <BANLkTi=efRkgtsHrmhjeefCUCMxnL3GTxg@mail.gmail.com>

if this task is similar in mapping destination numbers to routes as in Class
5 exchanges, mod_lcr could be applicable here.
-nandy


On Wed, Apr 13, 2011 at 9:09 AM, Budi wibowo <bwibowo at gmail.com> wrote:

> Thx anybody know how many mapping/rewrite can be performed by fs?
> (Based on experience)
>
> -----Original Message-----
> From: Ken Rice <krice at freeswitch.org>
> Sender: freeswitch-users-bounces at lists.freeswitch.org
> Date: Tue, 12 Apr 2011 19:52:46
> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] called number rewrite
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 me at nevian.org  Wed Apr 13 16:22:23 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Wed, 13 Apr 2011 16:22:23 +0400
Subject: [Freeswitch-users] Incompatible destination
In-Reply-To: <34533.65062.qm@web34305.mail.mud.yahoo.com>
References: <34533.65062.qm@web34305.mail.mud.yahoo.com>
Message-ID: <570351302697344@web16.yandex.ru>

hi

13.04.2011, 07:03, "FERNANDO VILLARROEL" <fvillarroel at yahoo.com>;:

> ?Hi All.
>
> ?Today i updated my FS to version:
>
> ?FreeSWITCH Version 1.0.head (git-54e5011 2011-04-12 13-35-39 -0500)
>
> ?Before the do updated the calls are worked fine.
>
> ?But now i can't made calls:
> ?http://pastebin.ca/2045671
>
> ?Anyone could help me?

You have disabled g729 in incoming profile and your endpoint knows only one codec at all.
Perhaps you not only updated binaries but also applied new configs in which 729 not present by default.
btw, I hope you know that 729 work only in passtrough (except commercial module) mode so nor IVR nor transcoding not functional


-- 
wbr,
Serge


From gchen00 at insightbb.com  Wed Apr 13 16:33:15 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Wed, 13 Apr 2011 08:33:15 -0400
Subject: [Freeswitch-users] FS does not repose to SIP OK message.
Message-ID: <fc7f86bf12222.4da55fcb@insightbb.com>



The FS1 should response back with ACK. It did not. 
The strange thing is that It works sometime.Here is the SIP trace from ngrep for both good and bad calls on FS1 server:SJPhone IP: 226.59.139.61FS1 IP:226.59.129.223FS2 IP:226.59.129.221You can see that FS2 sent several 200 OK and FS1 never reply back.
Good sip call:U 226.59.139.61:5060 -> 226.59.129.223:5080
? INVITE sip:5025155596 at fs2000.lightyear.net SIP/2.0..Via: SIP/2.0/UDP 226.59.139.61;branch=z9hG4bKd8318b3d000007494da459a
? 900006fac00000273;rport..From: "unknown" <sip:1009 at fs2000.lightyear.net>;tag=a344c583a24..To: <sip:5025155596 at fs2000.lig
? htyear.net>..Contact: <sip:1009 at 226.59.139.61>..Call-ID: 56BF61B3189F4EDEB8C9067D5823C8480xd8318b3d..CSeq: 1 INVITE..Max
? -Forwards: 70..User-Agent: SJphone/1.65.377a (SJ Labs)..Content-Length: 368..Content-Type: application/sdp..Supported: r
? eplaces,norefersub,timer....v=0..o=- 3511605289 3511605289 IN IP4 226.59.139.61..s=SJphone..c=IN IP4 226.59.139.61..t=0
? 0..m=audio 49352 RTP/AVP 3 97 98 8 0 101..c=IN IP4 226.59.139.61..a=rtpmap:3 GSM/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:9
? 8 iLBC/8000..a=fmtp:98 mode=20..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:10
? 1 0-16..a=setup:active..a=sendrecv..
#
U 226.59.129.223:5080 -> 226.59.139.61:5060
? SIP/2.0 100 Trying..Via: SIP/2.0/UDP 226.59.139.61;branch=z9hG4bKd8318b3d000007494da459a900006fac00000273;rport=5060..Fr
? om: "unknown" <sip:1009 at fs2000.lightyear.net>;tag=a344c583a24..To: <sip:5025155596 at fs2000.lightyear.net>..Call-ID: 56BF6
? 1B3189F4EDEB8C9067D5823C8480xd8318b3d..CSeq: 1 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-5310735 2011-04-07
? 15-47-30 -0500..Content-Length: 0....
#
U 226.59.129.223:5060 -> 226.59.129.221:5060
? INVITE sip:5596 at 226.59.129.221:5060 SIP/2.0..Via: SIP/2.0/UDP 226.59.129.223;rport;branch=z9hG4bK4g569HgUr3QXN..Max-Forw
? ards: 68..From: "unknown" <sip:1009 at 226.59.129.223>;tag=KQKpXe0m39X6e..To: <sip:5596 at 226.59.129.221:5060>..Call-ID: 4639
? 3d91-dfaf-122e-d1b0-e9d83e422dc6..CSeq: 10972180 INVITE..Contact: <sip:mod_sofia at 226.59.129.223:5060>..User-Agent: FreeS
? WITCH-mod_sofia/1.0.head-git-5310735 2011-04-07 15-47-30 -0500..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDAT
? E, 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, refe
? r..Content-Type: application/sdp..Content-Disposition: session..Content-Length: 317..X-FS-Support: update_display..Remot
? e-Party-ID: "unknown" <sip:1009 at 226.59.129.223>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1302587641 130
? 2587642 IN IP4 226.59.129.223..s=FreeSWITCH..c=IN IP4 226.59.129.223..t=0 0..m=audio 28848 RTP/AVP 3 98 99 9 0 8 101 13.
? .a=rtpmap:98 G7221/32000..a=fmtp:98 bitrate=48000..a=rtpmap:99 G7221/16000..a=fmtp:99 bitrate=32000..a=rtpmap:101 teleph
? one-event/8000..a=fmtp:101 0-16..a=ptime:20..
#
U 226.59.129.221:5060 -> 226.59.129.223:5060
? SIP/2.0 100 Trying..Via: SIP/2.0/UDP 226.59.129.223;rport=5060;branch=z9hG4bK4g569HgUr3QXN..From: "unknown" <sip:1009 at 21
? 6.49.129.223>;tag=KQKpXe0m39X6e..To: <sip:5596 at 226.59.129.221:5060>..Call-ID: 46393d91-dfaf-122e-d1b0-e9d83e422dc6..CSeq
? : 10972180 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110119T213949Z..Content-Length: 0....
#
U 226.59.129.221:5060 -> 226.59.129.223:5060
? SIP/2.0 200 OK..Via: SIP/2.0/UDP 226.59.129.223;rport=5060;branch=z9hG4bK4g569HgUr3QXN..From: "unknown" <sip:1009 at 226.59
? .129.223>;tag=KQKpXe0m39X6e..To: <sip:5596 at 226.59.129.221:5060>;tag=7r593SDymv51S..Call-ID: 46393d91-dfaf-122e-d1b0-e9d8
? 3e422dc6..CSeq: 10972180 INVITE..Contact: <sip:5596 at 226.59.129.221:5060;transport=udp>..User-Agent: FreeSWITCH-mod_sofia
? /1.0.7-hacked-20110119T213949Z..Accept: application/sdp..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..Cont
? ent-Type: application/sdp..Content-Disposition: session..Content-Length: 247..X-FS-Display-Name: 5596..X-FS-Display-Numb
? er: sip:5596 at 226.59.129.221..X-FS-Support: update_display..Remote-Party-ID: "5596" <sip:5596 at 226.59.129.221>;party=calli
? ng;privacy=off;screen=no....v=0..o=FreeSWITCH 1302597804 1302597805 IN IP4 226.59.129.221..s=FreeSWITCH..c=IN IP4 226.59
? .129.221..t=0 0..m=audio 18640 RTP/AVP 3 101 13..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16
? ..a=rtpmap:13 CN/8000..a=ptime:20..
#
U 226.59.129.223:5060 -> 226.59.129.221:5060
? ACK sip:5596 at 226.59.129.221:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 226.59.129.223;rport;branch=z9hG4bK5SyZBD1yNceg
? H..Max-Forwards: 70..From: "unknown" <sip:1009 at 226.59.129.223>;tag=KQKpXe0m39X6e..To: <sip:5596 at 226.59.129.221:5060>;tag
? =7r593SDymv51S..Call-ID: 46393d91-dfaf-122e-d1b0-e9d83e422dc6..CSeq: 10972180 ACK..Contact: <sip:mod_sofia at 226.59.129.22
? 3:5060>..Content-Length: 0....
#
U 226.59.129.223:5080 -> 226.59.139.61:5060
? SIP/2.0 200 OK..Via: SIP/2.0/UDP 226.59.139.61;branch=z9hG4bKd8318b3d000007494da459a900006fac00000273;rport=5060..From:
? "unknown" <sip:1009 at fs2000.lightyear.net>;tag=a344c583a24..To: <sip:5025155596 at fs2000.lightyear.net>;tag=KB4NB13DD35eg..
? Call-ID: 56BF61B3189F4EDEB8C9067D5823C8480xd8318b3d..CSeq: 1 INVITE..Contact: <sip:5025155596 at 226.59.129.223:5080;transp
? ort=udp>..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-5310735 2011-04-07 15-47-30 -0500..Accept: application/sdp..Allo
? w: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY..Supported: timer, precondition, pa
? th, replaces..Allow-Events: talk, hold, refer..Content-Type: application/sdp..Content-Disposition: session..Content-Leng
? th: 250..X-FS-Display-Name: 5596..X-FS-Display-Number: sip:5596 at 226.59.129.221..X-FS-Support: update_display..Remote-Par
? ty-ID: "5596" <sip:5596 at 226.59.129.223>;party=calling;privacy=off;screen=no....v=0..o=FreeSWITCH 1302591209 1302591210 I
? N IP4 226.59.129.223..s=FreeSWITCH..c=IN IP4 226.59.129.223..t=0 0..m=audio 25280 RTP/AVP 3 101..a=rtpmap:3 GSM/8000..a=
? rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..a=ptime:20..
#
U 226.59.139.61:5060 -> 226.59.129.223:5080
? ACK sip:5025155596 at 226.59.129.223:5080;transport=udp SIP/2.0..Via: SIP/2.0/UDP 226.59.139.61;branch=z9hG4bKd8318b3d00000
? 7494da459a900003d7800000277;rport..From: "unknown" <sip:1009 at fs2000.lightyear.net>;tag=a344c583a24..To: <sip:5025155596@
? fs2000.lightyear.net>;tag=KB4NB13DD35eg..Contact: <sip:1009 at 226.59.139.61>..Call-ID: 56BF61B3189F4EDEB8C9067D5823C8480xd
? 8318b3d..CSeq: 1 ACK..Max-Forwards: 70..User-Agent: SJphone/1.65.377a (SJ Labs)..Content-Length: 0....
#
U 226.59.129.223:5060 -> 226.59.129.221:5060
? INFO sip:5596 at 226.59.129.221:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 226.59.129.223;rport;branch=z9hG4bK62QrD8H2jN4
? 2c..Max-Forwards: 70..From: "unknown" <sip:1009 at 226.59.129.223>;tag=KQKpXe0m39X6e..To: <sip:5596 at 226.59.129.221:5060>;ta
? g=7r593SDymv51S..Call-ID: 46393d91-dfaf-122e-d1b0-e9d83e422dc6..CSeq: 10972181 INFO..Contact: <sip:mod_sofia at 226.59.129.
? 223:5060>..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-5310735 2011-04-07 15-47-30 -0500..Allow: INVITE, ACK, BYE, CAN
? CEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path,
? replaces..Content-Type: message/update_display..Content-Length: 0..X-FS-Display-Name: unknown..X-FS-Display-Number: 1009
? ....
#
U 226.59.129.221:5060 -> 226.59.129.223:5060
? SIP/2.0 200 OK..Via: SIP/2.0/UDP 226.59.129.223;rport=5060;branch=z9hG4bK62QrD8H2jN42c..From: "unknown" <sip:1009 at 226.59
? .129.223>;tag=KQKpXe0m39X6e..To: <sip:5596 at 226.59.129.221:5060>;tag=7r593SDymv51S..Call-ID: 46393d91-dfaf-122e-d1b0-e9d8
? 3e422dc6..CSeq: 10972181 INFO..User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110119T213949Z..Allow: INVITE, ACK, BYE,
? CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, pat
? h, replaces..Content-Length: 0....
#
U 226.59.139.61:5060 -> 226.59.129.223:5080
? BYE sip:5025155596 at 226.59.129.223:5080;transport=udp SIP/2.0..Via: SIP/2.0/UDP 226.59.139.61;branch=z9hG4bKd8318b3d00000
? 74b4da459e200004b8500000278;rport..From: "unknown" <sip:1009 at fs2000.lightyear.net>;tag=a344c583a24..To: <sip:5025155596@
? fs2000.lightyear.net>;tag=KB4NB13DD35eg..Contact: <sip:1009 at 226.59.139.61>..Call-ID: 56BF61B3189F4EDEB8C9067D5823C8480xd
? 8318b3d..CSeq: 2 BYE..Max-Forwards: 70..User-Agent: SJphone/1.65.377a (SJ Labs)..Content-Length: 0..Supported: replaces,
? norefersub,timer....
#
U 226.59.129.223:5080 -> 226.59.139.61:5060
? SIP/2.0 200 OK..Via: SIP/2.0/UDP 226.59.139.61;branch=z9hG4bKd8318b3d0000074b4da459e200004b8500000278;rport=5060..From:
? "unknown" <sip:1009 at fs2000.lightyear.net>;tag=a344c583a24..To: <sip:5025155596 at fs2000.lightyear.net>;tag=KB4NB13DD35eg..
? Call-ID: 56BF61B3189F4EDEB8C9067D5823C8480xd8318b3d..CSeq: 2 BYE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-5310735
? 2011-04-07 15-47-30 -0500..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY..Sup
? ported: timer, precondition, path, replaces..Content-Length: 0....
#
U 226.59.129.223:5060 -> 226.59.129.221:5060
? BYE sip:5596 at 226.59.129.221:5060;transport=udp SIP/2.0..Via: SIP/2.0/UDP 226.59.129.223;rport;branch=z9hG4bK7BHHF325FytN
? r..Max-Forwards: 70..From: "unknown" <sip:1009 at 226.59.129.223>;tag=KQKpXe0m39X6e..To: <sip:5596 at 226.59.129.221:5060>;tag
? =7r593SDymv51S..Call-ID: 46393d91-dfaf-122e-d1b0-e9d83e422dc6..CSeq: 10972182 BYE..Contact: <sip:mod_sofia at 226.59.129.22
? 3:5060>..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-5310735 2011-04-07 15-47-30 -0500..Allow: INVITE, ACK, BYE, CANCE
? L, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, re
? places..Reason: Q.850;cause=16;text="NORMAL_CLEARING"..Content-Length: 0....
#
U 226.59.129.221:5060 -> 226.59.129.223:5060
? SIP/2.0 200 OK..Via: SIP/2.0/UDP 226.59.129.223;rport=5060;branch=z9hG4bK7BHHF325FytNr..From: "unknown" <sip:1009 at 226.59
? .129.223>;tag=KQKpXe0m39X6e..To: <sip:5596 at 226.59.129.221:5060>;tag=7r593SDymv51S..Call-ID: 46393d91-dfaf-122e-d1b0-e9d8
? 3e422dc6..CSeq: 10972182 BYE..User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110119T213949Z..Allow: INVITE, ACK, BYE, C
? ANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path
? , replaces..Content-Length: 0....

Bad SIP call:U 226.59.139.61:5060 -> 226.59.129.223:5080
? INVITE sip:5025155596 at fs2000.lightyear.net SIP/2.0..Via: SIP/2.0/UDP 226.59.139.61;branch=z9hG4bKd8318b3d000007324da4586
? e0000031200000260;rport..From: "unknown" <sip:1009 at fs2000.lightyear.net>;tag=60654c536a55..To: <sip:5025155596 at fs2000.li
? ghtyear.net>..Contact: <sip:1009 at 226.59.139.61>..Call-ID: 44C45839A15C4201A81F99F69C8FDB4D0xd8318b3d..CSeq: 1 INVITE..Ma
? x-Forwards: 70..User-Agent: SJphone/1.65.377a (SJ Labs)..Content-Length: 368..Content-Type: application/sdp..Supported:
? replaces,norefersub,timer....v=0..o=- 3511604973 3511604973 IN IP4 226.59.139.61..s=SJphone..c=IN IP4 226.59.139.61..t=0
?? 0..m=audio 49346 RTP/AVP 3 97 98 8 0 101..c=IN IP4 226.59.139.61..a=rtpmap:3 GSM/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:
? 98 iLBC/8000..a=fmtp:98 mode=20..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:1
? 01 0-16..a=setup:active..a=sendrecv..
#
U 226.59.129.223:5080 -> 226.59.139.61:5060
? SIP/2.0 100 Trying..Via: SIP/2.0/UDP 226.59.139.61;branch=z9hG4bKd8318b3d000007324da4586e0000031200000260;rport=5060..Fr
? om: "unknown" <sip:1009 at fs2000.lightyear.net>;tag=60654c536a55..To: <sip:5025155596 at fs2000.lightyear.net>..Call-ID: 44C4
? 5839A15C4201A81F99F69C8FDB4D0xd8318b3d..CSeq: 1 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-5310735 2011-04-07
?? 15-47-30 -0500..Content-Length: 0....
#
U 226.59.129.223:5060 -> 226.59.129.221:5060
? INVITE sip:5596 at 226.59.129.221:5060 SIP/2.0..Via: SIP/2.0/UDP 226.59.129.223;rport;branch=z9hG4bKtp8er37QNaS2H..Max-Forw
? ards: 68..From: "unknown" <sip:1009 at 226.59.129.223>;tag=gv7BrXDacFUec..To: <sip:5596 at 226.59.129.221:5060>..Call-ID: 8a42
? e048-dfae-122e-d1b0-e9d83e422dc6..CSeq: 10972023 INVITE..Contact: <sip:mod_sofia at 226.59.129.223:5060>..User-Agent: FreeS
? WITCH-mod_sofia/1.0.head-git-5310735 2011-04-07 15-47-30 -0500..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDAT
? E, 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, refe
? r..Content-Type: application/sdp..Content-Disposition: session..Content-Length: 317..X-FS-Support: update_display..Remot
? e-Party-ID: "unknown" <sip:1009 at 226.59.129.223>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1302588008 130
? 2588009 IN IP4 226.59.129.223..s=FreeSWITCH..c=IN IP4 226.59.129.223..t=0 0..m=audio 28166 RTP/AVP 3 98 99 9 0 8 101 13.
? .a=rtpmap:98 G7221/32000..a=fmtp:98 bitrate=48000..a=rtpmap:99 G7221/16000..a=fmtp:99 bitrate=32000..a=rtpmap:101 teleph
? one-event/8000..a=fmtp:101 0-16..a=ptime:20..
#
U 226.59.129.221:5060 -> 226.59.129.223:5060
? SIP/2.0 100 Trying..Via: SIP/2.0/UDP 226.59.129.223;rport=5060;branch=z9hG4bKtp8er37QNaS2H..From: "unknown" <sip:1009 at 21
? 6.49.129.223>;tag=gv7BrXDacFUec..To: <sip:5596 at 226.59.129.221:5060>..Call-ID: 8a42e048-dfae-122e-d1b0-e9d83e422dc6..CSeq
? : 10972023 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.7-hacked-20110119T213949Z..Content-Length: 0....
#
U 226.59.129.221:5060 -> 226.59.129.223:5060
? SIP/2.0 200 OK..Via: SIP/2.0/UDP 226.59.129.223;rport=5060;branch=z9hG4bKtp8er37QNaS2H..From: "unknown" <sip:1009 at 226.59
? .129.223>;tag=gv7BrXDacFUec..To: <sip:5596 at 226.59.129.221:5060>;tag=4XSZy8tKX129p..Call-ID: 8a42e048-dfae-122e-d1b0-e9d8
? 3e422dc6..CSeq: 10972023 INVITE..Contact: <sip:5596 at 226.59.129.221:5060;transport=udp>..User-Agent: FreeSWITCH-mod_sofia
? /1.0.7-hacked-20110119T213949Z..Accept: application/sdp..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..Cont
? ent-Type: application/sdp..Content-Disposition: session..Content-Length: 247..X-FS-Display-Name: 5596..X-FS-Display-Numb
? er: sip:5596 at 226.59.129.221..X-FS-Support: update_display..Remote-Party-ID: "5596" <sip:5596 at 226.59.129.221>;party=calli
? ng;privacy=off;screen=no....v=0..o=FreeSWITCH 1302589491 1302589492 IN IP4 226.59.129.221..s=FreeSWITCH..c=IN IP4 226.59
? .129.221..t=0 0..m=audio 26638 RTP/AVP 3 101 13..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16
? ..a=rtpmap:13 CN/8000..a=ptime:20..
#
U 226.59.129.221:5060 -> 226.59.129.223:5060
? SIP/2.0 200 OK..Via: SIP/2.0/UDP 226.59.129.223;rport=5060;branch=z9hG4bKtp8er37QNaS2H..From: "unknown" <sip:1009 at 226.59
? .129.223>;tag=gv7BrXDacFUec..To: <sip:5596 at 226.59.129.221:5060>;tag=4XSZy8tKX129p..Call-ID: 8a42e048-dfae-122e-d1b0-e9d8
? 3e422dc6..CSeq: 10972023 INVITE..Contact: <sip:5596 at 226.59.129.221:5060;transport=udp>..User-Agent: FreeSWITCH-mod_sofia
? /1.0.7-hacked-20110119T213949Z..Accept: application/sdp..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..Cont
? ent-Type: application/sdp..Content-Disposition: session..Content-Length: 247..X-FS-Display-Name: 5596..X-FS-Display-Numb
? er: sip:5596 at 226.59.129.221..X-FS-Support: update_display..Remote-Party-ID: "5596" <sip:5596 at 226.59.129.221>;party=calli
? ng;privacy=off;screen=no....v=0..o=FreeSWITCH 1302589491 1302589492 IN IP4 226.59.129.221..s=FreeSWITCH..c=IN IP4 226.59
? .129.221..t=0 0..m=audio 26638 RTP/AVP 3 101 13..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16
? ..a=rtpmap:13 CN/8000..a=ptime:20..
#
U 226.59.129.221:5060 -> 226.59.129.223:5060
? SIP/2.0 200 OK..Via: SIP/2.0/UDP 226.59.129.223;rport=5060;branch=z9hG4bKtp8er37QNaS2H..From: "unknown" <sip:1009 at 226.59
? .129.223>;tag=gv7BrXDacFUec..To: <sip:5596 at 226.59.129.221:5060>;tag=4XSZy8tKX129p..Call-ID: 8a42e048-dfae-122e-d1b0-e9d8
? 3e422dc6..CSeq: 10972023 INVITE..Contact: <sip:5596 at 226.59.129.221:5060;transport=udp>..User-Agent: FreeSWITCH-mod_sofia
? /1.0.7-hacked-20110119T213949Z..Accept: application/sdp..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..Cont
? ent-Type: application/sdp..Content-Disposition: session..Content-Length: 247..X-FS-Display-Name: 5596..X-FS-Display-Numb
? er: sip:5596 at 226.59.129.221..X-FS-Support: update_display..Remote-Party-ID: "5596" <sip:5596 at 226.59.129.221>;party=calli
? ng;privacy=off;screen=no....v=0..o=FreeSWITCH 1302589491 1302589492 IN IP4 226.59.129.221..s=FreeSWITCH..c=IN IP4 226.59
? .129.221..t=0 0..m=audio 26638 RTP/AVP 3 101 13..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16
? ..a=rtpmap:13 CN/8000..a=ptime:20..
#
U 226.59.129.221:5060 -> 226.59.129.223:5060
? SIP/2.0 200 OK..Via: SIP/2.0/UDP 226.59.129.223;rport=5060;branch=z9hG4bKtp8er37QNaS2H..From: "unknown" <sip:1009 at 226.59
? .129.223>;tag=gv7BrXDacFUec..To: <sip:5596 at 226.59.129.221:5060>;tag=4XSZy8tKX129p..Call-ID: 8a42e048-dfae-122e-d1b0-e9d8
? 3e422dc6..CSeq: 10972023 INVITE..Contact: <sip:5596 at 226.59.129.221:5060;transport=udp>..User-Agent: FreeSWITCH-mod_sofia
? /1.0.7-hacked-20110119T213949Z..Accept: application/sdp..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..Cont
? ent-Type: application/sdp..Content-Disposition: session..Content-Length: 247..X-FS-Display-Name: 5596..X-FS-Display-Numb
? er: sip:5596 at 226.59.129.221..X-FS-Support: update_display..Remote-Party-ID: "5596" <sip:5596 at 226.59.129.221>;party=calli
? ng;privacy=off;screen=no....v=0..o=FreeSWITCH 1302589491 1302589492 IN IP4 226.59.129.221..s=FreeSWITCH..c=IN IP4 226.59
? .129.221..t=0 0..m=audio 26638 RTP/AVP 3 101 13..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16
? ..a=rtpmap:13 CN/8000..a=ptime:20..
U 226.59.129.221:5060 -> 226.59.129.223:5060
? SIP/2.0 200 OK..Via: SIP/2.0/UDP 226.59.129.223;rport=5060;branch=z9hG4bKtp8er37QNaS2H..From: "unknown" <sip:1009 at 226.59
? .129.223>;tag=gv7BrXDacFUec..To: <sip:5596 at 226.59.129.221:5060>;tag=4XSZy8tKX129p..Call-ID: 8a42e048-dfae-122e-d1b0-e9d8
? 3e422dc6..CSeq: 10972023 INVITE..Contact: <sip:5596 at 226.59.129.221:5060;transport=udp>..User-Agent: FreeSWITCH-mod_sofia
? /1.0.7-hacked-20110119T213949Z..Accept: application/sdp..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..Cont
? ent-Type: application/sdp..Content-Disposition: session..Content-Length: 247..X-FS-Display-Name: 5596..X-FS-Display-Numb
? er: sip:5596 at 226.59.129.221..X-FS-Support: update_display..Remote-Party-ID: "5596" <sip:5596 at 226.59.129.221>;party=calli
? ng;privacy=off;screen=no....v=0..o=FreeSWITCH 1302589491 1302589492 IN IP4 226.59.129.221..s=FreeSWITCH..c=IN IP4 226.59
? .129.221..t=0 0..m=audio 26638 RTP/AVP 3 101 13..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16
? ..a=rtpmap:13 CN/8000..a=ptime:20..
#
U 226.59.129.221:5060 -> 226.59.129.223:5060
? SIP/2.0 200 OK..Via: SIP/2.0/UDP 226.59.129.223;rport=5060;branch=z9hG4bKtp8er37QNaS2H..From: "unknown" <sip:1009 at 226.59
? .129.223>;tag=gv7BrXDacFUec..To: <sip:5596 at 226.59.129.221:5060>;tag=4XSZy8tKX129p..Call-ID: 8a42e048-dfae-122e-d1b0-e9d8
? 3e422dc6..CSeq: 10972023 INVITE..Contact: <sip:5596 at 226.59.129.221:5060;transport=udp>..User-Agent: FreeSWITCH-mod_sofia
? /1.0.7-hacked-20110119T213949Z..Accept: application/sdp..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..Cont
? ent-Type: application/sdp..Content-Disposition: session..Content-Length: 247..X-FS-Display-Name: 5596..X-FS-Display-Numb
? er: sip:5596 at 226.59.129.221..X-FS-Support: update_display..Remote-Party-ID: "5596" <sip:5596 at 226.59.129.221>;party=calli
? ng;privacy=off;screen=no....v=0..o=FreeSWITCH 1302589491 1302589492 IN IP4 226.59.129.221..s=FreeSWITCH..c=IN IP4 226.59
? .129.221..t=0 0..m=audio 26638 RTP/AVP 3 101 13..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16
? ..a=rtpmap:13 CN/8000..a=ptime:20..
#
U 226.59.129.221:5060 -> 226.59.129.223:5060
? SIP/2.0 200 OK..Via: SIP/2.0/UDP 226.59.129.223;rport=5060;branch=z9hG4bKtp8er37QNaS2H..From: "unknown" <sip:1009 at 226.59
? .129.223>;tag=gv7BrXDacFUec..To: <sip:5596 at 226.59.129.221:5060>;tag=4XSZy8tKX129p..Call-ID: 8a42e048-dfae-122e-d1b0-e9d8
? 3e422dc6..CSeq: 10972023 INVITE..Contact: <sip:5596 at 226.59.129.221:5060;transport=udp>..User-Agent: FreeSWITCH-mod_sofia
? /1.0.7-hacked-20110119T213949Z..Accept: application/sdp..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..Cont
? ent-Type: application/sdp..Content-Disposition: session..Content-Length: 247..X-FS-Display-Name: 5596..X-FS-Display-Numb
? er: sip:5596 at 226.59.129.221..X-FS-Support: update_display..Remote-Party-ID: "5596" <sip:5596 at 226.59.129.221>;party=calli
? ng;privacy=off;screen=no....v=0..o=FreeSWITCH 1302589491 1302589492 IN IP4 226.59.129.221..s=FreeSWITCH..c=IN IP4 226.59
? .129.221..t=0 0..m=audio 26638 RTP/AVP 3 101 13..a=rtpmap:3 GSM/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16
? ..a=rtpmap:13 CN/8000..a=ptime:20..

-----Original Message-----

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale

Sent: Tuesday, April 12, 2011 8:21 PM

To: FreeSWITCH Users Help

Subject: Re: [Freeswitch-users] FS does not repose to SIP OK message.



The ack is not being received.? Try your trace from the other side.

Try finding the misconfiguration and the NAT or SIP alg
on your network.

?

?

?

?

On Tue, Apr 12, 2011 at 3:00 PM, Gary Chen
<gchen00 at insightbb.com> wrote:

> Just update my test FS to newest snapshot:
FreeSWITCH Version 1.0.head

> (git-5310735 2011-04-07 15-47-30 -0500)

> I am using SJphone softphone to call into my test
FS1. This FS1 then forward

> the call to another FS2.

> FS2 will answer the call and start Music On
Hold.? Basically I am using

> SJphone to initiate a SIP call? and let FS2 to
answer the call with Music On

> Hold.

> It is working with older version of FS. Now after
update to this newest

> version. The call sometime will not go through. The
SJPhone just keep

> ringing until timeout. This happens maybe on half of
the calls. I also tried

> to use Asterisk to replace FS2 for Music On Hold and
it did the same thing.

> The following is the part of console sofia log info:

> 2011-04-12 15:29:06.285484 [DEBUG] mod_sofia.c:84

> sofia/internal/5596 at 226.59.129.221:5060 SOFIA INIT

> nua: nh_create_handle: entering

> nua: nua_handle_bind: entering

> nua: nua_invite: entering

> nua(0x18cc2c80): sent signal r_invite

> 2011-04-12 15:29:06.285484 [DEBUG] mod_sofia.c:124

> (sofia/internal/5596 at 226.59.129.221:5060) State
Change CS_INIT -> CS_ROUTING

> 2011-04-12 15:29:06.285484 [DEBUG]
switch_core_session.c:1116 Send signal

> sofia/internal/5596 at 226.59.129.221:5060 [BREAK]

> 2011-04-12 15:29:06.285484 [DEBUG]
switch_core_state_machine.c:361

> (sofia/internal/5596 at 226.59.129.221:5060) State INIT
going to sleep

> 2011-04-12 15:29:06.285484 [DEBUG]
switch_core_state_machine.c:325

> (sofia/internal/5596 at 226.59.129.221:5060) Running
State Change CS_ROUTING

> nua(0x18cc2c80): recv signal r_invite

> 2011-04-12 15:29:06.285484 [DEBUG]
switch_channel.c:1668

> (sofia/internal/5596 at 226.59.129.221:5060) Callstate
Change DOWN -> RINGING

> nua: nua_stack_set_params: entering

> soa_clone(static::0x18c66f60, 0x18c474e0,
0x18cc2c80) called

> soa_set_params(static::0x2aaabc072a60, ...) called

> soa_set_params(static::0x2aaabc072a60, ...) called

> soa_set_user_sdp(static::0x2aaabc072a60, (nil),
0x18ccb7b7, -1) called

> soa_set_capability_sdp(static::0x2aaabc072a60,
(nil), 0x18ccb7b7, -1) called

> 2011-04-12 15:29:06.285484 [DEBUG]
switch_core_state_machine.c:364

> (sofia/internal/5596 at 226.59.129.221:5060) State
ROUTING

> 2011-04-12 15:29:06.285484 [DEBUG] mod_sofia.c:147

> sofia/internal/5596 at 226.59.129.221:5060 SOFIA
ROUTING

> 2011-04-12 15:29:06.285484 [DEBUG]
switch_ivr_originate.c:66

> (sofia/internal/5596 at 226.59.129.221:5060) State
Change CS_ROUTING ->

> CS_CONSUME_MEDIA

> nua(0x18cc2c80): adding session usage

> 2011-04-12 15:29:06.285484 [DEBUG]
switch_core_session.c:1116 Send signal

> sofia/internal/5596 at 226.59.129.221:5060 [BREAK]

> 2011-04-12 15:29:06.285484 [DEBUG]
switch_core_state_machine.c:364

> (sofia/internal/5596 at 226.59.129.221:5060) State
ROUTING going to sleep

> 2011-04-12 15:29:06.285484 [DEBUG] switch_core_state_machine.c:325

> (sofia/internal/5596 at 226.59.129.221:5060) Running
State Change

> CS_CONSUME_MEDIA

> nta_leg_tcreate(0x2aaaac05e220)

> soa_init_offer_answer(static::0x2aaabc072a60) called

> soa_generate_offer(static::0x2aaabc072a60, 0) called

> soa_static_offer_answer_action(0x2aaabc072a60,
soa_generate_offer): called

> soa_static(0x2aaabc072a60, soa_generate_offer):
generating local description

> soa_static(0x2aaabc072a60, soa_generate_offer):
upgrade with local

> description

> 2011-04-12 15:29:06.285484 [DEBUG]
switch_core_state_machine.c:383

> (sofia/internal/5596 at 226.59.129.221:5060) State
CONSUME_MEDIA

> soa_sdp_mode_set(0x406f6c60, (nil), ""):
called

> 2011-04-12 15:29:06.285484 [DEBUG]
switch_core_state_machine.c:383

> (sofia/internal/5596 at 226.59.129.221:5060) State
CONSUME_MEDIA going to sleep

> soa_static(0x2aaabc072a60, soa_generate_offer):
storing local description

> soa_get_local_sdp(static::0x2aaabc072a60, [(nil)],
[0x406f6dc8],

> [0x406f6dd4]) called

> nta: selecting scheme sip

> tport_tsend(0x18c69750) tpn = */226.59.129.221:5060

> tport_resolve addrinfo = 226.59.129.221:5060

> tport_by_addrinfo(0x18c69750): not found by name
*/226.59.129.221:5060

> tport_vsend(0x18c69750): 1245 bytes of 1245 to
udp/226.59.129.221:5060

> tport_vsend returned 1245

> nta: sent INVITE (10982209) to */226.59.129.221:5060

> tport_pend(0x18c69750): pending 0x2aaaac059c30 for
udp/226.59.129.223:5060

> (already 0)

> nta: timer set to 32000 ms

> nta: timer shortened to 1000 ms

> nua(0x18cc2c80): call state changed: init ->
calling, sent offer

> soa_get_local_sdp(static::0x2aaabc072a60,
[0x406f6db8], [0x406f6db0],

> [(nil)]) called

> nua(0x18cc2c80): event i_state INVITE sent

> nua: nua_application_event: entering

> 2011-04-12 15:29:06.285484 [DEBUG] sofia.c:4761
Channel

> sofia/internal/5596 at 226.59.129.221:5060 entering
state [calling][0]

> nua: nua_handle_magic: entering

> tport_wakeup_pri(0x18c69750): events IN

> tport_recv_event(0x18c69750)

> tport_recv_iovec(0x18c69750) msg 0x18c87620 from
(udp/226.59.129.223:5060)

> has 344 bytes, veclen = 1

> tport_deliver(0x18c69750): msg 0x18c87620 (344
bytes) from

> udp/226.59.129.221:5060/sip next=(nil)

> nta: received 100 Trying for INVITE (10982209)

> nta: 100 Trying is going to a transaction

> nta_outgoing: RTT is 0.765 ms

> tport_release(0x18c69750): 0x2aaaac059c30 by
0x2aaabc05da60 with 0x18c87620

> (preliminary)

> tport_wakeup_pri(0x18c69750): events IN

> tport_recv_event(0x18c69750)

> tport_recv_iovec(0x18c69750) msg 0x2aaabc070b20 from

> (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1

> tport_deliver(0x18c69750): msg 0x2aaabc070b20 (1235
bytes) from

> udp/226.59.129.221:5060/sip next=(nil)

> nta: received 200 OK for INVITE (10982209)

> nta: 200 OK is going to a transaction

> tport_release(0x18c69750): 0x2aaaac059c30 by
0x2aaabc05da60 with

> 0x2aaabc070b20

> soa_set_remote_sdp(static::0x2aaabc072a60, (nil),
0x2aaabc0712ac, 247)

> called

> soa_process_answer(static::0x2aaabc072a60) called

> soa_static_offer_answer_action(0x2aaabc072a60,
soa_process_answer): called

> soa_sdp_mode_set(0x2aaabc070240, 0x2aaabc073500,
""): called

> soa_static(0x2aaabc072a60, soa_process_answer):
upgrade codecs with remote

> description

> soa_static(0x2aaabc072a60, soa_process_answer):
storing local description

> soa_activate(static::0x2aaabc072a60, (nil)) called

> nua(0x18cc2c80): INVITE: processed SDP answer in 200
OK

> nua(0x18cc2c80): event r_invite 200 OK

> nua(0x18cc2c80): call state changed: calling ->
completing, received answer

> soa_get_remote_sdp(static::0x2aaabc072a60,
[0x406f6828], [0x406f6820],

> [(nil)]) called

> soa_get_params(static::0x2aaabc072a60, ...) called

> nua: nua_application_event: entering

> nua(0x18cc2c80): event i_state 200 OK

> 2011-04-12 15:29:06.294311 [INFO] sofia.c:740

> sofia/internal/5596 at 226.59.129.221:5060 Update
Callee ID to "5596"

> <sip:5596 at 226.59.129.221>

> nta: timer not set

> tport_wakeup_pri(0x18c69750): events IN

> tport_recv_event(0x18c69750)

> tport_recv_iovec(0x18c69750) msg 0x2aaab4022620 from

> (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1

> tport_deliver(0x18c69750): msg 0x2aaab4022620 (1235
bytes) from

> udp/226.59.129.221:5060/sip next=(nil)

> nta: received 200 OK for INVITE (10982209)

> nta: 200 OK is going to a transaction

> nta: 200 OK is duplicate response to 10982209 INVITE

> ??????? Via:
SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg

> nta: timer set next to 31009 ms

> tport_wakeup_pri(0x18c69750): events IN

> tport_recv_event(0x18c69750)

> tport_recv_iovec(0x18c69750) msg 0x2aaab4022620 from

> (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1

> tport_deliver(0x18c69750): msg 0x2aaab4022620 (1235
bytes) from

> udp/226.59.129.221:5060/sip next=(nil)

> nta: received 200 OK for INVITE (10982209)

> nta: 200 OK is going to a transaction

> nta: 200 OK is duplicate response to 10982209 INVITE

> ??????? Via:
SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg

> tport_wakeup_pri(0x18c69750): events IN

> tport_recv_event(0x18c69750)

> tport_recv_iovec(0x18c69750) msg 0x2aaaac060220 from

> (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1

> tport_deliver(0x18c69750): msg 0x2aaaac060220 (1235
bytes) from

> udp/226.59.129.221:5060/sip next=(nil)

> nta: received 200 OK for INVITE (10982209)

> nta: 200 OK is going to a transaction

> nta: 200 OK is duplicate response to 10982209 INVITE

> ??????? Via:
SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg

> tport_wakeup_pri(0x18c69750): events IN

> tport_recv_event(0x18c69750)

> tport_recv_iovec(0x18c69750) msg 0x2aaaac060220 from

> (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1

> tport_deliver(0x18c69750): msg 0x2aaaac060220 (1235
bytes) from

> udp/226.59.129.221:5060/sip next=(nil)

> nta: received 200 OK for INVITE (10982209)

> nta: 200 OK is going to a transaction

> nta: 200 OK is duplicate response to 10982209 INVITE

> ??????? Via:
SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg

> tport_wakeup_pri(0x18c69750): events IN

> tport_recv_event(0x18c69750)

> tport_recv_iovec(0x18c69750) msg 0x2aaaac060220 from

> (udp/226.59.129.223:5060) has 1235 bytes, veclen = 1

> tport_deliver(0x18c69750): msg 0x2aaaac060220 (1235
bytes) from

> udp/226.59.129.221:5060/sip next=(nil)

> nta: received 200 OK for INVITE (10982209)

> nta: 200 OK is going to a transaction

> nta: 200 OK is duplicate response to 10982209 INVITE

> ??????? Via:
SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg

> tport_wakeup_pri(0x18c69750): events IN

> tport_recv_event(0x18c69750)

> tport_recv_iovec(0x18c69750) msg 0x18c87620 from
(udp/226.59.129.223:5060)

> has 1235 bytes, veclen = 1

> tport_deliver(0x18c69750): msg 0x18c87620 (1235
bytes) from

> udp/226.59.129.221:5060/sip next=(nil)

> nta: received 200 OK for INVITE (10982209)

> nta: 200 OK is going to a transaction

> nta: 200 OK is duplicate response to 10982209 INVITE

> ??????? Via:
SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg

> tport_wakeup_pri(0x18c69750): events IN

> tport_recv_event(0x18c69750)

> tport_recv_iovec(0x18c69750) msg 0x18c87620 from
(udp/226.59.129.223:5060)

> has 1235 bytes, veclen = 1

> tport_deliver(0x18c69750): msg 0x18c87620 (1235
bytes) from

> udp/226.59.129.221:5060/sip next=(nil)

> nta: received 200 OK for INVITE (10982209)

> nta: 200 OK is going to a transaction

> nta: 200 OK is duplicate response to 10982209 INVITE

> ??????? Via:
SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg

> tport_wakeup_pri(0x18c69750): events IN

> tport_recv_event(0x18c69750)

> tport_recv_iovec(0x18c69750) msg 0x18c87620 from
(udp/226.59.129.223:5060)

> has 1235 bytes, veclen = 1

> tport_deliver(0x18c69750): msg 0x18c87620 (1235
bytes) from

> udp/226.59.129.221:5060/sip next=(nil)

> nta: received 200 OK for INVITE (10982209)

> nta: 200 OK is going to a transaction

> nta: 200 OK is duplicate response to 10982209 INVITE

> ??????? Via:
SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg

> tport_wakeup_pri(0x18c4dcc0): events IN

> tport_recv_event(0x18c4dcc0)

> tport_recv_iovec(0x18c4dcc0) msg 0x2aaaac060220 from

> (udp/226.59.129.223:5080) has 876 bytes, veclen = 1

> tport_deliver(0x18c4dcc0): msg 0x2aaaac060220 (876
bytes) from

> udp/226.59.139.61:5080/sip next=(nil)

> nta: received INVITE
sip:5025155596 at fs2000.lightyear.net SIP/2.0 (CSeq 1)

> nta: INVITE (1) going to existing INVITE transaction

> nta: re-received INVITE request, retransmitting 100
reply

> tport_tsend(0x18c4dcc0) tpn = UDP/226.59.139.61:5060

> tport_resolve addrinfo = 226.59.139.61:5060

> tport_by_addrinfo(0x18c4dcc0): not found by name
UDP/226.59.139.61:5060

> tport_vsend(0x18c4dcc0): 397 bytes of 397 to
udp/226.59.139.61:5060

> tport_vsend returned 397

> tport_wakeup_pri(0x18c69750): events IN

> tport_recv_event(0x18c69750)

> tport_recv_iovec(0x18c69750) msg 0x18c87620 from
(udp/226.59.129.223:5060)

> has 1235 bytes, veclen = 1

> tport_deliver(0x18c69750): msg 0x18c87620 (1235
bytes) from

> udp/226.59.129.221:5060/sip next=(nil)

> nta: received 200 OK for INVITE (10982209)

> nta: 200 OK is going to a transaction

> nta: 200 OK is duplicate response to 10982209 INVITE

> ??????? Via:
SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg

> tport_wakeup_pri(0x18c69750): events IN

> tport_recv_event(0x18c69750)

> tport_recv_iovec(0x18c69750) msg 0x18c87620 from
(udp/226.59.129.223:5060)

> has 1235 bytes, veclen = 1

> tport_deliver(0x18c69750): msg 0x18c87620 (1235
bytes) from

> udp/226.59.129.221:5060/sip next=(nil)

> nta: received 200 OK for INVITE (10982209)

> nta: 200 OK is going to a transaction

> nta: 200 OK is duplicate response to 10982209 INVITE

> ??????? Via:
SIP/2.0/UDP 226.59.129.223 ;branch=z9hG4bKa9ee6QBjeKFtg

> 2011-04-12 15:29:36.002806 [DEBUG]
switch_channel.c:2563

> (sofia/internal/5596 at 226.59.129.221:5060) Callstate
Change RINGING -> HANGUP

> 2011-04-12 15:29:36.002806 [NOTICE]
switch_ivr_originate.c:3329 Hangup

> sofia/internal/5596 at 226.59.129.221:5060
[CS_CONSUME_MEDIA] [NO_ANSWER]

> 2011-04-12 15:29:36.002806 [DEBUG]
switch_channel.c:2579 Send signal

> sofia/internal/5596 at 226.59.129.221:5060 [KILL]

> 2011-04-12 15:29:36.002806 [DEBUG]
switch_core_session.c:1116 Send signal

> sofia/internal/5596 at 226.59.129.221:5060 [BREAK]

> 2011-04-12 15:29:36.002806 [INFO] mod_dptools.c:2647
Originate Failed.

> Cause: NO_ANSWER

> 2011-04-12 15:29:36.002806 [DEBUG]
switch_cpp.cpp:988

> sofia/external/1009 at fs2000.lightyear.net
destroy/unlink session from object

> 2011-04-12 15:29:36.002806 [DEBUG]
switch_core_state_machine.c:325

> (sofia/internal/5596 at 226.59.129.221:5060) Running
State Change CS_HANGUP

> 2011-04-12 15:29:36.002806 [DEBUG]
switch_core_state_machine.c:565

> (sofia/internal/5596 at 226.59.129.221:5060) State
HANGUP

> EXECUTE sofia/external/1009 at fs2000.lightyear.net
answer()

> 2011-04-12 15:29:36.003897 [DEBUG] sofia_glue.c:3014
AUDIO RTP

> [sofia/external/1009 at fs2000.lightyear.net]
226.59.129.223 port 27272 ->

> 226.59.139.61 port 49420 codec: 3 ms: 20

> nua: nua_handle_magic: entering

> 2011-04-12 15:29:36.003897 [DEBUG] switch_rtp.c:1623
Starting timer [soft]

> 160 bytes per 20ms

> nua: nua_application_event: entering

> 2011-04-12 15:29:36.003897 [DEBUG] mod_sofia.c:457
Channel

> sofia/internal/5596 at 226.59.129.221:5060 hanging up,
cause: NO_ANSWER

> 2011-04-12 15:29:36.004809 [DEBUG] sofia_glue.c:3276
Set 2833 dtmf send

> payload to 101

> 2011-04-12 15:29:36.004809 [DEBUG] sofia_glue.c:3281
Set 2833 dtmf receive

> payload to 101

> 2011-04-12 15:29:36.004809 [DEBUG] mod_sofia.c:681
Local SDP

> sofia/external/1009 at fs2000.lightyear.net:

> v=0

> o=FreeSWITCH 1302609304 1302609305 IN IP4 226.59.129.223

> s=FreeSWITCH

> c=IN IP4 226.59.129.223

>?

>?

>?

>?

>?

> _______________________________________________

> FreeSWITCH-users mailing list

> FreeSWITCH-users at lists.freeswitch.org

>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

> http://www.freeswitch.org

>?

>?

?

?

?

-- 

Anthony Minessale II

?

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

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From gchen00 at insightbb.com  Wed Apr 13 16:44:29 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Wed, 13 Apr 2011 08:44:29 -0400
Subject: [Freeswitch-users] Event queue is full.
Message-ID: <fc6fdc1b147df.4da5626d@insightbb.com>

FreeSWITCH Version 1.0.head (git-5310735 2011-04-07 15-47-30 -0500)

Last night I start Sipp to sent calls to my test? FS1 and FS1 in turn? forward calls to another FS2 playing Music On Hold. I got about 300 concurrent calls with media going through FS1 before I left. 
This morning, the FS1 is not responding. I fs_cli into it and got a lot of messages like this:
2011-04-13 08:12:11.766570 [CRIT] switch_event.c:1367 Event queue is full!
x2011-04-13 08:12:11.798571 [CRIT] switch_event.c:1367 Event queue is full!
2011-04-13 08:12:11.809567 [CRIT] switch_event.c:1367 Event queue is full!
2011-04-13 08:12:11.809567 [CRIT] switch_event.c:1367 Event queue is full!
2011-04-13 08:12:11.816565 [CRIT] switch_event.c:1367 Event queue is full!
2011-04-13 08:12:11.827574 [CRIT] switch_event.c:1367 Event queue is full!
....

What is the possible cause of this? The FS1 is still running, How do I cleanup the Event queue beside reboot FS?

Gary



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From eric at loopfx.com  Wed Apr 13 16:48:51 2011
From: eric at loopfx.com (Eric Beard)
Date: Wed, 13 Apr 2011 08:48:51 -0400
Subject: [Freeswitch-users] Lua session originate - freeswitch crash
In-Reply-To: <BANLkTi=f8cRhf1QL2tb4C28mv3E0MSEY4g@mail.gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DCA92D3D@doctorbob.loop.local>
	<1302633709973-6266239.post@n2.nabble.com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92E03@doctorbob.loop.local>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92EA1@doctorbob.loop.local>
	<BANLkTi=f8cRhf1QL2tb4C28mv3E0MSEY4g@mail.gmail.com>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DCA93058@doctorbob.loop.local>

Thanks Anthony,

I will update to latest.  I last did a pull about 2 weeks ago, FreeSWITCH Version 1.0.head (git-8c5586b 2011-04-01 14-22-43 -0500).

If I shouldn't be doing this in this script, then what is the recommended way to originate an outbound IVR call?

My current production application uses Microsoft Speech Server to send outbound customer service calls, with a B2BUA I wrote to communicate with terminators.  I'm switching my B2BUA out for FreeSwitch, and it's working great.  But if possible I'd like to remove MSS from the equation altogether.

Lua looks like a good way to control the call workflow.  Should I be using ESL from an external application instead?

Thanks for your time,

-----------------------
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 Anthony Minessale
Sent: Tuesday, April 12, 2011 7:28 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Lua session originate - freeswitch crash

1) You are not running the latest GIT.  Your scripts works fine on my
development box.
2) the originate method is deprecated.  you can just do
     local new_session =
freeswitch.Session("sofia/gateway/affinity/17277762768", session);
3) In either case you should not be doing what you do in this script
because the whole time you are playing the file to new_session the
original session is blocked not reading or writing any audio.


On Tue, Apr 12, 2011 at 3:02 PM, Eric Beard <eric at loopfx.com> wrote:
> Ok, so I got a little further by trial and error, and realized the 4th arg is due to lua function calling syntax. ?If you use the dot notation on a function declared with a colon, it expects "this" as the first arg.
>
> My current code looks like this:
>
> -- Originate an outbound call
> local new_session = freeswitch.Session();
> new_session:originate(session, "sofia/gateway/affinity/17277762768", 60);
> new_session:waitForAnswer(session);
> prompt = "/home/eric/test1.wav"
> freeswitch.consoleLog("INFO", "About to play prompt file " .. prompt .."\n")
> new_session:streamFile(prompt)
> new_session:hangup()
>
> The scary thing about this code is that it crashed freeSwitch when I ran it.
>
> This is the last thing I saw in the logs:
>
>
> -----------------------
> 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 Eric Beard
> Sent: Tuesday, April 12, 2011 3:15 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Lua session originate
>
> Thanks Jeff,
>
> When I try NULL I get this:
>
> freeswitch at internal> 2011-04-12 14:49:31.582049 [ERR] mod_lua.cpp:191 Error in originate (arg 2), ex
> pected 'CoreSession *' got 'string'
> stack traceback:
> ? ? ? ?[C]: in function 'originate'
> ? ? ? ?/usr/local/freeswitch/scripts/test_originate.lua:3: in main chunk
>
> I've been digging through the source code, and I can't find where 4 args are required. ?I see this in freeswitch_lua.cpp:
>
> int Session::originate(CoreSession *a_leg_session, char *dest, int timeout)
> {
> ? ? ? ?int x = CoreSession::originate(a_leg_session, dest, timeout);
>
> ? ? ? ?if (x) {
> ? ? ? ? ? ? ? ?setLUA(L);
> ? ? ? ?}
>
> ? ? ? ?return x;
> }
>
> -----------------------
> 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 Jeff Lenk
> Sent: Tuesday, April 12, 2011 2:42 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Lua session originate
>
> The fourth parameter is the switch_state_handler_table reference and for most
> uses(script language) you should just pass null ?but I dont use lua so I
> dont know the details on that.
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Lua-session-originate-tp6266121p6266239.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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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


From rhuddleston at gmail.com  Wed Apr 13 17:05:43 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Wed, 13 Apr 2011 09:05:43 -0400
Subject: [Freeswitch-users] Lua session originate - freeswitch crash
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DCA93058@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DCA92D3D@doctorbob.loop.local>
	<1302633709973-6266239.post@n2.nabble.com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92E03@doctorbob.loop.local>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92EA1@doctorbob.loop.local>
	<BANLkTi=f8cRhf1QL2tb4C28mv3E0MSEY4g@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA93058@doctorbob.loop.local>
Message-ID: <0d4b01cbf9db$7fe1ce50$7fa56af0$@com>

I'm sort of in the same boat... I've been working on a project completely in
LUA - using mod_lcr / mod_nibblebill etc - and another contributor wrote to
migrate to ESL... Not that I'm against a challenge - just disappointed I
spent so much time writing the LUA script to now have to convert to ESL.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eric
Beard
Sent: Wednesday, April 13, 2011 8:49 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Lua session originate - freeswitch crash

Thanks Anthony,

I will update to latest.  I last did a pull about 2 weeks ago, FreeSWITCH
Version 1.0.head (git-8c5586b 2011-04-01 14-22-43 -0500).

If I shouldn't be doing this in this script, then what is the recommended
way to originate an outbound IVR call?

My current production application uses Microsoft Speech Server to send
outbound customer service calls, with a B2BUA I wrote to communicate with
terminators.  I'm switching my B2BUA out for FreeSwitch, and it's working
great.  But if possible I'd like to remove MSS from the equation altogether.

Lua looks like a good way to control the call workflow.  Should I be using
ESL from an external application instead?

Thanks for your time,

-----------------------
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 Anthony
Minessale
Sent: Tuesday, April 12, 2011 7:28 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Lua session originate - freeswitch crash

1) You are not running the latest GIT.  Your scripts works fine on my
development box.
2) the originate method is deprecated.  you can just do
     local new_session =
freeswitch.Session("sofia/gateway/affinity/17277762768", session);
3) In either case you should not be doing what you do in this script
because the whole time you are playing the file to new_session the
original session is blocked not reading or writing any audio.


On Tue, Apr 12, 2011 at 3:02 PM, Eric Beard <eric at loopfx.com> wrote:
> Ok, so I got a little further by trial and error, and realized the 4th arg
is due to lua function calling syntax. ?If you use the dot notation on a
function declared with a colon, it expects "this" as the first arg.
>
> My current code looks like this:
>
> -- Originate an outbound call
> local new_session = freeswitch.Session();
> new_session:originate(session, "sofia/gateway/affinity/17277762768", 60);
> new_session:waitForAnswer(session);
> prompt = "/home/eric/test1.wav"
> freeswitch.consoleLog("INFO", "About to play prompt file " .. prompt
.."\n")
> new_session:streamFile(prompt)
> new_session:hangup()
>
> The scary thing about this code is that it crashed freeSwitch when I ran
it.
>
> This is the last thing I saw in the logs:
>
>
> -----------------------
> 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 Eric
Beard
> Sent: Tuesday, April 12, 2011 3:15 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Lua session originate
>
> Thanks Jeff,
>
> When I try NULL I get this:
>
> freeswitch at internal> 2011-04-12 14:49:31.582049 [ERR] mod_lua.cpp:191
Error in originate (arg 2), ex
> pected 'CoreSession *' got 'string'
> stack traceback:
> ? ? ? ?[C]: in function 'originate'
> ? ? ? ?/usr/local/freeswitch/scripts/test_originate.lua:3: in main chunk
>
> I've been digging through the source code, and I can't find where 4 args
are required. ?I see this in freeswitch_lua.cpp:
>
> int Session::originate(CoreSession *a_leg_session, char *dest, int
timeout)
> {
> ? ? ? ?int x = CoreSession::originate(a_leg_session, dest, timeout);
>
> ? ? ? ?if (x) {
> ? ? ? ? ? ? ? ?setLUA(L);
> ? ? ? ?}
>
> ? ? ? ?return x;
> }
>
> -----------------------
> 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 Jeff
Lenk
> Sent: Tuesday, April 12, 2011 2:42 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Lua session originate
>
> The fourth parameter is the switch_state_handler_table reference and for
most
> uses(script language) you should just pass null ?but I dont use lua so I
> dont know the details on that.
>
> --
> View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Lua-session-originate-tp626612
1p6266239.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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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



From brian at freeswitch.org  Wed Apr 13 17:29:55 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Apr 2011 08:29:55 -0500
Subject: [Freeswitch-users] Shared Call appearence, barging and presence
In-Reply-To: <1663096009.221390.1302663555843.JavaMail.root@zinnia1>
References: <1663096009.221390.1302663555843.JavaMail.root@zinnia1>
Message-ID: <76EDF350-3983-42DD-8FCD-726AA5BEBB62@freeswitch.org>

you might wanna make sure you have this set to shared because if all the little phone icons aren't split in two then you'll never get this working... ONE line not knowing its shared will cause them all to not function properly.

/b

On Apr 12, 2011, at 9:59 PM, Gourav Vohra wrote:

>        reg.1.type="private" 



From brian at freeswitch.org  Wed Apr 13 17:31:54 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Apr 2011 08:31:54 -0500
Subject: [Freeswitch-users] FS does not repose to SIP OK message.
In-Reply-To: <fc7f86bf12222.4da55fcb@insightbb.com>
References: <fc7f86bf12222.4da55fcb@insightbb.com>
Message-ID: <22576037-ECFC-4E4C-B688-B06A30A56C25@freeswitch.org>

Please redo the console log like your original one with 'sofia global siptrace on' and 'sofia loglevel all 9' .. this trace below is just chicken scratch and hard to follow.

/b

On Apr 13, 2011, at 7:33 AM, Gary Chen wrote:

> 
> 
> The FS1 should response back with ACK. It did not. 
> The strange thing is that It works sometime.Here is the SIP trace from ngrep for both good and bad calls on FS1 server:SJPhone IP: 226.59.139.61FS1 IP:226.59.129.223FS2 IP:226.59.129.221You can see that FS2 sent several 200 OK and FS1 never reply back.
> Good sip call:U 226.59.139.61:5060 -> 226.59.129.223:5080
>   INVITE sip:5025155596 at fs2000.lightyear.net SIP/2.0..Via: SIP/2.0/UDP 226.59.139.61;branch=z9hG4bKd8318b3d000007494da459a
>   900006fac00000273;rport..From: "unknown" <sip:1009 at fs2000.lightyear.net>;tag=a344c583a24..To: <sip:5025155596 at fs2000.lig
>   htyear.net>..Contact: <sip:1009 at 226.59.139.61>..Call-ID: 56BF61B3189F4EDEB8C9067D5823C8480xd8318b3d..CSeq: 1 INVITE..Max
>   -Forwards: 70..User-Agent: SJphone/1.65.377a (SJ Labs)..Content-Length: 368..Content-Type: application/sdp..Supported: r
>   eplaces,norefersub,timer....v=0..o=- 3511605289 3511605289 IN IP4 226.59.139.61..s=SJphone..c=IN IP4 226.59.139.61..t=0
>   0..m=audio 49352 RTP/AVP 3 97 98 8 0 101..c=IN IP4 226.59.139.61..a=rtpmap:3 GSM/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:9
>   8 iLBC/8000..a=fmtp:98 mode=20..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:10
>   1 0-16..a=setup:active..a=sendrecv..



From brian at freeswitch.org  Wed Apr 13 17:34:14 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 13 Apr 2011 08:34:14 -0500
Subject: [Freeswitch-users] Event queue is full.
In-Reply-To: <fc6fdc1b147df.4da5626d@insightbb.com>
References: <fc6fdc1b147df.4da5626d@insightbb.com>
Message-ID: <58F151E1-3F41-4A0B-BCE1-FF40385E0653@freeswitch.org>

Sounds like an old code base.. you sure that other box is using the latest code?

/b

On Apr 13, 2011, at 7:44 AM, Gary Chen wrote:

> FreeSWITCH Version 1.0.head (git-5310735 2011-04-07 15-47-30 -0500)
> 
> Last night I start Sipp to sent calls to my test  FS1 and FS1 in turn  forward calls to another FS2 playing Music On Hold. I got about 300 concurrent calls with media going through FS1 before I left. 
> This morning, the FS1 is not responding. I fs_cli into it and got a lot of messages like this:
> 2011-04-13 08:12:11.766570 [CRIT] switch_event.c:1367 Event queue is full!
> x2011-04-13 08:12:11.798571 [CRIT] switch_event.c:1367 Event queue is full!
> 2011-04-13 08:12:11.809567 [CRIT] switch_event.c:1367 Event queue is full!
> 2011-04-13 08:12:11.809567 [CRIT] switch_event.c:1367 Event queue is full!
> 2011-04-13 08:12:11.816565 [CRIT] switch_event.c:1367 Event queue is full!
> 2011-04-13 08:12:11.827574 [CRIT] switch_event.c:1367 Event queue is full!



From gchen00 at insightbb.com  Wed Apr 13 17:56:45 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Wed, 13 Apr 2011 09:56:45 -0400
Subject: [Freeswitch-users] Event queue is full.
Message-ID: <fc6f970017d8e.4da5735d@insightbb.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: Wednesday, April 13, 2011 9:34 AM

To: FreeSWITCH Users Help

Subject: Re: [Freeswitch-users] Event queue is full.

?

Sounds like an old code base.. you sure that other box is
using the latest code?

?

/b

?

On Apr 13, 2011, at 7:44 AM, Gary Chen wrote:

?

> FreeSWITCH Version 1.0.head (git-5310735 2011-04-07
15-47-30 -0500)

> 

> Last night I start Sipp to sent calls to my
test? FS1 and FS1 in turn? forward calls to another FS2 playing Music On
Hold. I got about 300 concurrent calls with media going through FS1 before I
left. 

> This morning, the FS1 is not responding. I fs_cli
into it and got a lot of messages like this:

> 2011-04-13 08:12:11.766570 [CRIT]
switch_event.c:1367 Event queue is full!

> x2011-04-13 08:12:11.798571 [CRIT]
switch_event.c:1367 Event queue is full!

> 2011-04-13 08:12:11.809567 [CRIT]
switch_event.c:1367 Event queue is full!

> 2011-04-13 08:12:11.809567 [CRIT]
switch_event.c:1367 Event queue is full!

> 2011-04-13 08:12:11.816565 [CRIT]
switch_event.c:1367 Event queue is full!

> 2011-04-13 08:12:11.827574 [CRIT]
switch_event.c:1367 Event queue is full!

?

?Call Flow: Sipp ---> FS1 --->FS2(play music on hold)FS1 version: FreeSWITCH Version 1.0.head (git-5310735 2011-04-07 15-47-30 -0500)FS2 Version: FreeSWITCH Version 1.0.7 (hacked-20110119T213949Z)FS2 has older version. But 'Event queue is full' only happen on FS1. Should have nothing to do with FS2. Right now FS2 still show 304 sessions. That means that it still playing music on 304 calls.I can not run fs_cli -x status because the event queue problem.? 
Gary



_______________________________________________

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

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From eric at loopfx.com  Wed Apr 13 17:59:07 2011
From: eric at loopfx.com (Eric Beard)
Date: Wed, 13 Apr 2011 09:59:07 -0400
Subject: [Freeswitch-users] Lua session originate - freeswitch crash
In-Reply-To: <0d4b01cbf9db$7fe1ce50$7fa56af0$@com>
References: <D4A666272A214C4DBBABFE569CA34CA401DCA92D3D@doctorbob.loop.local>
	<1302633709973-6266239.post@n2.nabble.com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92E03@doctorbob.loop.local>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92EA1@doctorbob.loop.local>
	<BANLkTi=f8cRhf1QL2tb4C28mv3E0MSEY4g@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA93058@doctorbob.loop.local>
	<0d4b01cbf9db$7fe1ce50$7fa56af0$@com>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DCA93100@doctorbob.loop.local>

I've been fiddling around with things this morning and I am using a combination of PHP and Lua.

I use php to simply originate the call (using regular sockets to talk to 8021, not the php mod for ESL, since I can't get it to compile).  In the originate command I specify a Lua script that handles the rest of the call.

I'm using the PHP functions described here:

http://wiki.freeswitch.org/wiki/PHP_Event_Socket

My PHP:


 $fp = event_socket_create("127.0.0.1", 8021, "ClueCon");
 $channelVars = "{ignore_early_media=true,origination_caller_id_number=18778502010}";
 $phoneNum = "17277762768";
 $notificationId = "x";
 $cmd = "bgapi originate $channelVars".
   "sofia/gateway/affinity/".$phoneNum." &lua('test2.lua $notificationId')";
 echo "Sending command: $cmd";
 $response = event_socket_request($fp, $cmd);
 echo $response;
 fclose($fp);

My Lua:


prompt = "/home/eric/test1.wav"
session:streamFile(prompt)
session:hangup()

This is working for the very basic scenario of just waiting for the number to pick up, playing a wav, and then hanging up.  If anybody can point out any major showstoppers that would prevent me from handling significant user interaction, I'd love to hear about them now.  One of the things I'd like to do is incorporate pocketsphinx so that I can do voice reco.  Is that possible from Lua?

Thanks,

-----------------------
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 Robert Huddleston
Sent: Wednesday, April 13, 2011 9:06 AM
To: 'FreeSWITCH Users Help'
Subject: Re: [Freeswitch-users] Lua session originate - freeswitch crash

I'm sort of in the same boat... I've been working on a project completely in
LUA - using mod_lcr / mod_nibblebill etc - and another contributor wrote to
migrate to ESL... Not that I'm against a challenge - just disappointed I
spent so much time writing the LUA script to now have to convert to ESL.


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Eric
Beard
Sent: Wednesday, April 13, 2011 8:49 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Lua session originate - freeswitch crash

Thanks Anthony,

I will update to latest.  I last did a pull about 2 weeks ago, FreeSWITCH
Version 1.0.head (git-8c5586b 2011-04-01 14-22-43 -0500).

If I shouldn't be doing this in this script, then what is the recommended
way to originate an outbound IVR call?

My current production application uses Microsoft Speech Server to send
outbound customer service calls, with a B2BUA I wrote to communicate with
terminators.  I'm switching my B2BUA out for FreeSwitch, and it's working
great.  But if possible I'd like to remove MSS from the equation altogether.

Lua looks like a good way to control the call workflow.  Should I be using
ESL from an external application instead?

Thanks for your time,

-----------------------
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 Anthony
Minessale
Sent: Tuesday, April 12, 2011 7:28 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Lua session originate - freeswitch crash

1) You are not running the latest GIT.  Your scripts works fine on my
development box.
2) the originate method is deprecated.  you can just do
     local new_session =
freeswitch.Session("sofia/gateway/affinity/17277762768", session);
3) In either case you should not be doing what you do in this script
because the whole time you are playing the file to new_session the
original session is blocked not reading or writing any audio.


On Tue, Apr 12, 2011 at 3:02 PM, Eric Beard <eric at loopfx.com> wrote:
> Ok, so I got a little further by trial and error, and realized the 4th arg
is due to lua function calling syntax. ?If you use the dot notation on a
function declared with a colon, it expects "this" as the first arg.
>
> My current code looks like this:
>
> -- Originate an outbound call
> local new_session = freeswitch.Session();
> new_session:originate(session, "sofia/gateway/affinity/17277762768", 60);
> new_session:waitForAnswer(session);
> prompt = "/home/eric/test1.wav"
> freeswitch.consoleLog("INFO", "About to play prompt file " .. prompt
.."\n")
> new_session:streamFile(prompt)
> new_session:hangup()
>
> The scary thing about this code is that it crashed freeSwitch when I ran
it.
>
> This is the last thing I saw in the logs:
>
>
> -----------------------
> 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 Eric
Beard
> Sent: Tuesday, April 12, 2011 3:15 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Lua session originate
>
> Thanks Jeff,
>
> When I try NULL I get this:
>
> freeswitch at internal> 2011-04-12 14:49:31.582049 [ERR] mod_lua.cpp:191
Error in originate (arg 2), ex
> pected 'CoreSession *' got 'string'
> stack traceback:
> ? ? ? ?[C]: in function 'originate'
> ? ? ? ?/usr/local/freeswitch/scripts/test_originate.lua:3: in main chunk
>
> I've been digging through the source code, and I can't find where 4 args
are required. ?I see this in freeswitch_lua.cpp:
>
> int Session::originate(CoreSession *a_leg_session, char *dest, int
timeout)
> {
> ? ? ? ?int x = CoreSession::originate(a_leg_session, dest, timeout);
>
> ? ? ? ?if (x) {
> ? ? ? ? ? ? ? ?setLUA(L);
> ? ? ? ?}
>
> ? ? ? ?return x;
> }
>
> -----------------------
> 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 Jeff
Lenk
> Sent: Tuesday, April 12, 2011 2:42 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Lua session originate
>
> The fourth parameter is the switch_state_handler_table reference and for
most
> uses(script language) you should just pass null ?but I dont use lua so I
> dont know the details on that.
>
> --
> View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/Lua-session-originate-tp626612
1p6266239.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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> 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 gourav at rentec.com  Wed Apr 13 18:28:57 2011
From: gourav at rentec.com (Gourav Vohra)
Date: Wed, 13 Apr 2011 10:28:57 -0400 (EDT)
Subject: [Freeswitch-users] Shared Call appearence, barging and presence
In-Reply-To: <76EDF350-3983-42DD-8FCD-726AA5BEBB62@freeswitch.org>
Message-ID: <1309783927.227302.1302704937950.JavaMail.root@zinnia1>



I have changed all lines to share but I am still encountering the same issue.

I also noticed the following:

After phone2 hangs up the call and picks it up again to make another call and then hangs up again - the light on the polycom phone (phone2) next to x2995 changes to red which would indicate that the line is in use (Since phone1 and phone3 are still on the call) whereas the light should change to red after the phone2 hangs up the call the first time.

If I barge into the call on phone3 and hangup - x2995 on phone3 indicates it's still in use (which is what I would expect in the above scenario).  

Following configs are now on phone2 and phone3.  
 
phone2 - x2995 is on this phone.

<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- Generated reg-advanced.cfg Configuration File -->
<polycomConfig xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="polycomConfig.xsd">
  <call call.transferOnConferenceEnd="1">
    <call.autoAnswer call.autoAnswer.H323="0" call.autoAnswer.SIP="0" call.autoAnswer.micMute="1" call.autoAnswer.ringClass="ringAutoAnswer" call.autoAnswer.videoMute="0">
    </call.autoAnswer>
    <call.autoOffHook call.autoOffHook.1.contact="" call.autoOffHook.1.enabled="0" call.autoOffHook.1.protocol="" call.autoOffHook.2.contact="" call.autoOffHook.2.enabled="0" call.autoOffHook.2.protocol="">
    </call.autoOffHook>
    <call.donotdisturb call.donotdisturb.perReg="0">
    </call.donotdisturb>
    <call.missedCallTracking call.missedCallTracking.1.enabled="1" call.missedCallTracking.2.enabled="1">
    </call.missedCallTracking>
    <call.serverMissedCall call.serverMissedCall.1.enabled="0" call.serverMissedCall.2.enabled="0">
    </call.serverMissedCall>
  </call>
  <device>
    <device.auth device.auth.localUserPassword="">
      <device.auth.localUserPassword device.auth.localUserPassword.set="0">
      </device.auth.localUserPassword>
    </device.auth>
    <device.em device.em.power="">
      <device.em.power device.em.power.set="0">
      </device.em.power>
    </device.em>
  </device>
<reg reg.1.displayName="test user1" 
        reg.1.address="2995" 
        reg.1.label="2995" 
        reg.1.type="shared" 
        reg.1.auth.userId="2995" 
        reg.1.auth.password="zebra" 
        reg.1.ringType="2" 
        reg.1.bargeInEnabled="1"
        reg.1.callsPerLineKey="4" 
        reg.1.csta="0" 
        reg.1.fwd.busy.contact="" 
        reg.1.fwd.busy.status="0" 
        reg.1.fwd.noanswer.contact="" 
        reg.1.fwd.noanswer.ringCount="0" 
        reg.1.fwd.noanswer.status="0" 
        reg.1.lineKeys="1" 
        reg.1.musicOnHold.uri="" 
        reg.1.outboundProxy.failOver.failBack.timeout="3600" 
        reg.1.outboundProxy.failOver.failRegistrationOn="1" 
        reg.1.outboundProxy.failOver.reRegisterOn="0" 
        reg.1.outboundProxy.address="192.168.100.33"
        reg.1.outboundProxy.port="5060" 
        reg.1.protocol.SIP="1" 
        reg.1.ringType="ringer2" 
        reg.1.serverFeatureControl.cf="1" 
        reg.1.serverFeatureControl.dnd="1" 
        reg.1.thirdPartyName="" 
        reg.1.useCompleteUriForRetrieve="1" 
        reg.1.server.1.expires="3600" 
        reg.1.server.1.expires.lineSeize="30" 
        reg.1.server.1.expires.overlap="60" 
        reg.1.server.1.lcs="0" 
        reg.1.server.1.retryMaxCount="3" 
        reg.1.server.1.retryTimeOut="0" 
        reg.1.server.2.expires="3600" 
        reg.1.server.2.expires.lineSeize="30" 
        reg.1.server.2.expires.overlap="60" 
        reg.1.server.2.lcs="0" 
        reg.1.server.2.retryMaxCount="3" 
        reg.1.server.2.retryTimeOut="0">
  </reg>
  <se se.destination="chassis">
    <se.rt>
      <se.rt.modification se.rt.modification.enabled="1">
      </se.rt.modification>
    </se.rt>
  </se>
  <up up.25mm="0" up.accessibilityFeatures="0" up.analogHeadsetOption="0" up.audioMode="0" up.callTypeOrderVoice="0" up.handsfreeMode="1" up.headsetMode="0" up.localClockEnabled="1" up.manualProtocolRouting="1" up.mwiVisible="0" up.onHookDialingEnabled="1" up.useDirectoryNames="0" up.welcomeSoundEnabled="1" up.welcomeSoundOnWarmBootEnabled="0">
    <up.2P5mmJack up.2P5mmJack.mode="0">
    </up.2P5mmJack>
    <up.audioSetup up.audioSetup.auxInput="2" up.audioSetup.auxOutput="2">
    </up.audioSetup>
    <up.backlight up.backlight.idleIntensity="1" up.backlight.onIntensity="3">
    </up.backlight>
    <up.idleBrowser up.idleBrowser.enabled="0">
    </up.idleBrowser>
    <up.manualProtocolRouting up.manualProtocolRouting.softKeys="1">
    </up.manualProtocolRouting>
    <up.offHookAction up.offHookAction.none="0">
    </up.offHookAction>
    <up.pictureFrame up.pictureFrame.folder="" up.pictureFrame.timePerImage="5">
    </up.pictureFrame>
    <up.screenSaver up.screenSaver.enabled="0" up.screenSaver.waitTime="15">
    </up.screenSaver>
    <up.toneControl up.toneControl.bass="0" up.toneControl.treble="0">
    </up.toneControl>
  </up>
  <voIpProt>
    <voIpProt.SIP>
      <voIpProt.SIP.assuredService>
        <voIpProt.SIP.assuredService.namespace>
          <voIpProt.SIP.assuredService.namespace.label>
            <voIpProt.SIP.assuredService.namespace.label.UCRdrsn voIpProt.SIP.assuredService.namespace.label.UCRdrsn.1="" voIpProt.SIP.assuredService.namespace.label.UCRdrsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.UCRdrsn>
            <voIpProt.SIP.assuredService.namespace.label.UCRdsn voIpProt.SIP.assuredService.namespace.label.UCRdsn.1="" voIpProt.SIP.assuredService.namespace.label.UCRdsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.UCRdsn>
            <voIpProt.SIP.assuredService.namespace.label.custom voIpProt.SIP.assuredService.namespace.label.custom.1="" voIpProt.SIP.assuredService.namespace.label.custom.2="">
            </voIpProt.SIP.assuredService.namespace.label.custom>
            <voIpProt.SIP.assuredService.namespace.label.drsn voIpProt.SIP.assuredService.namespace.label.drsn.1="" voIpProt.SIP.assuredService.namespace.label.drsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.drsn>
            <voIpProt.SIP.assuredService.namespace.label.dsn voIpProt.SIP.assuredService.namespace.label.dsn.1="" voIpProt.SIP.assuredService.namespace.label.dsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.dsn>
          </voIpProt.SIP.assuredService.namespace.label>
        </voIpProt.SIP.assuredService.namespace>
      </voIpProt.SIP.assuredService>
    </voIpProt.SIP>
  </voIpProt>
</polycomConfig>


phone3 - x2996 and x2995 are on this phone.

<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- Generated reg-advanced.cfg Configuration File -->
<polycomConfig xmlns:xsi="http://www.w3.org/2001/XMLSchema-instance" xsi:noNamespaceSchemaLocation="polycomConfig.xsd">
  <call call.transferOnConferenceEnd="1">
    <call.autoAnswer call.autoAnswer.H323="0" call.autoAnswer.SIP="0" call.autoAnswer.micMute="1" call.autoAnswer.ringClass="ringAutoAnswer" call.autoAnswer.videoMute="0">
    </call.autoAnswer>
    <call.autoOffHook call.autoOffHook.1.contact="" call.autoOffHook.1.enabled="0" call.autoOffHook.1.protocol="" call.autoOffHook.2.contact="" call.autoOffHook.2.enabled="0" call.autoOffHook.2.protocol="">
    </call.autoOffHook>
    <call.donotdisturb call.donotdisturb.perReg="0">
    </call.donotdisturb>
    <call.missedCallTracking call.missedCallTracking.1.enabled="1" call.missedCallTracking.2.enabled="1">
    </call.missedCallTracking>
    <call.serverMissedCall call.serverMissedCall.1.enabled="0" call.serverMissedCall.2.enabled="0">
    </call.serverMissedCall>
  </call>
  <device>
    <device.auth device.auth.localUserPassword="">
      <device.auth.localUserPassword device.auth.localUserPassword.set="0">
      </device.auth.localUserPassword>
    </device.auth>
    <device.em device.em.power="">
      <device.em.power device.em.power.set="0">
      </device.em.power>
    </device.em>
  </device>
<reg reg.1.displayName="test user" 
        reg.1.address="2996" 
        reg.1.label="2996" 
        reg.1.type="shared" 
        reg.1.auth.userId="2996" 
        reg.1.auth.password="zebra" 
        reg.1.ringType="2" 
        reg.1.bargeInEnabled="1"
        reg.1.callsPerLineKey="4" 
        reg.1.csta="0" 
        reg.1.fwd.busy.contact="" 
        reg.1.fwd.busy.status="0" 
        reg.1.fwd.noanswer.contact="" 
        reg.1.fwd.noanswer.ringCount="0" 
        reg.1.fwd.noanswer.status="0" 
        reg.1.lineKeys="1" 
        reg.1.musicOnHold.uri="" 
        reg.1.outboundProxy.failOver.failBack.timeout="3600" 
        reg.1.outboundProxy.failOver.failRegistrationOn="1" 
        reg.1.outboundProxy.failOver.reRegisterOn="0" 
        reg.1.outboundProxy.address="192.168.100.33"
        reg.1.outboundProxy.port="5060" 
        reg.1.protocol.SIP="1" 
        reg.1.ringType="ringer2" 
        reg.1.serverFeatureControl.cf="1" 
        reg.1.serverFeatureControl.dnd="1" 
        reg.1.thirdPartyName="" 
        reg.1.useCompleteUriForRetrieve="1" 
        reg.1.server.1.expires="3600" 
        reg.1.server.1.expires.lineSeize="30" 
        reg.1.server.1.expires.overlap="60" 
        reg.1.server.1.lcs="0" 
        reg.1.server.1.retryMaxCount="3" 
        reg.1.server.1.retryTimeOut="0" 
        reg.1.server.2.expires="3600" 
        reg.1.server.2.expires.lineSeize="30" 
        reg.1.server.2.expires.overlap="60" 
        reg.1.server.2.lcs="0" 
        reg.1.server.2.retryMaxCount="3" 
        reg.1.server.2.retryTimeOut="0"
        reg.2.displayName="test user1" 
        reg.2.address="2995" 
        reg.2.label="2995" 
        reg.2.type="shared" 
        reg.2.auth.userId="2995" 
        reg.2.auth.password="zebra" 
        reg.2.ringType="1" 
        reg.2.bargeInEnabled="1"
        reg.2.callsPerLineKey="4" 
        reg.2.csta="0" 
        reg.2.fwd.busy.contact="" 
        reg.2.fwd.busy.status="0" 
        reg.2.fwd.noanswer.contact="" 
        reg.2.fwd.noanswer.ringCount="0" 
        reg.2.fwd.noanswer.status="0" 
        reg.2.lineKeys="1" 
        reg.2.musicOnHold.uri="" 
        reg.2.outboundProxy.failOver.failBack.timeout="3600" 
        reg.2.outboundProxy.failOver.failRegistrationOn="1" 
        reg.2.outboundProxy.failOver.reRegisterOn="0" 
        reg.2.outboundProxy.address="192.168.100.33"
        reg.2.outboundProxy.port="5060" 
        reg.2.protocol.SIP="1" 
        reg.2.ringType="ringer2" 
        reg.2.serverFeatureControl.cf="1" 
        reg.2.serverFeatureControl.dnd="1" 
        reg.2.thirdPartyName="" 
        reg.2.useCompleteUriForRetrieve="1" 
        reg.2.server.1.expires="3600" 
        reg.2.server.1.expires.lineSeize="30" 
        reg.2.server.1.expires.overlap="60" 
        reg.2.server.1.lcs="0" 
        reg.2.server.1.retryMaxCount="3" 
        reg.2.server.1.retryTimeOut="0" 
        reg.2.server.2.expires="3600" 
        reg.2.server.2.expires.lineSeize="30" 
        reg.2.server.2.expires.overlap="60" 
        reg.2.server.2.lcs="0" 
        reg.2.server.2.retryMaxCount="3" 
        reg.2.server.2.retryTimeOut="0">
  </reg>
 <se se.destination="chassis">
    <se.rt>
      <se.rt.modification se.rt.modification.enabled="1">
      </se.rt.modification>
    </se.rt>
  </se>
  <up up.25mm="0" up.accessibilityFeatures="0" up.analogHeadsetOption="0" up.audioMode="0" up.callTypeOrderVoice="0" up.handsfreeMode="1" up.headsetMode="0" up.localClockEnabled="1" up.manualProtocolRouting="1" up.mwiVisible="0" up.onHookDialingEnabled="1" up.useDirectoryNames="0" up.welcomeSoundEnabled="1" up.welcomeSoundOnWarmBootEnabled="0">
    <up.2P5mmJack up.2P5mmJack.mode="0">
    </up.2P5mmJack>
    <up.audioSetup up.audioSetup.auxInput="2" up.audioSetup.auxOutput="2">
    </up.audioSetup>
    <up.backlight up.backlight.idleIntensity="1" up.backlight.onIntensity="3">
    </up.backlight>
    <up.idleBrowser up.idleBrowser.enabled="0">
    </up.idleBrowser>
    <up.manualProtocolRouting up.manualProtocolRouting.softKeys="1">
    </up.manualProtocolRouting>
    <up.offHookAction up.offHookAction.none="0">
    </up.offHookAction>
    <up.pictureFrame up.pictureFrame.folder="" up.pictureFrame.timePerImage="5">
    </up.pictureFrame>
    <up.screenSaver up.screenSaver.enabled="0" up.screenSaver.waitTime="15">
    </up.screenSaver>
    <up.toneControl up.toneControl.bass="0" up.toneControl.treble="0">
    </up.toneControl>
  </up>
  <voIpProt>
    <voIpProt.SIP>
      <voIpProt.SIP.assuredService>
        <voIpProt.SIP.assuredService.namespace>
          <voIpProt.SIP.assuredService.namespace.label>
            <voIpProt.SIP.assuredService.namespace.label.UCRdrsn voIpProt.SIP.assuredService.namespace.label.UCRdrsn.1="" voIpProt.SIP.assuredService.namespace.label.UCRdrsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.UCRdrsn>
            <voIpProt.SIP.assuredService.namespace.label.UCRdsn voIpProt.SIP.assuredService.namespace.label.UCRdsn.1="" voIpProt.SIP.assuredService.namespace.label.UCRdsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.UCRdsn>
            <voIpProt.SIP.assuredService.namespace.label.custom voIpProt.SIP.assuredService.namespace.label.custom.1="" voIpProt.SIP.assuredService.namespace.label.custom.2="">
            </voIpProt.SIP.assuredService.namespace.label.custom>
            <voIpProt.SIP.assuredService.namespace.label.drsn voIpProt.SIP.assuredService.namespace.label.drsn.1="" voIpProt.SIP.assuredService.namespace.label.drsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.drsn>
            <voIpProt.SIP.assuredService.namespace.label.dsn voIpProt.SIP.assuredService.namespace.label.dsn.1="" voIpProt.SIP.assuredService.namespace.label.dsn.2="">
            </voIpProt.SIP.assuredService.namespace.label.dsn>
          </voIpProt.SIP.assuredService.namespace.label>
        </voIpProt.SIP.assuredService.namespace>
      </voIpProt.SIP.assuredService>
    </voIpProt.SIP>
  </voIpProt>
</polycomConfig>

----- Original Message -----
From: "Brian West" <brian at freeswitch.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, April 13, 2011 9:29:55 AM
Subject: Re: [Freeswitch-users] Shared Call appearence, barging and presence

you might wanna make sure you have this set to shared because if all the little phone icons aren't split in two then you'll never get this working... ONE line not knowing its shared will cause them all to not function properly.

/b

On Apr 12, 2011, at 9:59 PM, Gourav Vohra wrote:

>        reg.1.type="private" 





From msc at freeswitch.org  Wed Apr 13 19:01:23 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Apr 2011 08:01:23 -0700
Subject: [Freeswitch-users] Lua session originate - freeswitch crash
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DCA93100@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DCA92D3D@doctorbob.loop.local>
	<1302633709973-6266239.post@n2.nabble.com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92E03@doctorbob.loop.local>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92EA1@doctorbob.loop.local>
	<BANLkTi=f8cRhf1QL2tb4C28mv3E0MSEY4g@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA93058@doctorbob.loop.local>
	<0d4b01cbf9db$7fe1ce50$7fa56af0$@com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA93100@doctorbob.loop.local>
Message-ID: <BANLkTima48ChJXssqETETsL4OaBagOo5zw@mail.gmail.com>

On Wed, Apr 13, 2011 at 6:59 AM, Eric Beard <eric at loopfx.com> wrote:

> I've been fiddling around with things this morning and I am using a
> combination of PHP and Lua.
>
> I use php to simply originate the call (using regular sockets to talk to
> 8021, not the php mod for ESL, since I can't get it to compile).  In the
> originate command I specify a Lua script that handles the rest of the call.
>
> I'm using the PHP functions described here:
>
> http://wiki.freeswitch.org/wiki/PHP_Event_Socket
>
> My PHP:
>
>
>  $fp = event_socket_create("127.0.0.1", 8021, "ClueCon");
>  $channelVars =
> "{ignore_early_media=true,origination_caller_id_number=18778502010}";
>  $phoneNum = "17277762768";
>  $notificationId = "x";
>  $cmd = "bgapi originate $channelVars".
>   "sofia/gateway/affinity/".$phoneNum." &lua('test2.lua $notificationId')";
>  echo "Sending command: $cmd";
>  $response = event_socket_request($fp, $cmd);
>  echo $response;
>  fclose($fp);
>
> My Lua:
>
>
> prompt = "/home/eric/test1.wav"
> session:streamFile(prompt)
> session:hangup()
>
> This is working for the very basic scenario of just waiting for the number
> to pick up, playing a wav, and then hanging up.  If anybody can point out
> any major showstoppers that would prevent me from handling significant user
> interaction, I'd love to hear about them now.  One of the things I'd like to
> do is incorporate pocketsphinx so that I can do voice reco.  Is that
> possible from Lua?
>

You can run pretty much any dialplan application from Lua using the
session:execute method. I don't see any issues with you originating the
calls via event socket and then handling the answered calls with a Lua dp
script.

-MC
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From msc at freeswitch.org  Wed Apr 13 19:07:21 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 13 Apr 2011 08:07:21 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today - SIPVicious
	(friendly-scanner) Author
Message-ID: <BANLkTinrNwCCxFxCbJEi-McX1g2W7y39Pw@mail.gmail.com>

Hello!

Please join us for the conference call today as we welcome Sandro Gauci,
author of SIPVicious. You may know this better by the name "friend-scanner"
that the script kiddies are using. Hear from the SIPVicious author to find
out what lead to him creating this tool, some of the challenges he's faced,
and how he's dealt with nefarious types who've been misusing it.

Today's agenda is here:
http://wiki.freeswitch.org/wiki/FS_weekly_2011_04_13

Talk to you all soon!

-MC
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From anthony.minessale at gmail.com  Wed Apr 13 19:16:11 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Apr 2011 10:16:11 -0500
Subject: [Freeswitch-users] Event queue is full.
In-Reply-To: <fc6f970017d8e.4da5735d@insightbb.com>
References: <fc6f970017d8e.4da5735d@insightbb.com>
Message-ID: <BANLkTimEHUm7u21rVzx-7uLGHXhT1fhFSQ@mail.gmail.com>

you are doing something abusive with sipp
we do not entertain and assist with Ddos or load testing related subjects.


On Wed, Apr 13, 2011 at 8:56 AM, Gary Chen <gchen00 at insightbb.com> wrote:
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
> West
> Sent: Wednesday, April 13, 2011 9:34 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Event queue is full.
>
>
>
> Sounds like an old code base.. you sure that other box is using the latest
> code?
>
>
>
> /b
>
>
>
> On Apr 13, 2011, at 7:44 AM, Gary Chen wrote:
>
>
>
>> FreeSWITCH Version 1.0.head (git-5310735 2011-04-07 15-47-30 -0500)
>
>>
>
>> Last night I start Sipp to sent calls to my test? FS1 and FS1 in turn
>> forward calls to another FS2 playing Music On Hold. I got about 300
>> concurrent calls with media going through FS1 before I left.
>
>> This morning, the FS1 is not responding. I fs_cli into it and got a lot of
>> messages like this:
>
>> 2011-04-13 08:12:11.766570 [CRIT] switch_event.c:1367 Event queue is full!
>
>> x2011-04-13 08:12:11.798571 [CRIT] switch_event.c:1367 Event queue is
>> full!
>
>> 2011-04-13 08:12:11.809567 [CRIT] switch_event.c:1367 Event queue is full!
>
>> 2011-04-13 08:12:11.809567 [CRIT] switch_event.c:1367 Event queue is full!
>
>> 2011-04-13 08:12:11.816565 [CRIT] switch_event.c:1367 Event queue is full!
>
>> 2011-04-13 08:12:11.827574 [CRIT] switch_event.c:1367 Event queue is full!
>
>
>
> ?Call Flow: Sipp ---> FS1 --->FS2(play music on hold)
>
> FS1 version: FreeSWITCH Version 1.0.head (git-5310735 2011-04-07 15-47-30
> -0500)
>
> FS2 Version: FreeSWITCH Version 1.0.7 (hacked-20110119T213949Z)
>
> FS2 has older version. But 'Event queue is full' only happen on FS1. Should
> have nothing to do with FS2. Right now FS2 still show 304 sessions. That
> means that it still playing music on 304 calls.
>
> I can not run fs_cli -x status because the event queue problem.
>
> Gary
>
> _______________________________________________
>
> FreeSWITCH-users mailing list
>
> FreeSWITCH-users at lists.freeswitch.org
>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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 13 19:17:44 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Apr 2011 10:17:44 -0500
Subject: [Freeswitch-users] FS does not repose to SIP OK message.
In-Reply-To: <22576037-ECFC-4E4C-B688-B06A30A56C25@freeswitch.org>
References: <fc7f86bf12222.4da55fcb@insightbb.com>
	<22576037-ECFC-4E4C-B688-B06A30A56C25@freeswitch.org>
Message-ID: <BANLkTimXwTPcL9=x6n3JnN361=s_CiSFNA@mail.gmail.com>

and put it on paste bin.
also maybe look for the problem while you're at it. its for sure some
kind of nat issue.


On Wed, Apr 13, 2011 at 8:31 AM, Brian West <brian at freeswitch.org> wrote:
> Please redo the console log like your original one with 'sofia global siptrace on' and 'sofia loglevel all 9' .. this trace below is just chicken scratch and hard to follow.
>
> /b
>
> On Apr 13, 2011, at 7:33 AM, Gary Chen wrote:
>
>>
>>
>> The FS1 should response back with ACK. It did not.
>> The strange thing is that It works sometime.Here is the SIP trace from ngrep for both good and bad calls on FS1 server:SJPhone IP: 226.59.139.61FS1 IP:226.59.129.223FS2 IP:226.59.129.221You can see that FS2 sent several 200 OK and FS1 never reply back.
>> Good sip call:U 226.59.139.61:5060 -> 226.59.129.223:5080
>> ? INVITE sip:5025155596 at fs2000.lightyear.net SIP/2.0..Via: SIP/2.0/UDP 226.59.139.61;branch=z9hG4bKd8318b3d000007494da459a
>> ? 900006fac00000273;rport..From: "unknown" <sip:1009 at fs2000.lightyear.net>;tag=a344c583a24..To: <sip:5025155596 at fs2000.lig
>> ? htyear.net>..Contact: <sip:1009 at 226.59.139.61>..Call-ID: 56BF61B3189F4EDEB8C9067D5823C8480xd8318b3d..CSeq: 1 INVITE..Max
>> ? -Forwards: 70..User-Agent: SJphone/1.65.377a (SJ Labs)..Content-Length: 368..Content-Type: application/sdp..Supported: r
>> ? eplaces,norefersub,timer....v=0..o=- 3511605289 3511605289 IN IP4 226.59.139.61..s=SJphone..c=IN IP4 226.59.139.61..t=0
>> ? 0..m=audio 49352 RTP/AVP 3 97 98 8 0 101..c=IN IP4 226.59.139.61..a=rtpmap:3 GSM/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:9
>> ? 8 iLBC/8000..a=fmtp:98 mode=20..a=rtpmap:8 PCMA/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:10
>> ? 1 0-16..a=setup:active..a=sendrecv..
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

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  Wed Apr 13 19:22:51 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 13 Apr 2011 16:22:51 +0100
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <BANLkTikyVkH_FyLsdhnkUBPOB-bZwjGcXQ@mail.gmail.com>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A28C423231C@exmachina.office.kapper.net>
	<BANLkTimrvA=Lgva-FAdXxWRhS-_UDuTgTA@mail.gmail.com>
	<BANLkTimFeP+r2uNk09BbVw7hAULf1MYvzA@mail.gmail.com>
	<BANLkTikyVkH_FyLsdhnkUBPOB-bZwjGcXQ@mail.gmail.com>
Message-ID: <BANLkTi=LQFHjQge0QzFtzCXKk4OMQ618yA@mail.gmail.com>

Have you tried enabling late negotiation on the sip profile?

Normally FS will pick a codec from the list before hitting dialplan, that
parameter delays the decision until the bridge and should send the full list
over.

-Steve


On 13 April 2011 08:53, Sam <u2nsam at gmail.com> wrote:

>
> I want to pass the exact information what leg A is sending for Media
> Description & Media Attributes and not connection information.
>
>
> Regards
> Sam
>
>
>
>
> On Tue, Apr 12, 2011 at 10:16 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> 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
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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  Wed Apr 13 19:32:38 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Apr 2011 10:32:38 -0500
Subject: [Freeswitch-users] Lua session originate - freeswitch crash
In-Reply-To: <BANLkTima48ChJXssqETETsL4OaBagOo5zw@mail.gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DCA92D3D@doctorbob.loop.local>
	<1302633709973-6266239.post@n2.nabble.com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92E03@doctorbob.loop.local>
	<D4A666272A214C4DBBABFE569CA34CA401DCA92EA1@doctorbob.loop.local>
	<BANLkTi=f8cRhf1QL2tb4C28mv3E0MSEY4g@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA93058@doctorbob.loop.local>
	<0d4b01cbf9db$7fe1ce50$7fa56af0$@com>
	<D4A666272A214C4DBBABFE569CA34CA401DCA93100@doctorbob.loop.local>
	<BANLkTima48ChJXssqETETsL4OaBagOo5zw@mail.gmail.com>
Message-ID: <BANLkTinU+-Y_rAqMhTK5j2zpTR7OUiqXFw@mail.gmail.com>

I didn't say you can't use LUA The only point I was trying to make is:


if you are running the lua script on an inbound call, then you just
create a new session and ignore the one the call came in on you have
an issue.

lua is best when you are trying to make an IVR or do advanced logic
that is too much for the XML dialplan.
You have a single channel and you use the channel data to do things on
that channel.

One big mistake people tend to make is they think that embedded
scripting is a shortcut to making complex apps without thinking
through the requirements.

Also doing things like trying to setup and bridge 2 channels together
in lua manually is unwise.  The originate syntax is the same
everywhere including with the bridge app.  The best thing to do is use
your script to figure out the params you need for the originate syntax
of the B leg then simply set that as a variable, exit the script and
pass that variable to the bridge app from your dialplan.

If you need to be fully asynchronous you for sure want to consider ESL.





On Wed, Apr 13, 2011 at 10:01 AM, Michael Collins <msc at freeswitch.org> wrote:
>
>
> On Wed, Apr 13, 2011 at 6:59 AM, Eric Beard <eric at loopfx.com> wrote:
>>
>> I've been fiddling around with things this morning and I am using a
>> combination of PHP and Lua.
>>
>> I use php to simply originate the call (using regular sockets to talk to
>> 8021, not the php mod for ESL, since I can't get it to compile). ?In the
>> originate command I specify a Lua script that handles the rest of the call.
>>
>> I'm using the PHP functions described here:
>>
>> http://wiki.freeswitch.org/wiki/PHP_Event_Socket
>>
>> My PHP:
>>
>>
>> ?$fp = event_socket_create("127.0.0.1", 8021, "ClueCon");
>> ?$channelVars =
>> "{ignore_early_media=true,origination_caller_id_number=18778502010}";
>> ?$phoneNum = "17277762768";
>> ?$notificationId = "x";
>> ?$cmd = "bgapi originate $channelVars".
>> ? "sofia/gateway/affinity/".$phoneNum." &lua('test2.lua
>> $notificationId')";
>> ?echo "Sending command: $cmd";
>> ?$response = event_socket_request($fp, $cmd);
>> ?echo $response;
>> ?fclose($fp);
>>
>> My Lua:
>>
>>
>> prompt = "/home/eric/test1.wav"
>> session:streamFile(prompt)
>> session:hangup()
>>
>> This is working for the very basic scenario of just waiting for the number
>> to pick up, playing a wav, and then hanging up. ?If anybody can point out
>> any major showstoppers that would prevent me from handling significant user
>> interaction, I'd love to hear about them now. ?One of the things I'd like to
>> do is incorporate pocketsphinx so that I can do voice reco. ?Is that
>> possible from Lua?
>
> You can run pretty much any dialplan application from Lua using the
> session:execute method. I don't see any issues with you originating the
> calls via event socket and then handling the answered calls with a Lua dp
> script.
> -MC
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

AIM: anthm
MSN:anthony_minessale at hotmail.com
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  Wed Apr 13 19:34:38 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 13 Apr 2011 17:34:38 +0200
Subject: [Freeswitch-users] Hangup a call with silence
In-Reply-To: <4D8D1911.9030303@rosengart.de>
References: <AANLkTi=T9hKyzb-KrdaKQZCg6YWet_TMLNc-NEvh3MY7@mail.gmail.com>
	<4D8D1911.9030303@rosengart.de>
Message-ID: <BANLkTikG_vsgTF8vfsxzUqDawWJ-+=qCOQ@mail.gmail.com>

The "rtp-timeout-sec" works fine when there is no RTP at all.
I want to hangup a call when there are RTP packets but with silence (energy
based).
Any way to do so?

Stephen

On Fri, Mar 25, 2011 at 11:37 PM, Frank Rosengart <frank at rosengart.de>wrote:

> On 03/25/2011 11:14 PM, Stephen Wilde wrote:
>
> > There is a way to auto disconnect the call where there is, for example,
> > 30s of silence?
>
> What about <param name="rtp-timeout-sec" value="30"/> in your sofia
> profile?
>
> Frank
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 acrow at integrafin.co.uk  Wed Apr 13 20:36:45 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Wed, 13 Apr 2011 17:36:45 +0100
Subject: [Freeswitch-users] Event queue is full.
In-Reply-To: <BANLkTimEHUm7u21rVzx-7uLGHXhT1fhFSQ@mail.gmail.com>
References: <fc6f970017d8e.4da5735d@insightbb.com>
	<BANLkTimEHUm7u21rVzx-7uLGHXhT1fhFSQ@mail.gmail.com>
Message-ID: <4DA5D11D.6030501@integrafin.co.uk>

Anthony,

Should load testing questions be directed off-list? Or perhaps by 
consultancy? It seems that there are certainly some valid cases for 
doing it, eg to ensure that your site is not vulnerable to a dDOS and is 
resourced correctly.

I can fully understand not wanting to discuss too deeply or go into 
numbers on a public searchable list with identifiable posters, but I 
worry if the subject is considered completely taboo.

Cheers

Alex



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From gchen00 at insightbb.com  Wed Apr 13 20:39:12 2011
From: gchen00 at insightbb.com (Gary Chen)
Date: Wed, 13 Apr 2011 12:39:12 -0400
Subject: [Freeswitch-users] Event queue is full.
Message-ID: <fc83cc43140f0.4da59970@insightbb.com>



?I am not doing load test and my sipp is fairly
simple. It just send call and wail for short time period and then send
bye to end the call. The system I am using has plenty of power. Last
night it only used about 15% of cpu for 300 concurrent calls.


We are in the process of exploring options to
replace our aging commercial sip feature server. I just want to make
sure FS does not have any memory leak issue and can run long enough
without rebooting.





-----Original Message-----

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
Minessale

Sent: Wednesday, April 13, 2011 11:16 AM

To: FreeSWITCH Users Help

Subject: Re: [Freeswitch-users] Event queue is full.

?

you are doing something abusive with sipp

we do not entertain and assist with Ddos or load testing
related subjects.

?

?

On Wed, Apr 13, 2011 at 8:56 AM, Gary Chen
<gchen00 at insightbb.com> wrote:

> -----Original Message-----

> From: freeswitch-users-bounces at lists.freeswitch.org

>
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian

> West

> Sent: Wednesday, April 13, 2011 9:34 AM

> To: FreeSWITCH Users Help

> Subject: Re: [Freeswitch-users] Event queue is full.

>?

>?

>?

> Sounds like an old code base.. you sure that other
box is using the latest

> code?

>?

>?

>?

> /b

>?

>?

>?

> On Apr 13, 2011, at 7:44 AM, Gary Chen wrote:

>?

>?

>?

>> FreeSWITCH Version 1.0.head (git-5310735
2011-04-07 15-47-30 -0500)

>?

>>?

>?

>> Last night I start Sipp to sent calls to my
test? FS1 and FS1 in turn

>> forward calls to another FS2 playing Music On Hold.
I got about 300

>> concurrent calls with media going through FS1
before I left.

>?

>> This morning, the FS1 is not responding. I
fs_cli into it and got a lot of

>> messages like this:

>?

>> 2011-04-13 08:12:11.766570 [CRIT]
switch_event.c:1367 Event queue is full!

>?

>> x2011-04-13 08:12:11.798571 [CRIT]
switch_event.c:1367 Event queue is

>> full!

>?

>> 2011-04-13 08:12:11.809567 [CRIT]
switch_event.c:1367 Event queue is full!

>?

>> 2011-04-13 08:12:11.809567 [CRIT]
switch_event.c:1367 Event queue is full!

>?

>> 2011-04-13 08:12:11.816565 [CRIT]
switch_event.c:1367 Event queue is full!

>?

>> 2011-04-13 08:12:11.827574 [CRIT]
switch_event.c:1367 Event queue is full!

>?

>?

>?

> ?Call Flow: Sipp ---> FS1 --->FS2(play
music on hold)

>?

> FS1 version: FreeSWITCH Version 1.0.head
(git-5310735 2011-04-07 15-47-30

> -0500)

>?

> FS2 Version: FreeSWITCH Version 1.0.7
(hacked-20110119T213949Z)

>?

> FS2 has older version. But 'Event queue is full'
only happen on FS1. Should

> have nothing to do with FS2. Right now FS2 still
show 304 sessions. That

> means that it still playing music on 304 calls.

>?

> I can not run fs_cli -x status because the event
queue problem.

>?

> Gary

>?

> _______________________________________________

>?

> FreeSWITCH-users mailing list

>?

> FreeSWITCH-users at lists.freeswitch.org

>?

>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

>?

>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

>?

> http://www.freeswitch.org

>?

> _______________________________________________

> FreeSWITCH-users mailing list

> FreeSWITCH-users at lists.freeswitch.org

>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users

> http://www.freeswitch.org

>?

>?

?

?

?

-- 

Anthony Minessale II

?

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

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

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From acrow at integrafin.co.uk  Wed Apr 13 20:47:51 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Wed, 13 Apr 2011 17:47:51 +0100
Subject: [Freeswitch-users] called number rewrite
In-Reply-To: <BANLkTikEMcBF-vK4QhTrb6ABw7fkDe=3Zw@mail.gmail.com>
References: <BANLkTikEMcBF-vK4QhTrb6ABw7fkDe=3Zw@mail.gmail.com>
Message-ID: <4DA5D3B7.1010903@integrafin.co.uk>

On 13/04/11 00:23, budi wibowo wrote:
> hi
> is it possible for freeswitch to change / rewrite called number.
> say, i call 1234567 and FS will change the called number to 44444444
>
>
> thx
>
> budi
>
>

You can also do this with regex, so eg match incoming digits and take, 
say, the last two and put something else in front.

eg

<condition field="destination_number" expression="60([1-5])">
<action application="bridge" data="sofia/external/76$1 at host"/>

or indeed

<action application="transfer" data"76$1 XML default/>
</condition>

For an incoming call, 1st to send to another system, second to send into 
the default dialplan. The variable $1 comes from what is matched inside 
the () brackets. Here any number 601-5 would call 761-5.

Cheers

Alex

-- 
This message is intended only for the addressee and may contain
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From anthony.minessale at gmail.com  Wed Apr 13 20:50:29 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 13 Apr 2011 11:50:29 -0500
Subject: [Freeswitch-users] Event queue is full.
In-Reply-To: <fc83cc43140f0.4da59970@insightbb.com>
References: <fc83cc43140f0.4da59970@insightbb.com>
Message-ID: <BANLkTikfMX3tcYYYxdktXVgZ1mdjOxQGPg@mail.gmail.com>

Every time we get someone saying "I'm just using sipp to make sure FS
doesn't suck first but there is a problem." and I mean every time so
far 100%, it was misuse or mis-configuration related to probing or
testing with sipp incorrectly.  Then we spend many hours of free
consulting getting it working.

We have several telcos running endless traffic in production on very
recent GIT who complain when they can't have thousands of calls
processed.

The only hint I can offer is to use our UA scenario file that properly
handles the various call outcomes and uses the correct dialog id

http://www.freeswitch.org/eg/load_test/dft_cap.xml

If you are getting something saying your event queue is full, it would
suggest you are somehow blocking the event subsystem by doing
something wrong.   Again we do not offer free support for this type of
topic.





On Wed, Apr 13, 2011 at 11:39 AM, Gary Chen <gchen00 at insightbb.com> wrote:
> ?I am not doing load test and my sipp is fairly simple. It just send call
> and wail for short time period and then send bye to end the call. The system
> I am using has plenty of power. Last night it only used about 15% of cpu for
> 300 concurrent calls.
>
> We are in the process of exploring options to replace our aging commercial
> sip feature server. I just want to make sure FS does not have any memory
> leak issue and can run long enough without rebooting.
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
> Minessale
> Sent: Wednesday, April 13, 2011 11:16 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Event queue is full.
>
>
>
> you are doing something abusive with sipp
>
> we do not entertain and assist with Ddos or load testing related subjects.
>
>
>
>
>
> On Wed, Apr 13, 2011 at 8:56 AM, Gary Chen <gchen00 at insightbb.com> wrote:
>
>> -----Original Message-----
>
>> From: freeswitch-users-bounces at lists.freeswitch.org
>
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
>
>> West
>
>> Sent: Wednesday, April 13, 2011 9:34 AM
>
>> To: FreeSWITCH Users Help
>
>> Subject: Re: [Freeswitch-users] Event queue is full.
>
>>
>
>>
>
>>
>
>> Sounds like an old code base.. you sure that other box is using the latest
>
>> code?
>
>>
>
>>
>
>>
>
>> /b
>
>>
>
>>
>
>>
>
>> On Apr 13, 2011, at 7:44 AM, Gary Chen wrote:
>
>>
>
>>
>
>>
>
>>> FreeSWITCH Version 1.0.head (git-5310735 2011-04-07 15-47-30 -0500)
>
>>
>
>>>
>
>>
>
>>> Last night I start Sipp to sent calls to my test? FS1 and FS1 in turn
>
>>> forward calls to another FS2 playing Music On Hold. I got about 300
>
>>> concurrent calls with media going through FS1 before I left.
>
>>
>
>>> This morning, the FS1 is not responding. I fs_cli into it and got a lot
>>> of
>
>>> messages like this:
>
>>
>
>>> 2011-04-13 08:12:11.766570 [CRIT] switch_event.c:1367 Event queue is
>>> full!
>
>>
>
>>> x2011-04-13 08:12:11.798571 [CRIT] switch_event.c:1367 Event queue is
>
>>> full!
>
>>
>
>>> 2011-04-13 08:12:11.809567 [CRIT] switch_event.c:1367 Event queue is
>>> full!
>
>>
>
>>> 2011-04-13 08:12:11.809567 [CRIT] switch_event.c:1367 Event queue is
>>> full!
>
>>
>
>>> 2011-04-13 08:12:11.816565 [CRIT] switch_event.c:1367 Event queue is
>>> full!
>
>>
>
>>> 2011-04-13 08:12:11.827574 [CRIT] switch_event.c:1367 Event queue is
>>> full!
>
>>
>
>>
>
>>
>
>> ?Call Flow: Sipp ---> FS1 --->FS2(play music on hold)
>
>>
>
>> FS1 version: FreeSWITCH Version 1.0.head (git-5310735 2011-04-07 15-47-30
>
>> -0500)
>
>>
>
>> FS2 Version: FreeSWITCH Version 1.0.7 (hacked-20110119T213949Z)
>
>>
>
>> FS2 has older version. But 'Event queue is full' only happen on FS1.
>> Should
>
>> have nothing to do with FS2. Right now FS2 still show 304 sessions. That
>
>> means that it still playing music on 304 calls.
>
>>
>
>> I can not run fs_cli -x status because the event queue problem.
>
>>
>
>> Gary
>
>>
>
>> _______________________________________________
>
>>
>
>> FreeSWITCH-users mailing list
>
>>
>
>> FreeSWITCH-users at lists.freeswitch.org
>
>>
>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>>
>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>>
>
>> http://www.freeswitch.org
>
>>
>
>> _______________________________________________
>
>> FreeSWITCH-users mailing list
>
>> FreeSWITCH-users at lists.freeswitch.org
>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
>> http://www.freeswitch.org
>
>>
>
>>
>
>
>
>
>
>
>
> --
>
> Anthony Minessale II
>
>
>
> 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 sameer2k3t at gmail.com  Wed Apr 13 21:36:56 2011
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Wed, 13 Apr 2011 22:36:56 +0500
Subject: [Freeswitch-users] codec negotiation
In-Reply-To: <BANLkTimuYKR2=2KETv6W-Lr7t75Dd5kOWg@mail.gmail.com>
References: <BANLkTi=mD2mv726VZrmMO+ud2_tMgjf4EA@mail.gmail.com>
	<BANLkTini2w_5uc-LZRJxy4VwdcUKcthk7w@mail.gmail.com>
	<BANLkTimuYKR2=2KETv6W-Lr7t75Dd5kOWg@mail.gmail.com>
Message-ID: <BANLkTikNywK3qO+VG3=6aiq06Ru132qb=g@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
here is complete trace
http://pastebin.freeswitch.org/16053


On Fri, Apr 8, 2011 at 5:17 PM, Sameer Khan <sameer2k3t at gmail.com> wrote:

> 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 acrow at integrafin.co.uk  Wed Apr 13 21:44:44 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Wed, 13 Apr 2011 18:44:44 +0100
Subject: [Freeswitch-users] Event queue is full.
In-Reply-To: <BANLkTikfMX3tcYYYxdktXVgZ1mdjOxQGPg@mail.gmail.com>
References: <fc83cc43140f0.4da59970@insightbb.com>
	<BANLkTikfMX3tcYYYxdktXVgZ1mdjOxQGPg@mail.gmail.com>
Message-ID: <4DA5E10C.30403@integrafin.co.uk>

On 13/04/11 17:50, Anthony Minessale wrote:
> Every time we get someone saying "I'm just using sipp to make sure FS
> doesn't suck first but there is a problem." and I mean every time so
> far 100%, it was misuse or mis-configuration related to probing or
> testing with sipp incorrectly.  Then we spend many hours of free
> consulting getting it working.
>
> We have several telcos running endless traffic in production on very
> recent GIT who complain when they can't have thousands of calls
> processed.
>
> The only hint I can offer is to use our UA scenario file that properly
> handles the various call outcomes and uses the correct dialog id
>
> http://www.freeswitch.org/eg/load_test/dft_cap.xml
>
> If you are getting something saying your event queue is full, it would
> suggest you are somehow blocking the event subsystem by doing
> something wrong.   Again we do not offer free support for this type of
> topic.
Anthony,

Thanks for the explanation and the link, glad it has been covered and 
thought has gone into it.

Cheers

Alex




From wstephen80 at gmail.com  Wed Apr 13 21:52:40 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 13 Apr 2011 19:52:40 +0200
Subject: [Freeswitch-users] Event queue is full.
In-Reply-To: <fc83cc43140f0.4da59970@insightbb.com>
References: <fc83cc43140f0.4da59970@insightbb.com>
Message-ID: <BANLkTik0SCDejdsj7rRNBEvW4Qv21yiW1w@mail.gmail.com>

I think the problem you see can be related to the test configuration, in my
experience, Freeswitch has not problem to handle 300 contemporary calls.

This is, for example my running Freeswitch (in production environment real
traffic) status from fs_cli:

status
UP 0 years, 5 days, 8 hours, 29 minutes, 6 seconds, 340 milliseconds, 102
microseconds
7253838 session(s) since startup
1789 session(s) 20/200
3000 session(s) max
min idle cpu 0.00/43.00

As you see there are 1789 sessions and from latest restart (for maintenance)
has handled more then 1.400.000 sessions per day.

Stephen


On Wed, Apr 13, 2011 at 6:39 PM, Gary Chen <gchen00 at insightbb.com> wrote:

>  I am not doing load test and my sipp is fairly simple. It just send call
> and wail for short time period and then send bye to end the call. The system
> I am using has plenty of power. Last night it only used about 15% of cpu for
> 300 concurrent calls.
>
> We are in the process of exploring options to replace our aging commercial
> sip feature server. I just want to make sure FS does not have any memory
> leak issue and can run long enough without rebooting.
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony
> Minessale
> Sent: Wednesday, April 13, 2011 11:16 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Event queue is full.
>
>
>
> you are doing something abusive with sipp
>
> we do not entertain and assist with Ddos or load testing related subjects.
>
>
>
>
>
> On Wed, Apr 13, 2011 at 8:56 AM, Gary Chen <gchen00 at insightbb.com> wrote:
>
> > -----Original Message-----
>
> > From: freeswitch-users-bounces at lists.freeswitch.org
>
> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Brian
>
> > West
>
> > Sent: Wednesday, April 13, 2011 9:34 AM
>
> > To: FreeSWITCH Users Help
>
> > Subject: Re: [Freeswitch-users] Event queue is full.
>
> >
>
> >
>
> >
>
> > Sounds like an old code base.. you sure that other box is using the
> latest
>
> > code?
>
> >
>
> >
>
> >
>
> > /b
>
> >
>
> >
>
> >
>
> > On Apr 13, 2011, at 7:44 AM, Gary Chen wrote:
>
> >
>
> >
>
> >
>
> >> FreeSWITCH Version 1.0.head (git-5310735 2011-04-07 15-47-30 -0500)
>
> >
>
> >>
>
> >
>
> >> Last night I start Sipp to sent calls to my test  FS1 and FS1 in turn
>
> >> forward calls to another FS2 playing Music On Hold. I got about 300
>
> >> concurrent calls with media going through FS1 before I left.
>
> >
>
> >> This morning, the FS1 is not responding. I fs_cli into it and got a lot
> of
>
> >> messages like this:
>
> >
>
> >> 2011-04-13 08:12:11.766570 [CRIT] switch_event.c:1367 Event queue is
> full!
>
> >
>
> >> x2011-04-13 08:12:11.798571 [CRIT] switch_event.c:1367 Event queue is
>
> >> full!
>
> >
>
> >> 2011-04-13 08:12:11.809567 [CRIT] switch_event.c:1367 Event queue is
> full!
>
> >
>
> >> 2011-04-13 08:12:11.809567 [CRIT] switch_event.c:1367 Event queue is
> full!
>
> >
>
> >> 2011-04-13 08:12:11.816565 [CRIT] switch_event.c:1367 Event queue is
> full!
>
> >
>
> >> 2011-04-13 08:12:11.827574 [CRIT] switch_event.c:1367 Event queue is
> full!
>
> >
>
> >
>
> >
>
> >  Call Flow: Sipp ---> FS1 --->FS2(play music on hold)
>
> >
>
> > FS1 version: FreeSWITCH Version 1.0.head (git-5310735 2011-04-07 15-47-30
>
> > -0500)
>
> >
>
> > FS2 Version: FreeSWITCH Version 1.0.7 (hacked-20110119T213949Z)
>
> >
>
> > FS2 has older version. But 'Event queue is full' only happen on FS1.
> Should
>
> > have nothing to do with FS2. Right now FS2 still show 304 sessions. That
>
> > means that it still playing music on 304 calls.
>
> >
>
> > I can not run fs_cli -x status because the event queue problem.
>
> >
>
> > Gary
>
> >
>
> > _______________________________________________
>
> >
>
> > FreeSWITCH-users mailing list
>
> >
>
> > FreeSWITCH-users at lists.freeswitch.org
>
> >
>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> >
>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> >
>
> > http://www.freeswitch.org
>
> >
>
> > _______________________________________________
>
> > FreeSWITCH-users mailing list
>
> > FreeSWITCH-users at lists.freeswitch.org
>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>
> > http://www.freeswitch.org
>
> >
>
> >
>
>
>
>
>
>
>
> --
>
> Anthony Minessale II
>
>
>
> 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
>
>
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From kris at kriskinc.com  Wed Apr 13 22:10:04 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 13 Apr 2011 14:10:04 -0400
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <1BDDB8AD-4CBD-4515-A7AD-693A5E875523@freeswitch.org>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A28C423231C@exmachina.office.kapper.net>
	<BANLkTimrvA=Lgva-FAdXxWRhS-_UDuTgTA@mail.gmail.com>
	<1BDDB8AD-4CBD-4515-A7AD-693A5E875523@freeswitch.org>
Message-ID: <BANLkTi=YM4x3O6Xc-JDtXVKymKdVbcmP4Q@mail.gmail.com>

Brian,

  For all of the confusion proxy media creates I still see cases where
it is useful...  It shouldn't be removed completely.

On Tue, Apr 12, 2011 at 8:55 PM, Brian West <brian at freeswitch.org> wrote:
> We aren't a proxy... we have transcended into this quasi proxy in some scenarios which mostly involve t.38... as for proxy media DO NOT USE IT. ?Just saying it might go away since the purpose of it is now not needed since we have full t.38.
>
> Thanks,
> Brian
>

-- 
Kristian Kielhofner


From me at nevian.org  Wed Apr 13 22:21:34 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Wed, 13 Apr 2011 22:21:34 +0400
Subject: [Freeswitch-users] codec negotiation
In-Reply-To: <BANLkTikNywK3qO+VG3=6aiq06Ru132qb=g@mail.gmail.com>
References: <BANLkTi=mD2mv726VZrmMO+ud2_tMgjf4EA@mail.gmail.com>
	<BANLkTini2w_5uc-LZRJxy4VwdcUKcthk7w@mail.gmail.com>
	<BANLkTimuYKR2=2KETv6W-Lr7t75Dd5kOWg@mail.gmail.com>
	<BANLkTikNywK3qO+VG3=6aiq06Ru132qb=g@mail.gmail.com>
Message-ID: <733771302718894@web8.yandex.ru>

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From sameer2k3t at gmail.com  Wed Apr 13 22:51:32 2011
From: sameer2k3t at gmail.com (Sameer Khan)
Date: Wed, 13 Apr 2011 23:51:32 +0500
Subject: [Freeswitch-users] codec negotiation
In-Reply-To: <733771302718894@web8.yandex.ru>
References: <BANLkTi=mD2mv726VZrmMO+ud2_tMgjf4EA@mail.gmail.com>
	<BANLkTini2w_5uc-LZRJxy4VwdcUKcthk7w@mail.gmail.com>
	<BANLkTimuYKR2=2KETv6W-Lr7t75Dd5kOWg@mail.gmail.com>
	<BANLkTikNywK3qO+VG3=6aiq06Ru132qb=g@mail.gmail.com>
	<733771302718894@web8.yandex.ru>
Message-ID: <BANLkTi=AL-gYSohQ6QWdsWaWvEEw5MNHkw@mail.gmail.com>

sorry i pasted wrong trace in pastebin


I tried it in this way as well <action application="export"
data="nolocal:absolute_codec_
string=PCMA,PCMU"/>';

but results were same

On Wed, Apr 13, 2011 at 11:21 PM, Serge S. Yuriev <me at nevian.org> wrote:

> Hi
>
> Maybe I'm wrong..
> As we can see from logs your abs_codec set incorrectly and not looks like
> shown in code bellow
>
> bridge({absolute_codec_string='.pcma;pcmu;.'}sofia/external/1212
> @outgoingip)
>
> Why there prefix and trailing dots?  Separator also expected as comma and
> perhaps you should write these uppercase
>
> 13.04.2011, 21:36, "Sameer Khan" <sameer2k3t at 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
> here is complete trace
> http://pastebin.freeswitch.org/16053
>
>
> On Fri, Apr 8, 2011 at 5:17 PM, Sameer Khan <sameer2k3t at gmail.com> wrote:
>
>> 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
>>>
>>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 14 00:21:20 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 13 Apr 2011 21:21:20 +0100
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <1BDDB8AD-4CBD-4515-A7AD-693A5E875523@freeswitch.org>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A28C423231C@exmachina.office.kapper.net>
	<BANLkTimrvA=Lgva-FAdXxWRhS-_UDuTgTA@mail.gmail.com>
	<1BDDB8AD-4CBD-4515-A7AD-693A5E875523@freeswitch.org>
Message-ID: <BANLkTimzWMB0x999AnMbDBaK_EJRToMVgw@mail.gmail.com>

"now not needed since we have full t.38"

What about codecs not implemented in FS? Not that there are many of those
left. :) It's the *only* good reason I can see for using it.

-Steve




On 13 April 2011 01:55, Brian West <brian at freeswitch.org> wrote:

> We aren't a proxy... we have transcended into this quasi proxy in some
> scenarios which mostly involve t.38... as for proxy media DO NOT USE IT.
>  Just saying it might go away since the purpose of it is now not needed
> since we have full t.38.
>
> Thanks,
> Brian
>
> On Apr 12, 2011, at 10:04 AM, Sam wrote:
>
> > I have done that, but i want to pass the exact SDP what i get from leg A
> to
> > leg B
> >
> > 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 elijah at crankenstein.com  Thu Apr 14 00:39:14 2011
From: elijah at crankenstein.com (elijah)
Date: Wed, 13 Apr 2011 13:39:14 -0700
Subject: [Freeswitch-users] occasional ~5s delay during bind_meta_app
	execute_extenstion
In-Reply-To: <BANLkTind1exnpMFgT-iS7sdbJqSi=QksOQ@mail.gmail.com>
References: <BANLkTikxT9VuejCLOUsbKCncLXv6HiEWiw@mail.gmail.com>
	<BANLkTin+HLFD6dnc=zWs_P-gftsgmpP+vw@mail.gmail.com>
	<BANLkTinnpJrLAg+XWs1izQ0MKE_ym+PVcQ@mail.gmail.com>
	<BANLkTind1exnpMFgT-iS7sdbJqSi=QksOQ@mail.gmail.com>
Message-ID: <BANLkTimJxkWnnsuJ1ZiqJ1ArcqQ=WMEEwg@mail.gmail.com>

Ok. What's 'pb'?

On Tue, Apr 12, 2011 at 9:09 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Apr 12, 2011 at 5:57 PM, elijah <elijah at crankenstein.com> wrote:
>
>> I'm afraid that cannot be the cause. I'm experiencing the delay before the
>> read is executed, immediately after a user presses *1 but before the
>> extension named 'dx' executes anything.
>
>
> Put these extensions into pb with a call log of this issue occurring. Use
> console loglevel 7.
> -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 paul at cupis.co.uk  Thu Apr 14 01:35:41 2011
From: paul at cupis.co.uk (Paul Cupis)
Date: Wed, 13 Apr 2011 22:35:41 +0100
Subject: [Freeswitch-users] occasional ~5s delay during bind_meta_app
 execute_extenstion
In-Reply-To: <BANLkTimJxkWnnsuJ1ZiqJ1ArcqQ=WMEEwg@mail.gmail.com>
References: <BANLkTikxT9VuejCLOUsbKCncLXv6HiEWiw@mail.gmail.com>	<BANLkTin+HLFD6dnc=zWs_P-gftsgmpP+vw@mail.gmail.com>	<BANLkTinnpJrLAg+XWs1izQ0MKE_ym+PVcQ@mail.gmail.com>	<BANLkTind1exnpMFgT-iS7sdbJqSi=QksOQ@mail.gmail.com>
	<BANLkTimJxkWnnsuJ1ZiqJ1ArcqQ=WMEEwg@mail.gmail.com>
Message-ID: <4DA6172D.8050500@cupis.co.uk>

On 13/04/11 21:39, elijah wrote:
> Ok. What's 'pb'?

http://pastebin.freeswitch.org

Regards,


From infos at madovsky.org  Thu Apr 14 02:17:57 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 13 Apr 2011 18:17:57 -0400
Subject: [Freeswitch-users] proxy SDP
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com><1B19ABD72889C245AE8EEE08AC24103A28C423231C@exmachina.office.kapper.net><BANLkTimrvA=Lgva-FAdXxWRhS-_UDuTgTA@mail.gmail.com><1BDDB8AD-4CBD-4515-A7AD-693A5E875523@freeswitch.org>
	<BANLkTi=YM4x3O6Xc-JDtXVKymKdVbcmP4Q@mail.gmail.com>
Message-ID: <86CDDEC506B5411AA8227253257519A6@e1705>

maybe change the param name to proxy_media_t38 ;)

----- Original Message ----- 
From: "Kristian Kielhofner" <kris at kriskinc.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, April 13, 2011 2:10 PM
Subject: Re: [Freeswitch-users] proxy SDP


Brian,

  For all of the confusion proxy media creates I still see cases where
it is useful...  It shouldn't be removed completely.

On Tue, Apr 12, 2011 at 8:55 PM, Brian West <brian at freeswitch.org> wrote:
> We aren't a proxy... we have transcended into this quasi proxy in some 
> scenarios which mostly involve t.38... as for proxy media DO NOT USE IT. 
> Just saying it might go away since the purpose of it is now not needed 
> since we have full t.38.
>
> Thanks,
> Brian
>

-- 
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 



From rogelio.perez at gmail.com  Thu Apr 14 02:49:01 2011
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Wed, 13 Apr 2011 19:49:01 -0300
Subject: [Freeswitch-users] Call state notifications
Message-ID: <4613C57B-AB5A-43FE-8DFE-76AF5492DA72@gmail.com>

Hi everyone,

I'm building a simple callshop panel for FS and I need to show the call state for every booth on a web page.
What is the best way to notify my application about call state changes (ringing, connected, disconnected)?
I'm using mod_python to send the notifications, but I'm not sure how to call the python script inside the dial plan.

Are there any examples out there?
Thanks!

Rogelio Perez

From krice at freeswitch.org  Thu Apr 14 03:09:54 2011
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 13 Apr 2011 18:09:54 -0500
Subject: [Freeswitch-users] Call state notifications
In-Reply-To: <4613C57B-AB5A-43FE-8DFE-76AF5492DA72@gmail.com>
Message-ID: <C9CB9772.13654%krice@freeswitch.org>

Event sockets is what you want....


On 4/13/11 5:49 PM, "Rogelio Perez" <rogelio.perez at gmail.com> wrote:

> Hi everyone,
> 
> I'm building a simple callshop panel for FS and I need to show the call state
> for every booth on a web page.
> What is the best way to notify my application about call state changes
> (ringing, connected, disconnected)?
> I'm using mod_python to send the notifications, but I'm not sure how to call
> the python script inside the dial plan.
> 
> Are there any examples out there?
> Thanks!
> 
> Rogelio Perez
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From rogelio.perez at gmail.com  Thu Apr 14 07:30:50 2011
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Thu, 14 Apr 2011 00:30:50 -0300
Subject: [Freeswitch-users]  Call state notifications
Message-ID: <C8A89431-F467-43B0-8D77-480DD2A1F4B4@gmail.com>

> Event sockets is what you want....

Thanks Ken, it looks like Event Sockets will do the job, but I'm not sure how.
I'm testing with netcat listening on port 8084, and then I've managed to connect to it by inserting this line on the dialplan:

<action application="socket" data="127.0.0.1:8084 async full"/>

Since I dont need to control the call but only receive notification events about the channel state changes I assume I have to send the command "myevents\n\n\" and the pass the variable socket_resume:true back, but this doesnt seem to be working.

sendmsg 
call-command: execute
execute-app-name: myevents\n\n\
socket_resume:true

This is my first attempt to write an application for FS so I need some guidance.
Thanks!




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From infos at madovsky.org  Thu Apr 14 07:33:49 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 13 Apr 2011 23:33:49 -0400
Subject: [Freeswitch-users] fax and jpg to tiff
Message-ID: <D49A74E3FE6D48E7BCCAFB7AFAFDE576@e1705>

sorry I don't have the personal email of
Steve Underwood (I saw a thread of him related of this subject), it's about a problem
of spandsp txfax and jpeg files.
I didn't succeed to find the right conversion to make the far end accept the fax.
I get always "far end cannot receive at the size of the image".
this problems occurs only from jpeg to tiff conversion.
I use imagemagick for that :

sudo /usr/bin/convert -page Letter -density 204x196 -resize 1728x1184 -units pixelsperinch -monochrome -compress Fax <fileToConvert>  <fileToSave>

I tried numrous options but no luck.

Thanks
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From infos at madovsky.org  Thu Apr 14 08:05:40 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 14 Apr 2011 00:05:40 -0400
Subject: [Freeswitch-users] fax and jpg to tiff
Message-ID: <1C10456422324E24A1C205759F4EFE1A@e1705>

is there the spandsp log

2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... 0...= 3G mobile network: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..1.= Can receive fax: Set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   1... ....= 2-D coding: Set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..00= Recording width: 215mm +- 1%
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... 10..= Recording length: Unlimited
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   1... ....= Extension indicator: Set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... .0..= Error correction mode (ECM): Non-ECM
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .0.. ....= T.6 coding: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   1... ....= Extension indicator: Set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... .0..= Polled sub-address: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... 0...= T.43 coding: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ...0 ....= Plane interleave: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   1... ....= Extension indicator: Set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... .0..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .0.. ....= Selective polling: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   1... ....= Extension indicator: Set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ...1= Sub-addressing: Set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..0.= Password: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   0... ....= Extension indicator: Not set
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Selected compression T.4 2-D (2)
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Trying to send file '/1302753792.tiff'
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Start sending document
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Minimum bits per row will be 0
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Starting page 1 of transfer
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Image width (200 pixels) not an acceptable FAX image width
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 The far end is incompatible
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from state 18 to 3
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Tx:  DCN with final frame tag
2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Tx:  ff 13 fb
2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 HDLC signal status is Carrier down (-1) in state 3
2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX
2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 0
2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 4
2011-04-14 00:03:31.624981 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Disconnecting
2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E
2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 0
2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 1
2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from state 3 to 2
2011-04-14 00:03:32.704744 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_E, state 2

  ----- Original Message ----- 
  From: Madovsky 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, April 13, 2011 11:33 PM
  Subject: fax and jpg to tiff


  sorry I don't have the personal email of
  Steve Underwood (I saw a thread of him related of this subject), it's about a problem
  of spandsp txfax and jpeg files.
  I didn't succeed to find the right conversion to make the far end accept the fax.
  I get always "far end cannot receive at the size of the image".
  this problems occurs only from jpeg to tiff conversion.
  I use imagemagick for that :

  sudo /usr/bin/convert -page Letter -density 204x196 -resize 1728x1184 -units pixelsperinch -monochrome -compress Fax <fileToConvert>  <fileToSave>

  I tried numrous options but no luck.

  Thanks
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From steveayre at gmail.com  Thu Apr 14 10:45:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 14 Apr 2011 07:45:01 +0100
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <86CDDEC506B5411AA8227253257519A6@e1705>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A28C423231C@exmachina.office.kapper.net>
	<BANLkTimrvA=Lgva-FAdXxWRhS-_UDuTgTA@mail.gmail.com>
	<1BDDB8AD-4CBD-4515-A7AD-693A5E875523@freeswitch.org>
	<BANLkTi=YM4x3O6Xc-JDtXVKymKdVbcmP4Q@mail.gmail.com>
	<86CDDEC506B5411AA8227253257519A6@e1705>
Message-ID: <BANLkTiksX56jd7zxWjrS1gwWc_J9_QT4nw@mail.gmail.com>

You misunderstand I think... it *isn't* needed for T38 any longer. Only in
very old versions.

-Steve


On 13 April 2011 23:17, Madovsky <infos at madovsky.org> wrote:

> maybe change the param name to proxy_media_t38 ;)
>
> ----- Original Message -----
> From: "Kristian Kielhofner" <kris at kriskinc.com>
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Sent: Wednesday, April 13, 2011 2:10 PM
> Subject: Re: [Freeswitch-users] proxy SDP
>
>
> Brian,
>
>  For all of the confusion proxy media creates I still see cases where
> it is useful...  It shouldn't be removed completely.
>
> On Tue, Apr 12, 2011 at 8:55 PM, Brian West <brian at freeswitch.org> wrote:
> > We aren't a proxy... we have transcended into this quasi proxy in some
> > scenarios which mostly involve t.38... as for proxy media DO NOT USE IT.
> > Just saying it might go away since the purpose of it is now not needed
> > since we have full t.38.
> >
> > Thanks,
> > Brian
> >
>
> --
> 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Apr 14 10:55:46 2011
From: sc_zhangming at sina.com (=?gb2312?B?1cXD9w==?=)
Date: Thu, 14 Apr 2011 14:55:46 +0800
Subject: [Freeswitch-users] mod_event_socket Whether can get freeswitch
	error messages
Message-ID: <8slc8n$of1hij@irxd5-201.sinamail.sina.com.cn>

freeswitch-users?

	          1. mod_event_socket Whether can get freeswitch error messages
			  2. how to get

?????????
??
 				

????????sc_zhangming at sina.com
??????????2011-04-14

From valery.kalinin at gmail.com  Thu Apr 14 12:57:38 2011
From: valery.kalinin at gmail.com (Valery Kalinin)
Date: Thu, 14 Apr 2011 14:57:38 +0600
Subject: [Freeswitch-users] Cannot compile freetdm
Message-ID: <BANLkTimhUWSeT48ZPeTihNMjC47sHL5-wA@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 sos at sokhapkin.dyndns.org  Thu Apr 14 14:22:52 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Apr 2011 06:22:52 -0400
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <BANLkTiksX56jd7zxWjrS1gwWc_J9_QT4nw@mail.gmail.com>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
	<86CDDEC506B5411AA8227253257519A6@e1705>
	<BANLkTiksX56jd7zxWjrS1gwWc_J9_QT4nw@mail.gmail.com>
Message-ID: <201104140622.52563.sos@sokhapkin.dyndns.org>

Proxy media is useful not for T38 only. It's a way for 2 NATed clients to 
communicate with each other if there are common codecs, no transcoding 
involved.

On Thursday 14 April 2011, Steven Ayre wrote:
> You misunderstand I think... it *isn't* needed for T38 any longer. Only in
> very old versions.
> 
> -Steve
> 
> On 13 April 2011 23:17, Madovsky <infos at madovsky.org> wrote:
> > maybe change the param name to proxy_media_t38 ;)
> > 
> > ----- Original Message -----
> > From: "Kristian Kielhofner" <kris at kriskinc.com>
> > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > Sent: Wednesday, April 13, 2011 2:10 PM
> > Subject: Re: [Freeswitch-users] proxy SDP
> > 
> > 
> > Brian,
> > 
> >  For all of the confusion proxy media creates I still see cases where
> > 
> > it is useful...  It shouldn't be removed completely.
> > 
> > On Tue, Apr 12, 2011 at 8:55 PM, Brian West <brian at freeswitch.org> wrote:
> > > We aren't a proxy... we have transcended into this quasi proxy in some
> > > scenarios which mostly involve t.38... as for proxy media DO NOT USE
> > > IT. Just saying it might go away since the purpose of it is now not
> > > needed since we have full t.38.
> > > 
> > > Thanks,
> > > Brian
> > 
> > --
> > 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
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From dnotivol at gmail.com  Thu Apr 14 15:32:44 2011
From: dnotivol at gmail.com (David Notivol)
Date: Thu, 14 Apr 2011 13:32:44 +0200
Subject: [Freeswitch-users] Non-transcoding G729 calls using Sangoma D100
Message-ID: <BANLkTimJiR=NnUNL4KtYZOSuQpLt9Jru7w@mail.gmail.com>

Hi all,

I'm using a Sangoma D100 card for transcoding with FreeSwitch. It seems the
normal behavior is a call not transcoding (g729-g729) is using a session in
the sangoma card, although it reports the call is not encoding nor decoding.

Since we can't load mod_g729 and mod_sangoma_g729 at the same time; do you
know if is there any way to avoid g729-g729 calls using Sangoma sessions?

Thanks in advance.


Regards,
David Notivol
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From steveayre at gmail.com  Thu Apr 14 16:10:20 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 14 Apr 2011 13:10:20 +0100
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <201104140622.52563.sos@sokhapkin.dyndns.org>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
	<86CDDEC506B5411AA8227253257519A6@e1705>
	<BANLkTiksX56jd7zxWjrS1gwWc_J9_QT4nw@mail.gmail.com>
	<201104140622.52563.sos@sokhapkin.dyndns.org>
Message-ID: <BANLkTi=0PMw9wVsTaFu-VSZ5EPo=mTR9NQ@mail.gmail.com>

Proxy media is not required for 2 NAT clients to talk to each other.

Transcoding won't be used if both legs use the same codec, and you can use
late-negotation/absolute_codec_string to encourage that.

-Steve


On 14 April 2011 11:22, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:

> Proxy media is useful not for T38 only. It's a way for 2 NATed clients to
> communicate with each other if there are common codecs, no transcoding
> involved.
>
> On Thursday 14 April 2011, Steven Ayre wrote:
> > You misunderstand I think... it *isn't* needed for T38 any longer. Only
> in
> > very old versions.
> >
> > -Steve
> >
> > On 13 April 2011 23:17, Madovsky <infos at madovsky.org> wrote:
> > > maybe change the param name to proxy_media_t38 ;)
> > >
> > > ----- Original Message -----
> > > From: "Kristian Kielhofner" <kris at kriskinc.com>
> > > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > > Sent: Wednesday, April 13, 2011 2:10 PM
> > > Subject: Re: [Freeswitch-users] proxy SDP
> > >
> > >
> > > Brian,
> > >
> > >  For all of the confusion proxy media creates I still see cases where
> > >
> > > it is useful...  It shouldn't be removed completely.
> > >
> > > On Tue, Apr 12, 2011 at 8:55 PM, Brian West <brian at freeswitch.org>
> wrote:
> > > > We aren't a proxy... we have transcended into this quasi proxy in
> some
> > > > scenarios which mostly involve t.38... as for proxy media DO NOT USE
> > > > IT. Just saying it might go away since the purpose of it is now not
> > > > needed since we have full t.38.
> > > >
> > > > Thanks,
> > > > Brian
> > >
> > > --
> > > 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
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sos at sokhapkin.dyndns.org  Thu Apr 14 16:28:12 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Thu, 14 Apr 2011 08:28:12 -0400
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <BANLkTi=0PMw9wVsTaFu-VSZ5EPo=mTR9NQ@mail.gmail.com>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
	<201104140622.52563.sos@sokhapkin.dyndns.org>
	<BANLkTi=0PMw9wVsTaFu-VSZ5EPo=mTR9NQ@mail.gmail.com>
Message-ID: <201104140828.12059.sos@sokhapkin.dyndns.org>

Do you mean

set proxy_media=true

is the equivalent of

set late_negotiation=true
bridge {absolute_codec_string=${ep_codec_string}}blah

and transcoding will never happen?

On Thursday 14 April 2011, Steven Ayre wrote:
> Proxy media is not required for 2 NAT clients to talk to each other.
> 
> Transcoding won't be used if both legs use the same codec, and you can use
> late-negotation/absolute_codec_string to encourage that.
> 
> -Steve
> 
> On 14 April 2011 11:22, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:
> > Proxy media is useful not for T38 only. It's a way for 2 NATed clients to
> > communicate with each other if there are common codecs, no transcoding
> > involved.
> > 
> > On Thursday 14 April 2011, Steven Ayre wrote:
> > > You misunderstand I think... it *isn't* needed for T38 any longer. Only
> > 
> > in
> > 
> > > very old versions.
> > > 
> > > -Steve
> > > 
> > > On 13 April 2011 23:17, Madovsky <infos at madovsky.org> wrote:
> > > > maybe change the param name to proxy_media_t38 ;)
> > > > 
> > > > ----- Original Message -----
> > > > From: "Kristian Kielhofner" <kris at kriskinc.com>
> > > > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> > > > Sent: Wednesday, April 13, 2011 2:10 PM
> > > > Subject: Re: [Freeswitch-users] proxy SDP
> > > > 
> > > > 
> > > > Brian,
> > > > 
> > > >  For all of the confusion proxy media creates I still see cases where
> > > > 
> > > > it is useful...  It shouldn't be removed completely.
> > > > 
> > > > On Tue, Apr 12, 2011 at 8:55 PM, Brian West <brian at freeswitch.org>
> > 
> > wrote:
> > > > > We aren't a proxy... we have transcended into this quasi proxy in
> > 
> > some
> > 
> > > > > scenarios which mostly involve t.38... as for proxy media DO NOT
> > > > > USE IT. Just saying it might go away since the purpose of it is
> > > > > now not needed since we have full t.38.
> > > > > 
> > > > > Thanks,
> > > > > Brian
> > > > 
> > > > --
> > > > 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
> > > > 
> > > > 
> > > > _______________________________________________
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > 
> > > > UNSUBSCRIBE:
> > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > 
> > > > http://www.freeswitch.org
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From steveayre at gmail.com  Thu Apr 14 17:02:20 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 14 Apr 2011 14:02:20 +0100
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <201104140828.12059.sos@sokhapkin.dyndns.org>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
	<201104140622.52563.sos@sokhapkin.dyndns.org>
	<BANLkTi=0PMw9wVsTaFu-VSZ5EPo=mTR9NQ@mail.gmail.com>
	<201104140828.12059.sos@sokhapkin.dyndns.org>
Message-ID: <BANLkTikeGaHpPO4Md_ieQhMzkzpr0n8+7w@mail.gmail.com>

I don't mean they're equivalent, because proxy_media has other sideaffects
such as disabling many features.

I mean there are different better ways to avoid transcoding.

What you suggested with bridge should indeed avoid it. Without the
absolute_codec_string the bleg would be offered all codecs supported by FS
which may be more than supported by the caller, however with that setting it
would only offer those supported by both the caller and FS. The callee will
pick one it supports (or return 488 Not Acceptable Here). The aleg will then
use the same codec as the bleg, so at that point both legs will be using the
same codec so no transcoding will take place (unless you try using something
like eavesdrop/record). You could then if you wanted handle the 488 Not
Acceptable Here by redialing without absolute_codec_string to try again,
this time the bleg could use a different codec than the aleg (transcoding).

-Steve


On 14 April 2011 13:28, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:

> Do you mean
>
> set proxy_media=true
>
> is the equivalent of
>
> set late_negotiation=true
> bridge {absolute_codec_string=${ep_codec_string}}blah
>
> and transcoding will never happen?
>
> On Thursday 14 April 2011, Steven Ayre wrote:
> > Proxy media is not required for 2 NAT clients to talk to each other.
> >
> > Transcoding won't be used if both legs use the same codec, and you can
> use
> > late-negotation/absolute_codec_string to encourage that.
> >
> > -Steve
> >
> > On 14 April 2011 11:22, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
> wrote:
> > > Proxy media is useful not for T38 only. It's a way for 2 NATed clients
> to
> > > communicate with each other if there are common codecs, no transcoding
> > > involved.
> > >
> > > On Thursday 14 April 2011, Steven Ayre wrote:
> > > > You misunderstand I think... it *isn't* needed for T38 any longer.
> Only
> > >
> > > in
> > >
> > > > very old versions.
> > > >
> > > > -Steve
> > > >
> > > > On 13 April 2011 23:17, Madovsky <infos at madovsky.org> wrote:
> > > > > maybe change the param name to proxy_media_t38 ;)
> > > > >
> > > > > ----- Original Message -----
> > > > > From: "Kristian Kielhofner" <kris at kriskinc.com>
> > > > > To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org
> >
> > > > > Sent: Wednesday, April 13, 2011 2:10 PM
> > > > > Subject: Re: [Freeswitch-users] proxy SDP
> > > > >
> > > > >
> > > > > Brian,
> > > > >
> > > > >  For all of the confusion proxy media creates I still see cases
> where
> > > > >
> > > > > it is useful...  It shouldn't be removed completely.
> > > > >
> > > > > On Tue, Apr 12, 2011 at 8:55 PM, Brian West <brian at freeswitch.org>
> > >
> > > wrote:
> > > > > > We aren't a proxy... we have transcended into this quasi proxy in
> > >
> > > some
> > >
> > > > > > scenarios which mostly involve t.38... as for proxy media DO NOT
> > > > > > USE IT. Just saying it might go away since the purpose of it is
> > > > > > now not needed since we have full t.38.
> > > > > >
> > > > > > Thanks,
> > > > > > Brian
> > > > >
> > > > > --
> > > > > 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
> > > > >
> > > > >
> > > > > _______________________________________________
> > > > > FreeSWITCH-users mailing list
> > > > > FreeSWITCH-users at lists.freeswitch.org
> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >
> > > > > UNSUBSCRIBE:
> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >
> > > > > http://www.freeswitch.org
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at 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  Thu Apr 14 17:25:45 2011
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 14 Apr 2011 21:25:45 +0800
Subject: [Freeswitch-users] fax and jpg to tiff
In-Reply-To: <1C10456422324E24A1C205759F4EFE1A@e1705>
References: <1C10456422324E24A1C205759F4EFE1A@e1705>
Message-ID: <4DA6F5D9.1020108@coppice.org>

Hi Madovsky,

Your log says very clearly what is wrong with the file you are trying to 
send. What further information are you looking for?

Steve


On 04/14/2011 12:05 PM, Madovsky wrote:
> is there the spandsp log
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... ...0= Store and forward Internet fax (T.37): Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... .0..= Real-time Internet fax (T.38): Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... 0...= 3G mobile network: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> ..0. ....= V.8 capabilities: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .0.. ....= Preferred octets: 256 octets
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... ...0= Ready to transmit a fax document (polling): Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... ..1.= Can receive fax: Set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> 1... ....= 2-D coding: Set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... ..00= Recording width: 215mm +- 1%
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... 10..= Recording length: Unlimited
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = 
> T3.85
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> 1... ....= Extension indicator: Set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... ..0.= Compressed/uncompressed mode: Compressed
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... .0..= Error correction mode (ECM): Non-ECM
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .0.. ....= T.6 coding: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> 1... ....= Extension indicator: Set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... ...0= "Field not valid" supported: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... ..0.= Multiple selective polling: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... .0..= Polled sub-address: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... 0...= T.43 coding: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> ...0 ....= Plane interleave: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .0.. ....= Reserved for the use of extended voice coding set: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> 1... ....= Extension indicator: Set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... ...1= R8x15.4lines/mm: Set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... ..0.= 300x300pels/25.4mm: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... .0..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... 0...= Inch-based resolution preferred: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> ...1 ....= Metric-based resolution preferred: Set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .0.. ....= Selective polling: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> 1... ....= Extension indicator: Set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... ...1= Sub-addressing: Set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... ..0.= Password: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .... .0..= Ready to transmit a data file (polling): Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> ...0 ....= Binary file transfer (BFT): Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> ..0. ....= Document transfer mode (DTM): Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> .0.. ....= Electronic data interchange (EDI): Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
> 0... ....= Extension indicator: Not set
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Selected compression T.4 2-D (2)
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Trying to send file '/1302753792.tiff'
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Start sending document
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Minimum bits per row will be 0
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Starting page 1 of transfer
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Image width (200 pixels) not an acceptable FAX image width
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 The 
> far end is incompatible
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Changing from state 18 to 3
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Tx:  DCN with final frame tag
> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Tx:  ff 13 fb
> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> HDLC signal status is Carrier down (-1) in state 3
> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX
> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set 
> rx type 0
> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set 
> tx type 4
> 2011-04-14 00:03:31.624981 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Send complete in phase T30_PHASE_D_TX, state 3
> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Send complete in phase T30_PHASE_D_TX, state 3
> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Disconnecting
> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Changing from phase T30_PHASE_D_TX to T30_PHASE_E
> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set 
> rx type 0
> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set 
> tx type 1
> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Changing from state 3 to 2
> 2011-04-14 00:03:32.704744 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
> Send complete in phase T30_PHASE_E, state 2
>
>     ----- Original Message -----
>     *From:* Madovsky <mailto:infos at madovsky.org>
>     *To:* freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     *Sent:* Wednesday, April 13, 2011 11:33 PM
>     *Subject:* fax and jpg to tiff
>
>     sorry I don't have the personal email of
>     Steve Underwood (I saw a thread of him related of this subject),
>     it's about a problem
>     of spandsp txfax and jpeg files.
>     I didn't succeed to find the right conversion to make the far end
>     accept the fax.
>     I get always "far end cannot receive at the size of the image".
>     this problems occurs only from jpeg to tiff conversion.
>     I use imagemagick for that :
>     sudo /usr/bin/convert -page Letter -density 204x196 -resize
>     1728x1184 -units pixelsperinch -monochrome -compress Fax
>     <fileToConvert> <fileToSave>
>     I tried numrous options but no luck.
>     Thanks
>
>



From lfurrea at gmail.com  Thu Apr 14 17:31:27 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Thu, 14 Apr 2011 07:31:27 -0600
Subject: [Freeswitch-users] Using txfax from a LUA script
Message-ID: <BANLkTimKsHW+JSi7jZ4ggUwpQEjy=2WoVQ@mail.gmail.com>

Hi all,

I noticed that I am not able to use TXFAX from a LUA script as follows:

obSession = freeswitch.Session("freetdm/1/1/9999")

if obSession:ready() then
    session:execute("txfax", "/rxfax.tiff")
else

My objective was to send a fax to a fax machine connected to an FXS.

I does work when I use:

originate freetdm/1/1/9999 &txfax(/rxfax.tiff)

But I wanted to be able to retry based on DISCONNECT CAUSES and the like.


Just for peace of mind I wanted to get some input from you on the causes of
this "limitation", and maybe get some suggestions as to any other options?


TIA
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From infos at madovsky.org  Thu Apr 14 19:20:30 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 14 Apr 2011 11:20:30 -0400
Subject: [Freeswitch-users] fax and jpg to tiff
References: <1C10456422324E24A1C205759F4EFE1A@e1705>
	<4DA6F5D9.1020108@coppice.org>
Message-ID: <A6A5793A27324ECE935D3D62C04DEF84@e1705>

Hi Steve,

if you talk about this line
> Image width (200 pixels) not an acceptable FAX image width
I tried many different size but it says always the same.
what image width is acceptable usually ?

Thanks

----- Original Message ----- 
From: "Steve Underwood" <steveu at coppice.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, April 14, 2011 9:25 AM
Subject: Re: [Freeswitch-users] fax and jpg to tiff


> Hi Madovsky,
> 
> Your log says very clearly what is wrong with the file you are trying to 
> send. What further information are you looking for?
> 
> Steve
> 
> 
> On 04/14/2011 12:05 PM, Madovsky wrote:
>> is there the spandsp log
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... ...0= Store and forward Internet fax (T.37): Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... .0..= Real-time Internet fax (T.38): Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... 0...= 3G mobile network: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> ..0. ....= V.8 capabilities: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .0.. ....= Preferred octets: 256 octets
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... ...0= Ready to transmit a fax document (polling): Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... ..1.= Can receive fax: Set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> 1... ....= 2-D coding: Set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... ..00= Recording width: 215mm +- 1%
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... 10..= Recording length: Unlimited
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = 
>> T3.85
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> 1... ....= Extension indicator: Set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... ..0.= Compressed/uncompressed mode: Compressed
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... .0..= Error correction mode (ECM): Non-ECM
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .0.. ....= T.6 coding: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> 1... ....= Extension indicator: Set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... ...0= "Field not valid" supported: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... ..0.= Multiple selective polling: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... .0..= Polled sub-address: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... 0...= T.43 coding: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> ...0 ....= Plane interleave: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .0.. ....= Reserved for the use of extended voice coding set: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> 1... ....= Extension indicator: Set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... ...1= R8x15.4lines/mm: Set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... ..0.= 300x300pels/25.4mm: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... .0..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... 0...= Inch-based resolution preferred: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> ...1 ....= Metric-based resolution preferred: Set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .0.. ....= Selective polling: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> 1... ....= Extension indicator: Set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... ...1= Sub-addressing: Set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... ..0.= Password: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .... .0..= Ready to transmit a data file (polling): Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> ...0 ....= Binary file transfer (BFT): Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> ..0. ....= Document transfer mode (DTM): Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> .0.. ....= Electronic data interchange (EDI): Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   
>> 0... ....= Extension indicator: Not set
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Selected compression T.4 2-D (2)
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Trying to send file '/1302753792.tiff'
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Start sending document
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Minimum bits per row will be 0
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Starting page 1 of transfer
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Image width (200 pixels) not an acceptable FAX image width
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 The 
>> far end is incompatible
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Changing from state 18 to 3
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Tx:  DCN with final frame tag
>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Tx:  ff 13 fb
>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> HDLC signal status is Carrier down (-1) in state 3
>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX
>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set 
>> rx type 0
>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set 
>> tx type 4
>> 2011-04-14 00:03:31.624981 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Send complete in phase T30_PHASE_D_TX, state 3
>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Send complete in phase T30_PHASE_D_TX, state 3
>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Disconnecting
>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Changing from phase T30_PHASE_D_TX to T30_PHASE_E
>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set 
>> rx type 0
>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set 
>> tx type 1
>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Changing from state 3 to 2
>> 2011-04-14 00:03:32.704744 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 
>> Send complete in phase T30_PHASE_E, state 2
>>
>>     ----- Original Message -----
>>     *From:* Madovsky <mailto:infos at madovsky.org>
>>     *To:* freeswitch-users at lists.freeswitch.org
>>     <mailto:freeswitch-users at lists.freeswitch.org>
>>     *Sent:* Wednesday, April 13, 2011 11:33 PM
>>     *Subject:* fax and jpg to tiff
>>
>>     sorry I don't have the personal email of
>>     Steve Underwood (I saw a thread of him related of this subject),
>>     it's about a problem
>>     of spandsp txfax and jpeg files.
>>     I didn't succeed to find the right conversion to make the far end
>>     accept the fax.
>>     I get always "far end cannot receive at the size of the image".
>>     this problems occurs only from jpeg to tiff conversion.
>>     I use imagemagick for that :
>>     sudo /usr/bin/convert -page Letter -density 204x196 -resize
>>     1728x1184 -units pixelsperinch -monochrome -compress Fax
>>     <fileToConvert> <fileToSave>
>>     I tried numrous options but no luck.
>>     Thanks
>>
>>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From steveu at coppice.org  Thu Apr 14 19:59:09 2011
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 14 Apr 2011 23:59:09 +0800
Subject: [Freeswitch-users] fax and jpg to tiff
In-Reply-To: <A6A5793A27324ECE935D3D62C04DEF84@e1705>
References: <1C10456422324E24A1C205759F4EFE1A@e1705>	<4DA6F5D9.1020108@coppice.org>
	<A6A5793A27324ECE935D3D62C04DEF84@e1705>
Message-ID: <4DA719CD.1080007@coppice.org>

Hi Madovsky,

The standard width for a normal sized fax, whether it is standard, fine 
or super-fine resolution length-wise, is 1728 pixels. Wider faxes (e.g. 
A3) are possible, but it is rare to find a machine which supports them.

Steve


On 04/14/2011 11:20 PM, Madovsky wrote:
> Hi Steve,
>
> if you talk about this line
>> Image width (200 pixels) not an acceptable FAX image width
> I tried many different size but it says always the same.
> what image width is acceptable usually ?
>
> Thanks
>
> ----- Original Message -----
> From: "Steve Underwood"<steveu at coppice.org>
> To: "FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
> Sent: Thursday, April 14, 2011 9:25 AM
> Subject: Re: [Freeswitch-users] fax and jpg to tiff
>
>
>> Hi Madovsky,
>>
>> Your log says very clearly what is wrong with the file you are trying to
>> send. What further information are you looking for?
>>
>> Steve
>>
>>
>> On 04/14/2011 12:05 PM, Madovsky wrote:
>>> is there the spandsp log
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... ...0= Store and forward Internet fax (T.37): Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... .0..= Real-time Internet fax (T.38): Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... 0...= 3G mobile network: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> ..0. ....= V.8 capabilities: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .0.. ....= Preferred octets: 256 octets
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... ...0= Ready to transmit a fax document (polling): Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... ..1.= Can receive fax: Set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> 1... ....= 2-D coding: Set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... ..00= Recording width: 215mm +- 1%
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... 10..= Recording length: Unlimited
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 =
>>> T3.85
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> 1... ....= Extension indicator: Set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... ..0.= Compressed/uncompressed mode: Compressed
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... .0..= Error correction mode (ECM): Non-ECM
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .0.. ....= T.6 coding: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> 1... ....= Extension indicator: Set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... ...0= "Field not valid" supported: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... ..0.= Multiple selective polling: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... .0..= Polled sub-address: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... 0...= T.43 coding: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> ...0 ....= Plane interleave: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .0.. ....= Reserved for the use of extended voice coding set: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> 1... ....= Extension indicator: Set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... ...1= R8x15.4lines/mm: Set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... ..0.= 300x300pels/25.4mm: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... .0..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... 0...= Inch-based resolution preferred: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> ...1 ....= Metric-based resolution preferred: Set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .0.. ....= Selective polling: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> 1... ....= Extension indicator: Set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... ...1= Sub-addressing: Set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... ..0.= Password: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .... .0..= Ready to transmit a data file (polling): Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> ...0 ....= Binary file transfer (BFT): Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> ..0. ....= Document transfer mode (DTM): Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> .0.. ....= Electronic data interchange (EDI): Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> 0... ....= Extension indicator: Not set
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Selected compression T.4 2-D (2)
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Trying to send file '/1302753792.tiff'
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Start sending document
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Minimum bits per row will be 0
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Starting page 1 of transfer
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Image width (200 pixels) not an acceptable FAX image width
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 The
>>> far end is incompatible
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Changing from state 18 to 3
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Tx:  DCN with final frame tag
>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Tx:  ff 13 fb
>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> HDLC signal status is Carrier down (-1) in state 3
>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX
>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>> rx type 0
>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>> tx type 4
>>> 2011-04-14 00:03:31.624981 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Send complete in phase T30_PHASE_D_TX, state 3
>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Send complete in phase T30_PHASE_D_TX, state 3
>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Disconnecting
>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Changing from phase T30_PHASE_D_TX to T30_PHASE_E
>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>> rx type 0
>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>> tx type 1
>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Changing from state 3 to 2
>>> 2011-04-14 00:03:32.704744 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>> Send complete in phase T30_PHASE_E, state 2
>>>
>>>      ----- Original Message -----
>>>      *From:* Madovsky<mailto:infos at madovsky.org>
>>>      *To:* freeswitch-users at lists.freeswitch.org
>>>      <mailto:freeswitch-users at lists.freeswitch.org>
>>>      *Sent:* Wednesday, April 13, 2011 11:33 PM
>>>      *Subject:* fax and jpg to tiff
>>>
>>>      sorry I don't have the personal email of
>>>      Steve Underwood (I saw a thread of him related of this subject),
>>>      it's about a problem
>>>      of spandsp txfax and jpeg files.
>>>      I didn't succeed to find the right conversion to make the far end
>>>      accept the fax.
>>>      I get always "far end cannot receive at the size of the image".
>>>      this problems occurs only from jpeg to tiff conversion.
>>>      I use imagemagick for that :
>>>      sudo /usr/bin/convert -page Letter -density 204x196 -resize
>>>      1728x1184 -units pixelsperinch -monochrome -compress Fax
>>>      <fileToConvert>  <fileToSave>
>>>      I tried numrous options but no luck.
>>>      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
>



From infos at madovsky.org  Thu Apr 14 20:01:05 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 14 Apr 2011 12:01:05 -0400
Subject: [Freeswitch-users] proxy SDP
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com><1B19ABD72889C245AE8EEE08AC24103A28C423231C@exmachina.office.kapper.net><BANLkTimrvA=Lgva-FAdXxWRhS-_UDuTgTA@mail.gmail.com><1BDDB8AD-4CBD-4515-A7AD-693A5E875523@freeswitch.org><BANLkTi=YM4x3O6Xc-JDtXVKymKdVbcmP4Q@mail.gmail.com><86CDDEC506B5411AA8227253257519A6@e1705>
	<BANLkTiksX56jd7zxWjrS1gwWc_J9_QT4nw@mail.gmail.com>
Message-ID: <07CC1DBD9F314CABB6E6735822053C55@e1705>

ok, so maybe remove the word "proxy" that confused a lot of people here (even me at the start) :)
  ----- Original Message ----- 
  From: Steven Ayre 
  To: FreeSWITCH Users Help 
  Sent: Thursday, April 14, 2011 2:45 AM
  Subject: Re: [Freeswitch-users] proxy SDP


  You misunderstand I think... it *isn't* needed for T38 any longer. Only in very old versions.

  -Steve



  On 13 April 2011 23:17, Madovsky <infos at madovsky.org> wrote:

    maybe change the param name to proxy_media_t38 ;)


    ----- Original Message -----
    From: "Kristian Kielhofner" <kris at kriskinc.com>
    To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
    Sent: Wednesday, April 13, 2011 2:10 PM
    Subject: Re: [Freeswitch-users] proxy SDP


    Brian,

     For all of the confusion proxy media creates I still see cases where
    it is useful...  It shouldn't be removed completely.

    On Tue, Apr 12, 2011 at 8:55 PM, Brian West <brian at freeswitch.org> wrote:
    > We aren't a proxy... we have transcended into this quasi proxy in some
    > scenarios which mostly involve t.38... as for proxy media DO NOT USE IT.
    > Just saying it might go away since the purpose of it is now not needed
    > since we have full t.38.
    >
    > Thanks,
    > Brian
    >

    --
    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


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




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


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at 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  Thu Apr 14 20:08:52 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 14 Apr 2011 12:08:52 -0400
Subject: [Freeswitch-users] fax and jpg to tiff
References: <1C10456422324E24A1C205759F4EFE1A@e1705>	<4DA6F5D9.1020108@coppice.org><A6A5793A27324ECE935D3D62C04DEF84@e1705>
	<4DA719CD.1080007@coppice.org>
Message-ID: <F15A7C0F6BC648DBBB939260F7BC21F5@e1705>

Ok Steve,

strange since I use exactly 

sudo /usr/bin/convert -page Letter -density 204x196 -resize 1728x1184 -units pixelsperinch -monochrome -compress Fax <pathToFile> <pathToNewFile>

but the spandsp log shows 881px as width :(

thanks 

----- Original Message ----- 
From: "Steve Underwood" <steveu at coppice.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, April 14, 2011 11:59 AM
Subject: Re: [Freeswitch-users] fax and jpg to tiff


> Hi Madovsky,
> 
> The standard width for a normal sized fax, whether it is standard, fine 
> or super-fine resolution length-wise, is 1728 pixels. Wider faxes (e.g. 
> A3) are possible, but it is rare to find a machine which supports them.
> 
> Steve
> 
> 
> On 04/14/2011 11:20 PM, Madovsky wrote:
>> Hi Steve,
>>
>> if you talk about this line
>>> Image width (200 pixels) not an acceptable FAX image width
>> I tried many different size but it says always the same.
>> what image width is acceptable usually ?
>>
>> Thanks
>>
>> ----- Original Message -----
>> From: "Steve Underwood"<steveu at coppice.org>
>> To: "FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
>> Sent: Thursday, April 14, 2011 9:25 AM
>> Subject: Re: [Freeswitch-users] fax and jpg to tiff
>>
>>
>>> Hi Madovsky,
>>>
>>> Your log says very clearly what is wrong with the file you are trying to
>>> send. What further information are you looking for?
>>>
>>> Steve
>>>
>>>
>>> On 04/14/2011 12:05 PM, Madovsky wrote:
>>>> is there the spandsp log
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...0= Store and forward Internet fax (T.37): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= Real-time Internet fax (T.38): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... 0...= 3G mobile network: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..0. ....= V.8 capabilities: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= Preferred octets: 256 octets
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...0= Ready to transmit a fax document (polling): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..1.= Can receive fax: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= 2-D coding: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..00= Recording width: 215mm +- 1%
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... 10..= Recording length: Unlimited
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 =
>>>> T3.85
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= Extension indicator: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..0.= Compressed/uncompressed mode: Compressed
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= Error correction mode (ECM): Non-ECM
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= T.6 coding: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= Extension indicator: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...0= "Field not valid" supported: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..0.= Multiple selective polling: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= Polled sub-address: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... 0...= T.43 coding: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ...0 ....= Plane interleave: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= Reserved for the use of extended voice coding set: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= Extension indicator: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...1= R8x15.4lines/mm: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..0.= 300x300pels/25.4mm: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... 0...= Inch-based resolution preferred: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ...1 ....= Metric-based resolution preferred: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= Selective polling: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= Extension indicator: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...1= Sub-addressing: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..0.= Password: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= Ready to transmit a data file (polling): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ...0 ....= Binary file transfer (BFT): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..0. ....= Document transfer mode (DTM): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= Electronic data interchange (EDI): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 0... ....= Extension indicator: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Selected compression T.4 2-D (2)
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Trying to send file '/1302753792.tiff'
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Start sending document
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Minimum bits per row will be 0
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Starting page 1 of transfer
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Image width (200 pixels) not an acceptable FAX image width
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 The
>>>> far end is incompatible
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Changing from state 18 to 3
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Tx:  DCN with final frame tag
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Tx:  ff 13 fb
>>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> HDLC signal status is Carrier down (-1) in state 3
>>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX
>>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>>> rx type 0
>>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>>> tx type 4
>>>> 2011-04-14 00:03:31.624981 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Send complete in phase T30_PHASE_D_TX, state 3
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Send complete in phase T30_PHASE_D_TX, state 3
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Disconnecting
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Changing from phase T30_PHASE_D_TX to T30_PHASE_E
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>>> rx type 0
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>>> tx type 1
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Changing from state 3 to 2
>>>> 2011-04-14 00:03:32.704744 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Send complete in phase T30_PHASE_E, state 2
>>>>
>>>>      ----- Original Message -----
>>>>      *From:* Madovsky<mailto:infos at madovsky.org>
>>>>      *To:* freeswitch-users at lists.freeswitch.org
>>>>      <mailto:freeswitch-users at lists.freeswitch.org>
>>>>      *Sent:* Wednesday, April 13, 2011 11:33 PM
>>>>      *Subject:* fax and jpg to tiff
>>>>
>>>>      sorry I don't have the personal email of
>>>>      Steve Underwood (I saw a thread of him related of this subject),
>>>>      it's about a problem
>>>>      of spandsp txfax and jpeg files.
>>>>      I didn't succeed to find the right conversion to make the far end
>>>>      accept the fax.
>>>>      I get always "far end cannot receive at the size of the image".
>>>>      this problems occurs only from jpeg to tiff conversion.
>>>>      I use imagemagick for that :
>>>>      sudo /usr/bin/convert -page Letter -density 204x196 -resize
>>>>      1728x1184 -units pixelsperinch -monochrome -compress Fax
>>>>      <fileToConvert>  <fileToSave>
>>>>      I tried numrous options but no luck.
>>>>      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
>>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Apr 14 20:29:51 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 14 Apr 2011 12:29:51 -0400
Subject: [Freeswitch-users] fax and jpg to tiff
References: <1C10456422324E24A1C205759F4EFE1A@e1705>	<4DA6F5D9.1020108@coppice.org><A6A5793A27324ECE935D3D62C04DEF84@e1705>
	<4DA719CD.1080007@coppice.org>
Message-ID: <627F6C8DFEEB4D06AEF69C3E199F5DDD@e1705>

Steve the link below is the same thread you had with callweaver

http://www.mail-archive.com/callweaver-users at callweaver.org/msg01241.html

thanks

----- Original Message ----- 
From: "Steve Underwood" <steveu at coppice.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, April 14, 2011 11:59 AM
Subject: Re: [Freeswitch-users] fax and jpg to tiff


> Hi Madovsky,
>
> The standard width for a normal sized fax, whether it is standard, fine
> or super-fine resolution length-wise, is 1728 pixels. Wider faxes (e.g.
> A3) are possible, but it is rare to find a machine which supports them.
>
> Steve
>
>
> On 04/14/2011 11:20 PM, Madovsky wrote:
>> Hi Steve,
>>
>> if you talk about this line
>>> Image width (200 pixels) not an acceptable FAX image width
>> I tried many different size but it says always the same.
>> what image width is acceptable usually ?
>>
>> Thanks
>>
>> ----- Original Message -----
>> From: "Steve Underwood"<steveu at coppice.org>
>> To: "FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
>> Sent: Thursday, April 14, 2011 9:25 AM
>> Subject: Re: [Freeswitch-users] fax and jpg to tiff
>>
>>
>>> Hi Madovsky,
>>>
>>> Your log says very clearly what is wrong with the file you are trying to
>>> send. What further information are you looking for?
>>>
>>> Steve
>>>
>>>
>>> On 04/14/2011 12:05 PM, Madovsky wrote:
>>>> is there the spandsp log
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...0= Store and forward Internet fax (T.37): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= Real-time Internet fax (T.38): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... 0...= 3G mobile network: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..0. ....= V.8 capabilities: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= Preferred octets: 256 octets
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...0= Ready to transmit a fax document (polling): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..1.= Can receive fax: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= 2-D coding: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..00= Recording width: 215mm +- 1%
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... 10..= Recording length: Unlimited
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 =
>>>> T3.85
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= Extension indicator: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..0.= Compressed/uncompressed mode: Compressed
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= Error correction mode (ECM): Non-ECM
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= T.6 coding: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= Extension indicator: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...0= "Field not valid" supported: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..0.= Multiple selective polling: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= Polled sub-address: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... 0...= T.43 coding: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ...0 ....= Plane interleave: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= Reserved for the use of extended voice coding set: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= Extension indicator: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...1= R8x15.4lines/mm: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..0.= 300x300pels/25.4mm: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... 0...= Inch-based resolution preferred: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ...1 ....= Metric-based resolution preferred: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= Selective polling: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= Extension indicator: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...1= Sub-addressing: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..0.= Password: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= Ready to transmit a data file (polling): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ...0 ....= Binary file transfer (BFT): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..0. ....= Document transfer mode (DTM): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= Electronic data interchange (EDI): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 0... ....= Extension indicator: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Selected compression T.4 2-D (2)
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Trying to send file '/1302753792.tiff'
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Start sending document
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Minimum bits per row will be 0
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Starting page 1 of transfer
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Image width (200 pixels) not an acceptable FAX image width
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 The
>>>> far end is incompatible
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Changing from state 18 to 3
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Tx:  DCN with final frame tag
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Tx:  ff 13 fb
>>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> HDLC signal status is Carrier down (-1) in state 3
>>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX
>>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>>> rx type 0
>>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>>> tx type 4
>>>> 2011-04-14 00:03:31.624981 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Send complete in phase T30_PHASE_D_TX, state 3
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Send complete in phase T30_PHASE_D_TX, state 3
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Disconnecting
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Changing from phase T30_PHASE_D_TX to T30_PHASE_E
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>>> rx type 0
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>>> tx type 1
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Changing from state 3 to 2
>>>> 2011-04-14 00:03:32.704744 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Send complete in phase T30_PHASE_E, state 2
>>>>
>>>>      ----- Original Message -----
>>>>      *From:* Madovsky<mailto:infos at madovsky.org>
>>>>      *To:* freeswitch-users at lists.freeswitch.org
>>>>      <mailto:freeswitch-users at lists.freeswitch.org>
>>>>      *Sent:* Wednesday, April 13, 2011 11:33 PM
>>>>      *Subject:* fax and jpg to tiff
>>>>
>>>>      sorry I don't have the personal email of
>>>>      Steve Underwood (I saw a thread of him related of this subject),
>>>>      it's about a problem
>>>>      of spandsp txfax and jpeg files.
>>>>      I didn't succeed to find the right conversion to make the far end
>>>>      accept the fax.
>>>>      I get always "far end cannot receive at the size of the image".
>>>>      this problems occurs only from jpeg to tiff conversion.
>>>>      I use imagemagick for that :
>>>>      sudo /usr/bin/convert -page Letter -density 204x196 -resize
>>>>      1728x1184 -units pixelsperinch -monochrome -compress Fax
>>>>      <fileToConvert>  <fileToSave>
>>>>      I tried numrous options but no luck.
>>>>      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
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From vivid333 at 163.com  Thu Apr 14 05:21:27 2011
From: vivid333 at 163.com (vivid333)
Date: Thu, 14 Apr 2011 09:21:27 +0800 (CST)
Subject: [Freeswitch-users] The Usage of uuid_preprocess && echo cancellation
Message-ID: <6224958b.125d6.12f51993b3d.Coremail.vivid333@163.com>

Hi:
          1. can any one know the usage of uuid_preprocess?
          2. according to the FreeSwitch code, this command would add a bug to the uuid session, initialize echo parameters, and in functions switch_core_session_read_frame/switch_core_session_write_frame do echo handle. but  the echo still exist. do any one help me ? thanks..

      Echo Scenes:
      Telephone 1000, 1001 registered the FreeSwitch, 1000 call 1001, 1001 answered( two of them are handfree, producing scary sound);

Try Solution:
Launch the command-line ulility:
#> fs_cli
#>: show channels
uuid1, uuid2

#>:uuid_preprocess uuid1 recho_cancel=true;
#>:uuid_preprocess uuid2 recho_cancel=true;

but the echo still exist. can any one help me, thanks!
 
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From vivid333 at 163.com  Thu Apr 14 07:02:35 2011
From: vivid333 at 163.com (vivid)
Date: Thu, 14 Apr 2011 11:02:35 +0800
Subject: [Freeswitch-users] FreeSwitch Debug Full Log(version 1.0.6)
Message-ID: <4DA663CB.2040107@163.com>

how can I use FreeSwitch to trace *the Function call Procedure* in *fs_cli*.
(using Command? configuration?)


[DEBUG] *switch_ivr_bridge.c*:911 switch_ivr_multi_threaded_bridge()
sofia/external/1XXX4951027 receive message [BRIDGE]
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From alex8207744 at gmail.com  Thu Apr 14 19:12:52 2011
From: alex8207744 at gmail.com (Alex Zhou)
Date: Thu, 14 Apr 2011 23:12:52 +0800
Subject: [Freeswitch-users] Why the c program does not work?Event-Name is
	allways "SOCKET_DATA"
Message-ID: <BANLkTikn38SZvMzvs=t7k_5cRs=8vgrxmw@mail.gmail.com>

Hi! I'm new user to freeswitch.I write a c program to dial and play a wav
file. I want to know hangup-cause.But it does not work.The Event-Name i got
is "SOCKET_DATA"  not "CHANNEL_HANGUP". Can you help me to fix it and get
hangup-cause?

Then i will dial lot of number but only on work time
(9:00-11:00,14:00-17:00).So I want to use crontab to send custom event to
crotrol dial start or stop.So how to send and receive custom event ?
Beside the crotab and custom event,any other better idea to control when to
start or stop ?

Thanks!

#include <stdio.h>
#include <stdlib.h>
#include <esl.h>


int main(void)
{
    esl_handle_t handle = {{0}};

    esl_status_t status1,status2,status3;
    int done = 0;
    esl_status_t status;
    time_t exp = 0;

    //esl_global_set_default_logger(7);
    status1 = esl_connect(&handle, "localhost", 8021, NULL, "ClueCon");

    esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_ANSWER
CHANNEL_ORIGINATE CHANNEL_HANGUP");

    status2 = esl_send_recv(&handle, "api originate user/1001
&playback(/tmp/a.wav)\n\n");

    if (handle.last_sr_event && handle.last_sr_event->body) {
        printf("Sucess --- %s\n", handle.last_sr_event->body);
    } else {
        // this is unlikely to happen with api or bgapi (which is hardcoded
above) but prefix but may be true for other commands
        printf("Fail --- %s\n", handle.last_sr_reply);
    }

    while((status = esl_recv_event(&handle, 1,NULL)) != ESL_FAIL)
    {
                if (status == ESL_SUCCESS)
        {
            char *type = esl_event_get_header(handle.last_event,
"content-type");
            printf("--- %s\n", type);
            printf("--- %s\n",esl_event_get_header(handle.last_event,
"Event-Name"));
            //next to get Hangup-Cause from Event-Name:CHANNEL_HANGUP
        }
    }
    esl_disconnect(&handle);
    return 0;
}


[freeswitch at fs ~]$ ./test
Sucess --- +OK 3fe60603-0ed4-4a6d-9322-dd906433b3cc

--- text/event-plain
--- SOCKET_DATA
--- text/event-plain
--- SOCKET_DATA
--- text/event-plain
--- SOCKET_DATA
--- text/event-plain
--- SOCKET_DATA
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From uttampegu at gmail.com  Thu Apr 14 20:11:45 2011
From: uttampegu at gmail.com (uttampegu at gmail.com)
Date: Thu, 14 Apr 2011 16:11:45 +0000
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 58, Issue 92
In-Reply-To: <mailman.7427.1302797334.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.7427.1302797334.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <1589104011-1302797464-cardhu_decombobulator_blackberry.rim.net-1883329271-@b27.c13.bise7.blackberry>

Is there any easy step by step guide to configure freeswitch work a inbound IVR ?
Regards
Uttam
Sent from BlackBerry? on Airtel

-----Original Message-----
From: freeswitch-users-request at lists.freeswitch.org
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Thu, 14 Apr 2011 20:08:54 
To: <freeswitch-users at lists.freeswitch.org>
Reply-To: freeswitch-users at lists.freeswitch.org
Subject: FreeSWITCH-users Digest, Vol 58, Issue 92

Send FreeSWITCH-users mailing list submissions to
	freeswitch-users at lists.freeswitch.org

To subscribe or unsubscribe via the World Wide Web, visit
	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
	freeswitch-users-request at lists.freeswitch.org

You can reach the person managing the list at
	freeswitch-users-owner at lists.freeswitch.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-users digest..."


From hamid.bilal at techway.it  Thu Apr 14 15:53:40 2011
From: hamid.bilal at techway.it (Hamid Bilal)
Date: Thu, 14 Apr 2011 13:53:40 +0200
Subject: [Freeswitch-users] Relay Proxy-Authentication requests to the caller
Message-ID: <002d01cbfa9a$99abc5b0$cd035110$@techway.it>

Hi All!

 

I would like to know if  there is a param we could set in sofia profile to
let freeswitch relay Proxy-Authentication requests to the caller instead of
absorbing it.

 

Regards,

Hamid

 

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From helmut.kuper at ewetel.de  Thu Apr 14 20:39:33 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Thu, 14 Apr 2011 18:39:33 +0200
Subject: [Freeswitch-users] fax and jpg to tiff
In-Reply-To: <F15A7C0F6BC648DBBB939260F7BC21F5@e1705>
References: <1C10456422324E24A1C205759F4EFE1A@e1705>	<4DA6F5D9.1020108@coppice.org><A6A5793A27324ECE935D3D62C04DEF84@e1705>	<4DA719CD.1080007@coppice.org>
	<F15A7C0F6BC648DBBB939260F7BC21F5@e1705>
Message-ID: <4DA72345.6060807@ewetel.de>

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

Hi Madovsky,

I guess u u r using imagemagick's convert. So give this a try:

convert -page Letter -density 204x196 -resize 1728!x1184 -units
pixelsperinch -monochrome -compress Fax <pathToFile> <pathToNewFile>

The "!" after 1728 forces the width to exactly 1728 pixels.


Am 14.04.2011 18:08, schrieb Madovsky:
> Ok Steve,
>  
> strange since I use exactly
>  
> sudo /usr/bin/convert -page Letter -density 204x196 -resize 1728x1184
> -units pixelsperinch -monochrome -compress Fax <pathToFile> <pathToNewFile>
>  
> but the spandsp log shows 881px as width :(
>  
> thanks 

regards
helmut
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAk2nI0UACgkQ4tZeNddg3dws5gCfRdYEVhf/iKMDH7Mq8ASwtELb
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From mcampbellsmith at gmail.com  Wed Apr 13 15:32:09 2011
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Wed, 13 Apr 2011 21:32:09 +1000
Subject: [Freeswitch-users] Unable to leave voicemail message longer than a
	few seconds
Message-ID: <BANLkTimUoL8g6tt-iH1htNw2oeadN_sUOg@mail.gmail.com>

HI!

I am having an issue which must have been introduced at my last upgrade of
FS (FreeSWITCH Version 1.0.head (git-b737851 2011-03-25 00-13-39 +0100))

Callers are unable to leave voicemail messages of any length; the voicemail
application terminates the recording and transfers to the ivr.

My dialplan is quite simple. Extract here:

                        <action application="bridge"
data="${group_extension}"/>
                        <action application="set_user"
data="${dialed_extension}@${domain}"/>
                        <action application="answer"/>
                        <action application="sleep" data="1000"/>
                        <action application="voicemail" data="default
${domain_name} ${dialed_extension}"/>
                  </condition>
            </extension>

And the only place the message length value is defined is in
voicemail.conf.xml in the autoloads conf directory.  It is set to default.
      <param name="max-record-len" value="300"/>

Below is an extract of the debug logs.  I am only allowed to leave a message
in this case for 6 seconds (between times 2011-04-13 21:26:41.098532
and 2011-04-13 21:26:47.37119)

Dialplan: sofia/internal/1000 at 192.168.1.120 Action set(call_timeout=22)
Dialplan: sofia/internal/1000 at 192.168.1.120 Action
set(sip_contact_user=${caller_id_number})
Dialplan: sofia/internal/1000 at 192.168.1.120 Action
export(sip_contact_user=${caller_id_number})
Dialplan: sofia/internal/1000 at 192.168.1.120 Action info()
Dialplan: sofia/internal/1000 at 192.168.1.120 Action bind_meta_app(1 a s
execute_extension::4001 XML default)
Dialplan: sofia/internal/1000 at 192.168.1.120 Action bind_meta_app(3 b s
execute_extension::cf XML features)
Dialplan: sofia/internal/1000 at 192.168.1.120 Action bind_meta_app(2 a s
execute_extension::checkpin XML features)
Dialplan: sofia/internal/1000 at 192.168.1.120 Action bind_meta_app(9 b s
record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
Dialplan: sofia/internal/1000 at 192.168.1.120 Action
bridge(${group_extension})
Dialplan: sofia/internal/1000 at 192.168.1.120 Action
set_user(${dialed_extension}@${domain})
Dialplan: sofia/internal/1000 at 192.168.1.120 Action answer()
Dialplan: sofia/internal/1000 at 192.168.1.120 Action sleep(1000)
Dialplan: sofia/internal/1000 at 192.168.1.120 Action voicemail(default
${domain_name} ${dialed_extension})
:
:
2011-04-13 21:26:29.611121 [DEBUG] switch_core_session.c:709 Send signal
sofia/internal/1000 at 192.168.1.120 [BREAK]
2011-04-13 21:26:29.611121 [DEBUG] switch_channel.c:2813 (sofia/internal/
1000 at 192.168.1.120) Callstate Change EARLY -> ACTIVE
2011-04-13 21:26:29.611121 [NOTICE] mod_dptools.c:929 Channel
[sofia/internal/1000 at 192.168.1.120] has been answered
EXECUTE sofia/internal/1000 at 192.168.1.120 sleep(1000)
2011-04-13 21:26:29.702581 [DEBUG] sofia.c:4744 Channel sofia/internal/
1000 at 192.168.1.120 entering state [ready][200]
2011-04-13 21:26:29.758583 [DEBUG] switch_rtp.c:3082 Correct ip/port
confirmed.
EXECUTE sofia/internal/1000 at 192.168.1.120 voicemail(default
markcs.dyndns.org 1000)
2011-04-13 21:26:30.750645 [DEBUG] switch_ivr_play_say.c:1291 Codec
Activated L16 at 8000hz 1 channels 30ms
2011-04-13 21:26:40.050617 [DEBUG] switch_ivr_play_say.c:1635 done playing
file
2011-04-13 21:26:40.050617 [DEBUG] switch_ivr_play_say.c:63 No language
specified - Using [en]
2011-04-13 21:26:41.098532 [DEBUG] switch_ivr_play_say.c:581 Raw Codec
Activated
2011-04-13 21:26:41.098532 [DEBUG] switch_core_codec.c:116 sofia/internal/
1000 at 192.168.1.120 Push codec L16:70

2011-04-13 21:26:47.371195 [DEBUG] switch_core_codec.c:141 sofia/internal/
1000 at 192.168.1.120 Restore previous codec PCMU:0.
2011-04-13 21:26:47.371195 [DEBUG] switch_ivr_play_say.c:63 No language
specified - Using [en]
2011-04-13 21:26:47.418939 [DEBUG] switch_ivr_play_say.c:244 Handle
play-file:[voicemail/vm-press.wav] (en:en)
2011-04-13 21:26:47.418939 [DEBUG] switch_ivr_play_say.c:1291 Codec
Activated L16 at 8000hz 1 channels 30ms
2011-04-13 21:26:47.758529 [DEBUG] switch_ivr_play_say.c:1635 done playing
file
2011-04-13 21:26:47.878531 [DEBUG] switch_ivr_play_say.c:244 Handle say:[1]
(en:en)
2011-04-13 21:26:47.878531 [DEBUG] switch_ivr_play_say.c:1291 Codec
Activated L16 at 8000hz 1 channels 30ms

Any ideas why this would be?  I checked Jira but couldn't find any reference
to bugs there.

Thanks!
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From mcampbellsmith at gmail.com  Thu Apr 14 14:44:04 2011
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Thu, 14 Apr 2011 20:44:04 +1000
Subject: [Freeswitch-users] Unable to leave voicemail message longer
	than a few seconds
In-Reply-To: <BANLkTimUoL8g6tt-iH1htNw2oeadN_sUOg@mail.gmail.com>
References: <BANLkTimUoL8g6tt-iH1htNw2oeadN_sUOg@mail.gmail.com>
Message-ID: <BANLkTinRFEZFtA4p3jMyyaknatqhJC59rQ@mail.gmail.com>

I fixed this by removing the contiue="false" part of the extension:
<extension name="homegroup" continue="false">

Why does this affect the voicemail application?


On Wed, Apr 13, 2011 at 9:32 PM, Mark Campbell-Smith <
mcampbellsmith at gmail.com> wrote:

> HI!
>
> I am having an issue which must have been introduced at my last upgrade of
> FS (FreeSWITCH Version 1.0.head (git-b737851 2011-03-25 00-13-39 +0100))
>
> Callers are unable to leave voicemail messages of any length; the voicemail
> application terminates the recording and transfers to the ivr.
>
> My dialplan is quite simple. Extract here:
>
>                         <action application="bridge"
> data="${group_extension}"/>
>                         <action application="set_user"
> data="${dialed_extension}@${domain}"/>
>                         <action application="answer"/>
>                         <action application="sleep" data="1000"/>
>                         <action application="voicemail" data="default
> ${domain_name} ${dialed_extension}"/>
>                   </condition>
>             </extension>
>
> And the only place the message length value is defined is in
> voicemail.conf.xml in the autoloads conf directory.  It is set to default.
>       <param name="max-record-len" value="300"/>
>
> Below is an extract of the debug logs.  I am only allowed to leave a
> message in this case for 6 seconds (between times 2011-04-13 21:26:41.098532
> and 2011-04-13 21:26:47.37119)
>
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action set(call_timeout=22)
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action
> set(sip_contact_user=${caller_id_number})
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action
> export(sip_contact_user=${caller_id_number})
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action info()
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action bind_meta_app(1 a s
> execute_extension::4001 XML default)
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action bind_meta_app(3 b s
> execute_extension::cf XML features)
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action bind_meta_app(2 a s
> execute_extension::checkpin XML features)
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action bind_meta_app(9 b s
> record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav)
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action
> bridge(${group_extension})
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action
> set_user(${dialed_extension}@${domain})
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action answer()
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action sleep(1000)
> Dialplan: sofia/internal/1000 at 192.168.1.120 Action voicemail(default
> ${domain_name} ${dialed_extension})
> :
> :
> 2011-04-13 21:26:29.611121 [DEBUG] switch_core_session.c:709 Send signal
> sofia/internal/1000 at 192.168.1.120 [BREAK]
> 2011-04-13 21:26:29.611121 [DEBUG] switch_channel.c:2813 (sofia/internal/
> 1000 at 192.168.1.120) Callstate Change EARLY -> ACTIVE
> 2011-04-13 21:26:29.611121 [NOTICE] mod_dptools.c:929 Channel
> [sofia/internal/1000 at 192.168.1.120] has been answered
> EXECUTE sofia/internal/1000 at 192.168.1.120 sleep(1000)
> 2011-04-13 21:26:29.702581 [DEBUG] sofia.c:4744 Channel sofia/internal/
> 1000 at 192.168.1.120 entering state [ready][200]
> 2011-04-13 21:26:29.758583 [DEBUG] switch_rtp.c:3082 Correct ip/port
> confirmed.
> EXECUTE sofia/internal/1000 at 192.168.1.120 voicemail(default
> markcs.dyndns.org 1000)
> 2011-04-13 21:26:30.750645 [DEBUG] switch_ivr_play_say.c:1291 Codec
> Activated L16 at 8000hz 1 channels 30ms
> 2011-04-13 21:26:40.050617 [DEBUG] switch_ivr_play_say.c:1635 done playing
> file
> 2011-04-13 21:26:40.050617 [DEBUG] switch_ivr_play_say.c:63 No language
> specified - Using [en]
> 2011-04-13 21:26:41.098532 [DEBUG] switch_ivr_play_say.c:581 Raw Codec
> Activated
> 2011-04-13 21:26:41.098532 [DEBUG] switch_core_codec.c:116 sofia/internal/
> 1000 at 192.168.1.120 Push codec L16:70
>
> 2011-04-13 21:26:47.371195 [DEBUG] switch_core_codec.c:141 sofia/internal/
> 1000 at 192.168.1.120 Restore previous codec PCMU:0.
> 2011-04-13 21:26:47.371195 [DEBUG] switch_ivr_play_say.c:63 No language
> specified - Using [en]
> 2011-04-13 21:26:47.418939 [DEBUG] switch_ivr_play_say.c:244 Handle
> play-file:[voicemail/vm-press.wav] (en:en)
> 2011-04-13 21:26:47.418939 [DEBUG] switch_ivr_play_say.c:1291 Codec
> Activated L16 at 8000hz 1 channels 30ms
> 2011-04-13 21:26:47.758529 [DEBUG] switch_ivr_play_say.c:1635 done playing
> file
> 2011-04-13 21:26:47.878531 [DEBUG] switch_ivr_play_say.c:244 Handle say:[1]
> (en:en)
> 2011-04-13 21:26:47.878531 [DEBUG] switch_ivr_play_say.c:1291 Codec
> Activated L16 at 8000hz 1 channels 30ms
>
> Any ideas why this would be?  I checked Jira but couldn't find any
> reference to bugs there.
>
> Thanks!
>
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From romon.zaman at gmail.com  Wed Apr 13 16:16:42 2011
From: romon.zaman at gmail.com (romon.zaman)
Date: Wed, 13 Apr 2011 18:16:42 +0600
Subject: [Freeswitch-users] fax tone detection failed
References: <mailman.7319.1302674417.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <3BE856AEC9884941B3F865AE687F573B@hometarat>

hi,

i am trying to detect fax tone on outgoing calls.
here is my javascript code

+++++++++++++++++++++++

session1 = new 
Session("{ignore_early_media=false}sofia/gateway/flowroute/"+argv[0]+"");

session1.execute("tone_detect", "fax 1100 r +30000 set 
'fax_tone_detected=true' 1");
msleep(argv[1]);
session1.hangup();

++++++++++++++++++++++++

i  checked pcap file and  using wireshark , i can hear sone tone on rtp 
audio playback.

but freeswitch cant detect that..

is there any other solution to find out specific tone like fax??

thanks in advance


----- Original Message ----- 
From: <freeswitch-users-request at lists.freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Wednesday, April 13, 2011 12:00 PM
Subject: FreeSWITCH-users Digest, Vol 58, Issue 81


> Send FreeSWITCH-users mailing list submissions to
> freeswitch-users at lists.freeswitch.org
>
> To subscribe or unsubscribe via the World Wide Web, visit
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> freeswitch-users-request at lists.freeswitch.org
>
> You can reach the person managing the list at
> freeswitch-users-owner at lists.freeswitch.org
>
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
>
>
>
> __________ Information from ESET Smart Security, version of virus 
> signature database 6014 (20110404) __________
>
> The message was checked by ESET Smart Security.
>
> http://www.eset.com
>
>


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


> Today's Topics:
>
>   1. Incompatible destination (FERNANDO VILLARROEL)
>   2. FreeSWITCH: No Longer The Best Kept Secret In OSS VoIP
>      Software (Michael Collins)
>   3. Re: occasional ~5s delay during bind_meta_app
>      execute_extenstion (Michael Collins)
>   4. Re: Incompatible destination (Michael Collins)
>   5. Re: Freeswitch Server Down (ovvenkat)
>   6. Re: Freeswitch Server Down (guru singh)
>
>
>
> __________ Information from ESET Smart Security, version of virus 
> signature database 6014 (20110404) __________
>
> The message was checked by ESET Smart Security.
>
> http://www.eset.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
>
>
>
> __________ Information from ESET Smart Security, version of virus 
> signature database 6014 (20110404) __________
>
> The message was checked by ESET Smart Security.
>
> http://www.eset.com
>
> 


__________ Information from ESET Smart Security, version of virus signature database 6014 (20110404) __________

The message was checked by ESET Smart Security.

http://www.eset.com





From msc at freeswitch.org  Thu Apr 14 20:43:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Apr 2011 09:43:30 -0700
Subject: [Freeswitch-users] fax tone detection failed
In-Reply-To: <3BE856AEC9884941B3F865AE687F573B@hometarat>
References: <mailman.7319.1302674417.24555.freeswitch-users@lists.freeswitch.org>
	<3BE856AEC9884941B3F865AE687F573B@hometarat>
Message-ID: <BANLkTi=nVRfGHVxe-W24BKvkhfDjj_DVhg@mail.gmail.com>

On Wed, Apr 13, 2011 at 5:16 AM, romon.zaman <romon.zaman at gmail.com> wrote:

> hi,
>
> i am trying to detect fax tone on outgoing calls.
> here is my javascript code
>
> +++++++++++++++++++++++
>
> session1 = new
> Session("{ignore_early_media=false}sofia/gateway/flowroute/"+argv[0]+"");
>
> session1.execute("tone_detect", "fax 1100 r +30000 set
> 'fax_tone_detected=true' 1");
> msleep(argv[1]);
> session1.hangup();
>
> ++++++++++++++++++++++++
>
> i  checked pcap file and  using wireshark , i can hear sone tone on rtp
> audio playback.
>
> but freeswitch cant detect that..
>
> is there any other solution to find out specific tone like fax??
>
> thanks in advance
>

Was this the one we talked about yesterday where your fax machine was really
spitting out a tone around 2096Hz?

-MC
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From michal.bielicki at seventhsignal.de  Thu Apr 14 20:47:29 2011
From: michal.bielicki at seventhsignal.de (=?utf-8?B?TWljaGFsIEJpZWxpY2tp?=)
Date: Thu, 14 Apr 2011 18:47:29 +0200
Subject: [Freeswitch-users] =?utf-8?q?Antw=2E=3A__fax_and_jpg_to_tiff?=
Message-ID: <mailman.2.1302799658.24552.freeswitch-users@lists.freeswitch.org>

Danke f?r die Information

Gesendet mit meinem HTC

----- Reply message -----
Von: "Madovsky" <infos at madovsky.org>
An: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Betreff: [Freeswitch-users] fax and jpg to tiff
Datum: Do., Apr. 14, 2011 18:29


Steve the link below is the same thread you had with callweaver

http://www.mail-archive.com/callweaver-users at callweaver.org/msg01241.html

thanks

----- Original Message ----- 
From: "Steve Underwood" <steveu at coppice.org>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, April 14, 2011 11:59 AM
Subject: Re: [Freeswitch-users] fax and jpg to tiff


> Hi Madovsky,
>
> The standard width for a normal sized fax, whether it is standard, fine
> or super-fine resolution length-wise, is 1728 pixels. Wider faxes (e.g.
> A3) are possible, but it is rare to find a machine which supports them.
>
> Steve
>
>
> On 04/14/2011 11:20 PM, Madovsky wrote:
>> Hi Steve,
>>
>> if you talk about this line
>>> Image width (200 pixels) not an acceptable FAX image width
>> I tried many different size but it says always the same.
>> what image width is acceptable usually ?
>>
>> Thanks
>>
>> ----- Original Message -----
>> From: "Steve Underwood"<steveu at coppice.org>
>> To: "FreeSWITCH Users Help"<freeswitch-users at lists.freeswitch.org>
>> Sent: Thursday, April 14, 2011 9:25 AM
>> Subject: Re: [Freeswitch-users] fax and jpg to tiff
>>
>>
>>> Hi Madovsky,
>>>
>>> Your log says very clearly what is wrong with the file you are trying to
>>> send. What further information are you looking for?
>>>
>>> Steve
>>>
>>>
>>> On 04/14/2011 12:05 PM, Madovsky wrote:
>>>> is there the spandsp log
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...0= Store and forward Internet fax (T.37): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= Real-time Internet fax (T.38): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... 0...= 3G mobile network: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..0. ....= V.8 capabilities: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= Preferred octets: 256 octets
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...0= Ready to transmit a fax document (polling): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..1.= Can receive fax: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= 2-D coding: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..00= Recording width: 215mm +- 1%
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... 10..= Recording length: Unlimited
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 =
>>>> T3.85
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= Extension indicator: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..0.= Compressed/uncompressed mode: Compressed
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= Error correction mode (ECM): Non-ECM
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= T.6 coding: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= Extension indicator: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...0= "Field not valid" supported: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..0.= Multiple selective polling: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= Polled sub-address: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... 0...= T.43 coding: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ...0 ....= Plane interleave: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= Reserved for the use of extended voice coding set: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= Extension indicator: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...1= R8x15.4lines/mm: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..0.= 300x300pels/25.4mm: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... 0...= Inch-based resolution preferred: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ...1 ....= Metric-based resolution preferred: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= Selective polling: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 1... ....= Extension indicator: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ...1= Sub-addressing: Set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... ..0.= Password: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .... .0..= Ready to transmit a data file (polling): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ...0 ....= Binary file transfer (BFT): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> ..0. ....= Document transfer mode (DTM): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> .0.. ....= Electronic data interchange (EDI): Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> 0... ....= Extension indicator: Not set
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Selected compression T.4 2-D (2)
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Trying to send file '/1302753792.tiff'
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Start sending document
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Minimum bits per row will be 0
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Starting page 1 of transfer
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Image width (200 pixels) not an acceptable FAX image width
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 The
>>>> far end is incompatible
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Changing from state 18 to 3
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Tx:  DCN with final frame tag
>>>> 2011-04-14 00:03:30.484254 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Tx:  ff 13 fb
>>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> HDLC signal status is Carrier down (-1) in state 3
>>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX
>>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>>> rx type 0
>>>> 2011-04-14 00:03:30.564178 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>>> tx type 4
>>>> 2011-04-14 00:03:31.624981 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Send complete in phase T30_PHASE_D_TX, state 3
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Send complete in phase T30_PHASE_D_TX, state 3
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Disconnecting
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Changing from phase T30_PHASE_D_TX to T30_PHASE_E
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>>> rx type 0
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set
>>>> tx type 1
>>>> 2011-04-14 00:03:31.705007 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Changing from state 3 to 2
>>>> 2011-04-14 00:03:32.704744 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
>>>> Send complete in phase T30_PHASE_E, state 2
>>>>
>>>>      ----- Original Message -----
>>>>      *From:* Madovsky<mailto:infos at madovsky.org>
>>>>      *To:* freeswitch-users at lists.freeswitch.org
>>>>      <mailto:freeswitch-users at lists.freeswitch.org>
>>>>      *Sent:* Wednesday, April 13, 2011 11:33 PM
>>>>      *Subject:* fax and jpg to tiff
>>>>
>>>>      sorry I don't have the personal email of
>>>>      Steve Underwood (I saw a thread of him related of this subject),
>>>>      it's about a problem
>>>>      of spandsp txfax and jpeg files.
>>>>      I didn't succeed to find the right conversion to make the far end
>>>>      accept the fax.
>>>>      I get always "far end cannot receive at the size of the image".
>>>>      this problems occurs only from jpeg to tiff conversion.
>>>>      I use imagemagick for that :
>>>>      sudo /usr/bin/convert -page Letter -density 204x196 -resize
>>>>      1728x1184 -units pixelsperinch -monochrome -compress Fax
>>>>      <fileToConvert>  <fileToSave>
>>>>      I tried numrous options but no luck.
>>>>      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
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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  Thu Apr 14 20:50:29 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 14 Apr 2011 12:50:29 -0400
Subject: [Freeswitch-users] fax and jpg to tiff
References: <1C10456422324E24A1C205759F4EFE1A@e1705>	<4DA6F5D9.1020108@coppice.org><A6A5793A27324ECE935D3D62C04DEF84@e1705>	<4DA719CD.1080007@coppice.org><F15A7C0F6BC648DBBB939260F7BC21F5@e1705>
	<4DA72345.6060807@ewetel.de>
Message-ID: <194F3FCAB32848889F2ACD51D7658607@e1705>

Gold medal for Helmut !! :D
I invite you to the next oktober feist ! :)

Thanks

Franck

----- Original Message ----- 
From: "Helmut Kuper" <helmut.kuper at ewetel.de>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, April 14, 2011 12:39 PM
Subject: Re: [Freeswitch-users] fax and jpg to tiff


> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Madovsky,
>
> I guess u u r using imagemagick's convert. So give this a try:
>
> convert -page Letter -density 204x196 -resize 1728!x1184 -units
> pixelsperinch -monochrome -compress Fax <pathToFile> <pathToNewFile>
>
> The "!" after 1728 forces the width to exactly 1728 pixels.
>
>
> Am 14.04.2011 18:08, schrieb Madovsky:
>> Ok Steve,
>>
>> strange since I use exactly
>>
>> sudo /usr/bin/convert -page Letter -density 204x196 -resize 1728x1184
>> -units pixelsperinch -monochrome -compress Fax <pathToFile> 
>> <pathToNewFile>
>>
>> but the spandsp log shows 881px as width :(
>>
>> thanks
>
> regards
> helmut
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.10 (MingW32)
> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/
>
> iEYEARECAAYFAk2nI0UACgkQ4tZeNddg3dws5gCfRdYEVhf/iKMDH7Mq8ASwtELb
> 9VwAnidyc42JJbnp52BxaXj3AyGR1OlN
> =jM8D
> -----END PGP SIGNATURE-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From mikepetty at gmail.com  Thu Apr 14 20:46:10 2011
From: mikepetty at gmail.com (mike petty)
Date: Thu, 14 Apr 2011 09:46:10 -0700
Subject: [Freeswitch-users] "Nets" Capability
Message-ID: <BANLkTi=gfkLcrJDny-_y2neo72_acPXgJA@mail.gmail.com>

I have a quick question regarding freeswitch?s capabilities.  I?ve
downloaded it and am diving into it right now.  One part of the
functionality that I?m looking for is something like radio nets.



In this scenario, there would be multiple ?nets? or conference room-like
channels, and the user would essentially be in all of them, but be able to
set talk/listen capabilities on each individually.



For instance, you might have a ?Master Coordination Net? a ?Troubleshooting
Net? and a ?Local Area Net?.



The user would be in all of those, and set Listen on all, or some of the
nets, but only talk on the ?Troubleshooting Net?, but then would like to
talk on Master Coord and Local Area, so he would set talk on both of those
and his voice would go out to just those listening on those two nets?  Most
of the users would be SIP clients, but some would be coming in off of PSTN
and other interfaces.



It seems like this should be possible, but right now I?m sorting through all
of the different modules and settings and everything, and am really starting
to wonder if this is possible or not.


Eventually, my plan would be to move the Talk/Listen status to external push
button switches that I read in via some digital inputs, and then I would
write a little program to set the talk/listen based upon the button status
to provide a control panel type of interface for a user.


Thanks.  Freeswitch looks amazing so far, and so much more intuitive than
Asterisk!



Mike
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From romon.zaman at gmail.com  Thu Apr 14 20:57:58 2011
From: romon.zaman at gmail.com (romon.zaman)
Date: Thu, 14 Apr 2011 22:57:58 +0600
Subject: [Freeswitch-users] fax tone detection failed
References: <mailman.7319.1302674417.24555.freeswitch-users@lists.freeswitch.org><3BE856AEC9884941B3F865AE687F573B@hometarat>
	<BANLkTi=nVRfGHVxe-W24BKvkhfDjj_DVhg@mail.gmail.com>
Message-ID: <B11730398BF74A299031A8B958D497D3@hometarat>

hello MC,
you are right..
before this published here, you help me to get a solution.

actually, my fax machine was not sending standard fax tone.  but it worked with actual tone frequency.

so, if someone ever face this problem, recommendation is to get pcap file and extract audio file from RTP with wireshark and analyze audio with audicity to get exact frequency of tone.

and when u use tone_detect with exact tone frequency, it will work perfectly.


thanks for  your help

  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Thursday, April 14, 2011 10:43 PM
  Subject: Re: [Freeswitch-users] fax tone detection failed





  On Wed, Apr 13, 2011 at 5:16 AM, romon.zaman <romon.zaman at gmail.com> wrote:

    hi,

    i am trying to detect fax tone on outgoing calls.
    here is my javascript code

    +++++++++++++++++++++++

    session1 = new
    Session("{ignore_early_media=false}sofia/gateway/flowroute/"+argv[0]+"");

    session1.execute("tone_detect", "fax 1100 r +30000 set
    'fax_tone_detected=true' 1");
    msleep(argv[1]);
    session1.hangup();

    ++++++++++++++++++++++++

    i  checked pcap file and  using wireshark , i can hear sone tone on rtp
    audio playback.

    but freeswitch cant detect that..

    is there any other solution to find out specific tone like fax??

    thanks in advance



  Was this the one we talked about yesterday where your fax machine was really spitting out a tone around 2096Hz?


  -MC


__________ Information from ESET Smart Security, version of virus signature database 6041 (20110414) __________

The message was checked by ESET Smart Security.

http://www.eset.com

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From dave at dchorton.com  Thu Apr 14 21:08:24 2011
From: dave at dchorton.com (=?utf-8?B?ZGF2ZUBkY2hvcnRvbi5jb20=?=)
Date: Thu, 14 Apr 2011 13:08:24 -0400
Subject: [Freeswitch-users]
	=?utf-8?q?FreeSWITCH-users_Digest=2C_Vol_58=2C?=
	=?utf-8?q?_Issue_92?=
Message-ID: <mailman.3.1302800937.24552.freeswitch-users@lists.freeswitch.org>



Sent from my HTC Inspire? 4G on AT&T

----- Reply message -----
From: freeswitch-users-request at lists.freeswitch.org
To: <freeswitch-users at lists.freeswitch.org>
Subject: FreeSWITCH-users Digest, Vol 58, Issue 92
Date: Thu, Apr 14, 2011 12:08 pm


Today's Topics:

   1. Re: fax and jpg to tiff (Steve Underwood)
   2. Re: proxy SDP (Madovsky)
   3. Re: fax and jpg to tiff (Madovsky)

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From msc at freeswitch.org  Thu Apr 14 21:12:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Apr 2011 10:12:10 -0700
Subject: [Freeswitch-users] Call state notifications
In-Reply-To: <C8A89431-F467-43B0-8D77-480DD2A1F4B4@gmail.com>
References: <C8A89431-F467-43B0-8D77-480DD2A1F4B4@gmail.com>
Message-ID: <BANLkTikgTPekcDf3LtaN8DRPfoDq99dGNg@mail.gmail.com>

Rogelio,

Are you trying to make a single web page that shows the status of multiple
calls? If so then you probably want what we call "inbound" event socket
versus the "outbound" event socket that you described below. (Inbound and
outbound are from the perspective of FreeSWITCH, so when you have a dialplan
with the socket app it makes an "outbound" socket connection to your program
that is listening on the IP/port...)

If you want to sit and watch the event socket and see all the channel state
change events then I recommend a few things:

#1 - read up on the event socket on the wiki, especially the event socket
commands like "filter"
#2 - get a feel for what events come down the pipe by using fs_cli:
  open fs_cli
  type "/log 0" to turn off logging
  type "/events plain all" to turn on events

Just sit and watch the console. You'll be amazed (and overwhelmed) at all of
the events and information that come to you.

#3 - See if you can set a filter to get only the events you want. While
still at the fs_cli and while watching all the events come to you, try
typing this: "/filter Event-Name CHANNEL_STATE_CHANGE"

That will filter out everything *except* channel state changes. Then make a
call and watch the screen. Do stuff like transfers and hangups, etc. and see
what the events look like.

If that all seems like too much then you can cheat and just send a "show
channels" every 2 seconds and parse the results. ;)

Have fun!
-MC

On Wed, Apr 13, 2011 at 8:30 PM, Rogelio Perez <rogelio.perez at gmail.com>wrote:

> > Event sockets is what you want....
>
> Thanks Ken, it looks like Event Sockets will do the job, but I'm not sure
> how.
> I'm testing with netcat listening on port 8084, and then I've managed to
> connect to it by inserting this line on the dialplan:
>
> <action application="socket" data="127.0.0.1:8084 async full"/>
>
> Since I dont need to control the call but only receive notification events
> about the channel state changes I assume I have to send the command
> "myevents\n\n\" and the pass the variable socket_resume:true back, but this
> doesnt seem to be working.
>
> sendmsg
> call-command: execute
> execute-app-name: myevents\n\n\
> socket_resume:true
>
>
> This is my first attempt to write an application for FS so I need some
> guidance.
> 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  Thu Apr 14 21:14:12 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Apr 2011 10:14:12 -0700
Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 58, Issue 92
In-Reply-To: <1589104011-1302797464-cardhu_decombobulator_blackberry.rim.net-1883329271-@b27.c13.bise7.blackberry>
References: <mailman.7427.1302797334.24555.freeswitch-users@lists.freeswitch.org>
	<1589104011-1302797464-cardhu_decombobulator_blackberry.rim.net-1883329271-@b27.c13.bise7.blackberry>
Message-ID: <BANLkTimMi3qSh7rTENfzRbapfRRjWZ=j_w@mail.gmail.com>

FYI, I think you accidentally replied to a digest message. Next time be sure
to send a new message.

The best way to learn about inbound IVRs and FreeSWITCH is to look at sample
extension 5000. Also, get the FreeSWITCH book and look at chapter 6.

-MC

On Thu, Apr 14, 2011 at 9:11 AM, <uttampegu at gmail.com> wrote:

> Is there any easy step by step guide to configure freeswitch work a inbound
> IVR ?
> Regards
> Uttam
> Sent from BlackBerry? on Airtel
>
>
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From msc at freeswitch.org  Thu Apr 14 21:20:17 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Apr 2011 10:20:17 -0700
Subject: [Freeswitch-users] Using txfax from a LUA script
In-Reply-To: <BANLkTimKsHW+JSi7jZ4ggUwpQEjy=2WoVQ@mail.gmail.com>
References: <BANLkTimKsHW+JSi7jZ4ggUwpQEjy=2WoVQ@mail.gmail.com>
Message-ID: <BANLkTi=9xbaifeFgbpkXLw9+04d=V2=y-Q@mail.gmail.com>

What happens when you call the txfax app from Lua? Does it even attempt to
run? Just curious. In any case, I think you can do a combo of dp and Lua by
transferring the call to an extension that does the bridge and if it fails
you can check the failure cause in a Lua script and try again. You'll just
need a chan var for the number of attempts.

-MC

On Thu, Apr 14, 2011 at 6:31 AM, Luis F Urrea <lfurrea at gmail.com> wrote:

> Hi all,
>
> I noticed that I am not able to use TXFAX from a LUA script as follows:
>
> obSession = freeswitch.Session("freetdm/1/1/9999")
>
> if obSession:ready() then
>     session:execute("txfax", "/rxfax.tiff")
> else
>
> My objective was to send a fax to a fax machine connected to an FXS.
>
> I does work when I use:
>
> originate freetdm/1/1/9999 &txfax(/rxfax.tiff)
>
> But I wanted to be able to retry based on DISCONNECT CAUSES and the like.
>
>
> Just for peace of mind I wanted to get some input from you on the causes of
> this "limitation", and maybe get some suggestions as to any other options?
>
>
> TIA
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From avi at avimarcus.net  Thu Apr 14 21:32:10 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 14 Apr 2011 20:32:10 +0300
Subject: [Freeswitch-users] "Nets" Capability
In-Reply-To: <BANLkTi=gfkLcrJDny-_y2neo72_acPXgJA@mail.gmail.com>
References: <BANLkTi=gfkLcrJDny-_y2neo72_acPXgJA@mail.gmail.com>
Message-ID: <BANLkTims6GDaLL-UMacrVn+DJ=njffA1yQ@mail.gmail.com>

I've not used conferences terribly much, but I think this is how you'll have
to use it.

First: freeswitch has flags on the conferences - you can set members to be
deaf (can't hear), mute (can't speak) or both (among other things)
Second: I think you'll have to create your nets, and then.. each user will
need his own conference, otherwise, he could only be in one conference at a
time. Then, you would use some ESL-foo or something like "bind_digit_action"
to let them choose to join a net, and to modify the talking to each net
(e.g. by un/setting the deaf for each of the conferences). (You'd have to
bridge each conference into his own, too, so he could hear/speak). Actually,
I'm not really sure how you'd handle the speaking.. how to choose which to
speak to, and how to make sure he can hear, but they won't feed back into
the conference..

This sounds a bit messy, and I'd be interested if there was another way to
do it.

-Avi


On Thu, Apr 14, 2011 at 7:46 PM, mike petty <mikepetty at gmail.com> wrote:

> I have a quick question regarding freeswitch?s capabilities.  I?ve
> downloaded it and am diving into it right now.  One part of the
> functionality that I?m looking for is something like radio nets.
>
>
>
> In this scenario, there would be multiple ?nets? or conference room-like
> channels, and the user would essentially be in all of them, but be able to
> set talk/listen capabilities on each individually.
>
>
>
> For instance, you might have a ?Master Coordination Net? a ?Troubleshooting
> Net? and a ?Local Area Net?.
>
>
>
> The user would be in all of those, and set Listen on all, or some of the
> nets, but only talk on the ?Troubleshooting Net?, but then would like to
> talk on Master Coord and Local Area, so he would set talk on both of those
> and his voice would go out to just those listening on those two nets?  Most
> of the users would be SIP clients, but some would be coming in off of PSTN
> and other interfaces.
>
>
>
> It seems like this should be possible, but right now I?m sorting through
> all of the different modules and settings and everything, and am really
> starting to wonder if this is possible or not.
>
>
> Eventually, my plan would be to move the Talk/Listen status to external
> push button switches that I read in via some digital inputs, and then I
> would write a little program to set the talk/listen based upon the button
> status to provide a control panel type of interface for a user.
>
>
> Thanks.  Freeswitch looks amazing so far, and so much more intuitive than
> Asterisk!
>
>
>
> 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 cmrienzo at gmail.com  Thu Apr 14 21:32:12 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 14 Apr 2011 13:32:12 -0400
Subject: [Freeswitch-users] fax tone detection failed
In-Reply-To: <B11730398BF74A299031A8B958D497D3@hometarat>
References: <mailman.7319.1302674417.24555.freeswitch-users@lists.freeswitch.org>
	<3BE856AEC9884941B3F865AE687F573B@hometarat>
	<BANLkTi=nVRfGHVxe-W24BKvkhfDjj_DVhg@mail.gmail.com>
	<B11730398BF74A299031A8B958D497D3@hometarat>
Message-ID: <BANLkTim=AefBvyDnQcSVpKg0BAVfxifojQ@mail.gmail.com>

2100 Hz is the ANS tone from the receiving side of the fax call.  That's
what I detect on my outbound calls.


On Thu, Apr 14, 2011 at 12:57 PM, romon.zaman <romon.zaman at gmail.com> wrote:

>  hello MC,
> you are right..
> before this published here, you help me to get a solution.
>
> actually, my fax machine was not sending standard fax tone.  but it worked
> with actual tone frequency.
>
> so, if someone ever face this problem, recommendation is to get pcap file
> and extract audio file from RTP with wireshark and analyze audio with
> audicity to get exact frequency of tone.
>
> and when u use tone_detect with exact tone frequency, it will work
> perfectly.
>
>
> thanks for  your help
>
>
> ----- Original Message -----
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, April 14, 2011 10:43 PM
> *Subject:* Re: [Freeswitch-users] fax tone detection failed
>
>
>
> On Wed, Apr 13, 2011 at 5:16 AM, romon.zaman <romon.zaman at gmail.com>wrote:
>
>> hi,
>>
>> i am trying to detect fax tone on outgoing calls.
>> here is my javascript code
>>
>> +++++++++++++++++++++++
>>
>> session1 = new
>> Session("{ignore_early_media=false}sofia/gateway/flowroute/"+argv[0]+"");
>>
>> session1.execute("tone_detect", "fax 1100 r +30000 set
>> 'fax_tone_detected=true' 1");
>> msleep(argv[1]);
>> session1.hangup();
>>
>> ++++++++++++++++++++++++
>>
>> i  checked pcap file and  using wireshark , i can hear sone tone on rtp
>> audio playback.
>>
>> but freeswitch cant detect that..
>>
>> is there any other solution to find out specific tone like fax??
>>
>> thanks in advance
>>
>
> Was this the one we talked about yesterday where your fax machine was
> really spitting out a tone around 2096Hz?
>
> -MC
>
>
>
> __________ Information from ESET Smart Security, version of virus signature
> database 6041 (20110414) __________
>
>
> The message was checked by ESET Smart Security.
>
> http://www.eset.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 dave at dchorton.com  Thu Apr 14 21:36:56 2011
From: dave at dchorton.com (Dave Horton)
Date: Thu, 14 Apr 2011 13:36:56 -0400
Subject: [Freeswitch-users] Relay Proxy-Authentication requests to the
	caller
In-Reply-To: <mailman.7437.1302799201.24555.freeswitch-users@lists.freeswitch.org>
References: <mailman.7437.1302799201.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <4058C325-0AC0-4878-B86E-95EF11329086@dchorton.com>


I had the same question/issue myself, and would be interested to know if there is a way.  For myself, I had to do two things to get this to work:

1.  Disable authentication so that FS would not itself generate a challenge.  In vars.xml I added this:

 <X-PRE-PROCESS cmd="set" data="internal_auth_calls=false"/>

and in my internal.xml (because calls were coming in on my internal profile):

   <param name="accept-blind-auth" value="true"/>

2. Then I had to hack mod_sofia to create a new channel variable containing the received Proxy-Authorization on the inbound leg, and also let me set it on the outbound leg.

After that it worked, but I would still prefer a solution that didn't require changing mod_sofia, if one exists...

>>Hi All!
>> 
>>I would like to know if  there is a param we could set in sofia profile to let freeswitch relay Proxy-Authentication requests to the caller instead of absorbing it.
>> 
>>Regards,
>>Hamid
 

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From lfurrea at gmail.com  Thu Apr 14 21:38:50 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Thu, 14 Apr 2011 11:38:50 -0600
Subject: [Freeswitch-users] Using txfax from a LUA script
In-Reply-To: <BANLkTi=9xbaifeFgbpkXLw9+04d=V2=y-Q@mail.gmail.com>
References: <BANLkTimKsHW+JSi7jZ4ggUwpQEjy=2WoVQ@mail.gmail.com>
	<BANLkTi=9xbaifeFgbpkXLw9+04d=V2=y-Q@mail.gmail.com>
Message-ID: <BANLkTi=ZYOy+ykQDaYHHn0Rt-S00Z3p86g@mail.gmail.com>

I wouldn't know if it tries to run, it just sits there for a while until the
channel is hung with a timeout I assume then the script continues..

Like I can do

freeswitch.consoleLog("info", "Hello World 1! " .. "\n")
   session:execute("txfax", "/rxfax.tiff")
   freeswitch.consoleLog("info", "Hello World 2! " .. "\n")

and see Hello World 1 and Hello World 2

then hangs up.

Michael, when you say transfer to ext that does the bridge you mean
something like:

extension name="test_txfax_stream">
     <condition field="destination_number" expression="^\*90012$">
       <action application="txfax" data="txfax.tiff"/>
       ...
     </condition>
   </extension>

What exactly do I need to bridge to?

On Thu, Apr 14, 2011 at 11:20 AM, Michael Collins <msc at freeswitch.org>wrote:

> What happens when you call the txfax app from Lua? Does it even attempt to
> run? Just curious. In any case, I think you can do a combo of dp and Lua by
> transferring the call to an extension that does the bridge and if it fails
> you can check the failure cause in a Lua script and try again. You'll just
> need a chan var for the number of attempts.
>
> -MC
>
> On Thu, Apr 14, 2011 at 6:31 AM, Luis F Urrea <lfurrea at gmail.com> wrote:
>
>> Hi all,
>>
>> I noticed that I am not able to use TXFAX from a LUA script as follows:
>>
>> obSession = freeswitch.Session("freetdm/1/1/9999")
>>
>> if obSession:ready() then
>>     session:execute("txfax", "/rxfax.tiff")
>> else
>>
>> My objective was to send a fax to a fax machine connected to an FXS.
>>
>> I does work when I use:
>>
>> originate freetdm/1/1/9999 &txfax(/rxfax.tiff)
>>
>> But I wanted to be able to retry based on DISCONNECT CAUSES and the like.
>>
>>
>> Just for peace of mind I wanted to get some input from you on the causes
>> of this "limitation", and maybe get some suggestions as to any other
>> options?
>>
>>
>> TIA
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Apr 14 21:54:19 2011
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 15 Apr 2011 01:54:19 +0800
Subject: [Freeswitch-users] fax tone detection failed
In-Reply-To: <BANLkTi=nVRfGHVxe-W24BKvkhfDjj_DVhg@mail.gmail.com>
References: <mailman.7319.1302674417.24555.freeswitch-users@lists.freeswitch.org>	<3BE856AEC9884941B3F865AE687F573B@hometarat>
	<BANLkTi=nVRfGHVxe-W24BKvkhfDjj_DVhg@mail.gmail.com>
Message-ID: <4DA734CB.9040808@coppice.org>

On 04/15/2011 12:43 AM, Michael Collins wrote:
>
>
> On Wed, Apr 13, 2011 at 5:16 AM, romon.zaman <romon.zaman at gmail.com 
> <mailto:romon.zaman at gmail.com>> wrote:
>
>     hi,
>
>     i am trying to detect fax tone on outgoing calls.
>     here is my javascript code
>
>     +++++++++++++++++++++++
>
>     session1 = new
>     Session("{ignore_early_media=false}sofia/gateway/flowroute/"+argv[0]+"");
>
>     session1.execute("tone_detect", "fax 1100 r +30000 set
>     'fax_tone_detected=true' 1");
>     msleep(argv[1]);
>     session1.hangup();
>
>     ++++++++++++++++++++++++
>
>     i  checked pcap file and  using wireshark , i can hear sone tone
>     on rtp
>     audio playback.
>
>     but freeswitch cant detect that..
>
>     is there any other solution to find out specific tone like fax??
>
>     thanks in advance
>
>
> Was this the one we talked about yesterday where your fax machine was 
> really spitting out a tone around 2096Hz?
>
Of course it spits out around 2096Hz. An answering FAX machine sends a 
burst of 2100Hz tone, and then V.21 preamble. Its the calling machine 
which sends 1100Hz.

Steve



From steveu at coppice.org  Thu Apr 14 21:58:23 2011
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 15 Apr 2011 01:58:23 +0800
Subject: [Freeswitch-users] fax tone detection failed
In-Reply-To: <BANLkTim=AefBvyDnQcSVpKg0BAVfxifojQ@mail.gmail.com>
References: <mailman.7319.1302674417.24555.freeswitch-users@lists.freeswitch.org>	<3BE856AEC9884941B3F865AE687F573B@hometarat>	<BANLkTi=nVRfGHVxe-W24BKvkhfDjj_DVhg@mail.gmail.com>	<B11730398BF74A299031A8B958D497D3@hometarat>
	<BANLkTim=AefBvyDnQcSVpKg0BAVfxifojQ@mail.gmail.com>
Message-ID: <4DA735BF.5020507@coppice.org>

On 04/15/2011 01:32 AM, Christopher Rienzo wrote:
> 2100 Hz is the ANS tone from the receiving side of the fax call.  
> That's what I detect on my outbound calls.
An answering machine sends 2100Hz, but its very unreliable to rely on 
hearing it. In spandsp there is a module which does a much more thorough 
job of detecting a FAX machine. It will accept the V.21 preamble, which 
normally follows the 2100Hz tone, as a substitute for the 2100Hz itself. 
That way a call which starts passing audio a bit late (which they often 
do) still results in proper FAX detection. We need to expose that 
functionality in FreeSwitch, but have not done so yet.

Steve



From rogelio.perez at gmail.com  Thu Apr 14 22:02:49 2011
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Thu, 14 Apr 2011 15:02:49 -0300
Subject: [Freeswitch-users] Call state notifications
In-Reply-To: <BANLkTikgTPekcDf3LtaN8DRPfoDq99dGNg@mail.gmail.com>
References: <C8A89431-F467-43B0-8D77-480DD2A1F4B4@gmail.com>
	<BANLkTikgTPekcDf3LtaN8DRPfoDq99dGNg@mail.gmail.com>
Message-ID: <FBFBC0A6-7DDE-4F36-BD03-7D1D0559AC25@gmail.com>

Hi Michael,

This is exactly what I needed.
I assumed that outbound was the way to go because FS will push the notifications to me, but now I see that inbound is more like subscribing to the events notifications.
I have enough info to continue playing, thanks!

Rogelio


On Apr 14, 2011, at 2:12 PM, Michael Collins wrote:

> Rogelio,
> 
> Are you trying to make a single web page that shows the status of multiple calls? If so then you probably want what we call "inbound" event socket versus the "outbound" event socket that you described below. (Inbound and outbound are from the perspective of FreeSWITCH, so when you have a dialplan with the socket app it makes an "outbound" socket connection to your program that is listening on the IP/port...)
> 
> If you want to sit and watch the event socket and see all the channel state change events then I recommend a few things:
> 
> #1 - read up on the event socket on the wiki, especially the event socket commands like "filter"
> #2 - get a feel for what events come down the pipe by using fs_cli:
>   open fs_cli
>   type "/log 0" to turn off logging
>   type "/events plain all" to turn on events
> 
> Just sit and watch the console. You'll be amazed (and overwhelmed) at all of the events and information that come to you.
> 
> #3 - See if you can set a filter to get only the events you want. While still at the fs_cli and while watching all the events come to you, try typing this: "/filter Event-Name CHANNEL_STATE_CHANGE"
> 
> That will filter out everything *except* channel state changes. Then make a call and watch the screen. Do stuff like transfers and hangups, etc. and see what the events look like. 
> 
> If that all seems like too much then you can cheat and just send a "show channels" every 2 seconds and parse the results. ;)
> 
> Have fun!
> -MC
> 
> On Wed, Apr 13, 2011 at 8:30 PM, Rogelio Perez <rogelio.perez at gmail.com> wrote:
> > Event sockets is what you want....
> 
> Thanks Ken, it looks like Event Sockets will do the job, but I'm not sure how.
> I'm testing with netcat listening on port 8084, and then I've managed to connect to it by inserting this line on the dialplan:
> 
> <action application="socket" data="127.0.0.1:8084 async full"/>
> 
> Since I dont need to control the call but only receive notification events about the channel state changes I assume I have to send the command "myevents\n\n\" and the pass the variable socket_resume:true back, but this doesnt seem to be working.
> 
> sendmsg 
> call-command: execute
> execute-app-name: myevents\n\n\
> socket_resume:true
> 
> This is my first attempt to write an application for FS so I need some guidance.
> 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 kris at kriskinc.com  Thu Apr 14 22:07:35 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 14 Apr 2011 14:07:35 -0400
Subject: [Freeswitch-users] fax tone detection failed
In-Reply-To: <4DA735BF.5020507@coppice.org>
References: <mailman.7319.1302674417.24555.freeswitch-users@lists.freeswitch.org>
	<3BE856AEC9884941B3F865AE687F573B@hometarat>
	<BANLkTi=nVRfGHVxe-W24BKvkhfDjj_DVhg@mail.gmail.com>
	<B11730398BF74A299031A8B958D497D3@hometarat>
	<BANLkTim=AefBvyDnQcSVpKg0BAVfxifojQ@mail.gmail.com>
	<4DA735BF.5020507@coppice.org>
Message-ID: <BANLkTi=U5bLuwrVeOj+axtc9xdh3yEVmEA@mail.gmail.com>

This would be awesome because (as you say) you can't count on the
receiving machine sending the 2100Hz tone...

On Thu, Apr 14, 2011 at 1:58 PM, Steve Underwood <steveu at coppice.org> wrote:
> On 04/15/2011 01:32 AM, Christopher Rienzo wrote:
>> 2100 Hz is the ANS tone from the receiving side of the fax call.
>> That's what I detect on my outbound calls.
> An answering machine sends 2100Hz, but its very unreliable to rely on
> hearing it. In spandsp there is a module which does a much more thorough
> job of detecting a FAX machine. It will accept the V.21 preamble, which
> normally follows the 2100Hz tone, as a substitute for the 2100Hz itself.
> That way a call which starts passing audio a bit late (which they often
> do) still results in proper FAX detection. We need to expose that
> functionality in FreeSwitch, but have not done so yet.
>
> Steve

-- 
Kristian Kielhofner


From kris at kriskinc.com  Thu Apr 14 22:11:59 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 14 Apr 2011 14:11:59 -0400
Subject: [Freeswitch-users] Relay Proxy-Authentication requests to the
	caller
In-Reply-To: <4058C325-0AC0-4878-B86E-95EF11329086@dchorton.com>
References: <mailman.7437.1302799201.24555.freeswitch-users@lists.freeswitch.org>
	<4058C325-0AC0-4878-B86E-95EF11329086@dchorton.com>
Message-ID: <BANLkTinDXVz=c2BXoR37wUbYx3MNPU+mvw@mail.gmail.com>

These requests have baffled me...

Are you trying to use FreeSWITCH as an outbound proxy (or something)?

This type of low-level SIP mangling is usually more well suited to
OpenSIPS, Kamailio, etc.  Even then you'd have some work cut out for
you...

On Thu, Apr 14, 2011 at 1:36 PM, Dave Horton <dave at dchorton.com> wrote:
>
> I had the same question/issue myself, and would be interested to know if
> there is a way. ?For myself, I had to do two things to get this to work:
> 1. ?Disable authentication so that FS would not itself generate a challenge.
> ?In vars.xml I added this:
> ?<X-PRE-PROCESS cmd="set" data="internal_auth_calls=false"/>
> and in my internal.xml (because calls were coming in on my internal
> profile):
> ?? <param name="accept-blind-auth" value="true"/>
> 2. Then I had to hack mod_sofia to create a new channel variable containing
> the received Proxy-Authorization on the inbound leg, and also let me set it
> on the outbound leg.
> After that it worked, but I would still prefer a solution that didn't
> require changing mod_sofia, if one exists...
>
>>>Hi All!
>
>>>
>
>>>I would like to know if ?there is a param we could set in sofia profile to
>>> let freeswitch relay Proxy-Authentication requests to the caller instead of
>>> absorbing it.
>
>>>
>
>>>Regards,
>
>>>Hamid
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 dave at dchorton.com  Thu Apr 14 22:18:25 2011
From: dave at dchorton.com (Dave Horton)
Date: Thu, 14 Apr 2011 14:18:25 -0400
Subject: [Freeswitch-users] Relay Proxy-Authentication requests to the
	caller
In-Reply-To: <BANLkTinDXVz=c2BXoR37wUbYx3MNPU+mvw@mail.gmail.com>
References: <mailman.7437.1302799201.24555.freeswitch-users@lists.freeswitch.org>
	<4058C325-0AC0-4878-B86E-95EF11329086@dchorton.com>
	<BANLkTinDXVz=c2BXoR37wUbYx3MNPU+mvw@mail.gmail.com>
Message-ID: <8207E7C4-ECD5-415D-BA16-5AA9AC843C89@dchorton.com>


In my case, I am trying to use it as a simple transcoding server.  And the rich media support of codecs, and the ease of managing multiple sip profiles and setting up a back-to-back user agent scenario lend themselves quite well to the task IMO.  I don't think the other solutions handle media or transcoding at all (correct me if I'm wrong) so trying to build the solution using those platforms would actually leave quite a lot of heavy lifting to do.  

I have built a lot of SIP apps over the past 10 years, on a lot of different networks, and I have found that it is critical to be able to tweak SIP headers to make these interop.  I wish it weren't the case, but it has been and still remains so.  FS seems to only allow true custom headers (e.g., X- headers) to be manipulated at this level, and I find it to be a shortcoming.


On Apr 14, 2011, at 2:11 PM, Kristian Kielhofner wrote:

These requests have baffled me...

Are you trying to use FreeSWITCH as an outbound proxy (or something)?

This type of low-level SIP mangling is usually more well suited to
OpenSIPS, Kamailio, etc.  Even then you'd have some work cut out for
you...

On Thu, Apr 14, 2011 at 1:36 PM, Dave Horton <dave at dchorton.com> wrote:
> 
> I had the same question/issue myself, and would be interested to know if
> there is a way.  For myself, I had to do two things to get this to work:
> 1.  Disable authentication so that FS would not itself generate a challenge.
>  In vars.xml I added this:
>  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=false"/>
> and in my internal.xml (because calls were coming in on my internal
> profile):
>    <param name="accept-blind-auth" value="true"/>
> 2. Then I had to hack mod_sofia to create a new channel variable containing
> the received Proxy-Authorization on the inbound leg, and also let me set it
> on the outbound leg.
> After that it worked, but I would still prefer a solution that didn't
> require changing mod_sofia, if one exists...
> 
>>> Hi All!
> 
>>> 
> 
>>> I would like to know if  there is a param we could set in sofia profile to
>>> let freeswitch relay Proxy-Authentication requests to the caller instead of
>>> absorbing it.
> 
>>> 
> 
>>> Regards,
> 
>>> Hamid
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 cmrienzo at gmail.com  Thu Apr 14 22:37:58 2011
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Thu, 14 Apr 2011 14:37:58 -0400
Subject: [Freeswitch-users] fax tone detection failed
In-Reply-To: <4DA735BF.5020507@coppice.org>
References: <mailman.7319.1302674417.24555.freeswitch-users@lists.freeswitch.org>
	<3BE856AEC9884941B3F865AE687F573B@hometarat>
	<BANLkTi=nVRfGHVxe-W24BKvkhfDjj_DVhg@mail.gmail.com>
	<B11730398BF74A299031A8B958D497D3@hometarat>
	<BANLkTim=AefBvyDnQcSVpKg0BAVfxifojQ@mail.gmail.com>
	<4DA735BF.5020507@coppice.org>
Message-ID: <BANLkTi=BO5LKTC3+fu=dDfgKbDNnCkz7oA@mail.gmail.com>

Looks like modem_connect_tones_rx is the module... it's simple to plug that
into a media bug.


On Thu, Apr 14, 2011 at 1:58 PM, Steve Underwood <steveu at coppice.org> wrote:

> On 04/15/2011 01:32 AM, Christopher Rienzo wrote:
> > 2100 Hz is the ANS tone from the receiving side of the fax call.
> > That's what I detect on my outbound calls.
> An answering machine sends 2100Hz, but its very unreliable to rely on
> hearing it. In spandsp there is a module which does a much more thorough
> job of detecting a FAX machine. It will accept the V.21 preamble, which
> normally follows the 2100Hz tone, as a substitute for the 2100Hz itself.
> That way a call which starts passing audio a bit late (which they often
> do) still results in proper FAX detection. We need to expose that
> functionality in FreeSwitch, but have not done so yet.
>
> 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 infos at madovsky.org  Thu Apr 14 22:44:34 2011
From: infos at madovsky.org (Madovsky)
Date: Thu, 14 Apr 2011 14:44:34 -0400
Subject: [Freeswitch-users] Relay Proxy-Authentication requests to
	thecaller
References: <mailman.7437.1302799201.24555.freeswitch-users@lists.freeswitch.org><4058C325-0AC0-4878-B86E-95EF11329086@dchorton.com><BANLkTinDXVz=c2BXoR37wUbYx3MNPU+mvw@mail.gmail.com>
	<8207E7C4-ECD5-415D-BA16-5AA9AC843C89@dchorton.com>
Message-ID: <57D4E617638C42E78E9795F27CAC9C07@e1705>

Dave,

you're wrong to use FS as transcoder only
as Kristian, I and other said
real SIP proxy servers as Opensips, Kamalio are built to use it as you need


----- Original Message ----- 
From: "Dave Horton" <dave at dchorton.com>
To: "Kristian Kielhofner" <kris at kriskinc.com>
Cc: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Thursday, April 14, 2011 2:18 PM
Subject: Re: [Freeswitch-users] Relay Proxy-Authentication requests to 
thecaller


>
> In my case, I am trying to use it as a simple transcoding server.  And the 
> rich media support of codecs, and the ease of managing multiple sip 
> profiles and setting up a back-to-back user agent scenario lend themselves 
> quite well to the task IMO.  I don't think the other solutions handle 
> media or transcoding at all (correct me if I'm wrong) so trying to build 
> the solution using those platforms would actually leave quite a lot of 
> heavy lifting to do.
>
> I have built a lot of SIP apps over the past 10 years, on a lot of 
> different networks, and I have found that it is critical to be able to 
> tweak SIP headers to make these interop.  I wish it weren't the case, but 
> it has been and still remains so.  FS seems to only allow true custom 
> headers (e.g., X- headers) to be manipulated at this level, and I find it 
> to be a shortcoming.
>
>
> On Apr 14, 2011, at 2:11 PM, Kristian Kielhofner wrote:
>
> These requests have baffled me...
>
> Are you trying to use FreeSWITCH as an outbound proxy (or something)?
>
> This type of low-level SIP mangling is usually more well suited to
> OpenSIPS, Kamailio, etc.  Even then you'd have some work cut out for
> you...
>
> On Thu, Apr 14, 2011 at 1:36 PM, Dave Horton <dave at dchorton.com> wrote:
>>
>> I had the same question/issue myself, and would be interested to know if
>> there is a way.  For myself, I had to do two things to get this to work:
>> 1.  Disable authentication so that FS would not itself generate a 
>> challenge.
>>  In vars.xml I added this:
>>  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=false"/>
>> and in my internal.xml (because calls were coming in on my internal
>> profile):
>>    <param name="accept-blind-auth" value="true"/>
>> 2. Then I had to hack mod_sofia to create a new channel variable 
>> containing
>> the received Proxy-Authorization on the inbound leg, and also let me set 
>> it
>> on the outbound leg.
>> After that it worked, but I would still prefer a solution that didn't
>> require changing mod_sofia, if one exists...
>>
>>>> Hi All!
>>
>>>>
>>
>>>> I would like to know if  there is a param we could set in sofia profile 
>>>> to
>>>> let freeswitch relay Proxy-Authentication requests to the caller 
>>>> instead of
>>>> absorbing it.
>>
>>>>
>>
>>>> Regards,
>>
>>>> Hamid
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 



From kris at kriskinc.com  Thu Apr 14 22:47:12 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 14 Apr 2011 14:47:12 -0400
Subject: [Freeswitch-users] Relay Proxy-Authentication requests to the
	caller
In-Reply-To: <8207E7C4-ECD5-415D-BA16-5AA9AC843C89@dchorton.com>
References: <mailman.7437.1302799201.24555.freeswitch-users@lists.freeswitch.org>
	<4058C325-0AC0-4878-B86E-95EF11329086@dchorton.com>
	<BANLkTinDXVz=c2BXoR37wUbYx3MNPU+mvw@mail.gmail.com>
	<8207E7C4-ECD5-415D-BA16-5AA9AC843C89@dchorton.com>
Message-ID: <BANLkTim9Z6Smc8NPK82AOLJ-TShwcsH8JQ@mail.gmail.com>

Kamailio is making strides in handling media and transcoding but
practically speaking that's still some ways away.

It is true that "tweaking SIP headers" for interop is very much a fact
of life.  URI formats, URI parameters, caller id presentation types,
etc, etc.  However, at the point you're asking a B2BUA to pass a
Proxy-Auth header from one leg to the next I think you should rethink
your overall design and architecture...

Of course all of these projects are open source and you're free to
tweak/modify/mangle/break them any way you see fit and that's the
beauty of open source.  You just can't expect the developers that have
to support this code for eternity to accept a patch that so clearly
goes against the fundamental design of the software.

FreeSWITCH is a lot of things but when it comes to bridging calls it's
*always* a SIP B2BUA and that comes along with a fairly well (for SIP)
defined definition for how multiple legs are handled.  This is exactly
why the descriptive term for Kamailio has moved from "sip proxy" to
"sip server" over the years.

On Thu, Apr 14, 2011 at 2:18 PM, Dave Horton <dave at dchorton.com> wrote:
>
> In my case, I am trying to use it as a simple transcoding server. ?And the rich media support of codecs, and the ease of managing multiple sip profiles and setting up a back-to-back user agent scenario lend themselves quite well to the task IMO. ?I don't think the other solutions handle media or transcoding at all (correct me if I'm wrong) so trying to build the solution using those platforms would actually leave quite a lot of heavy lifting to do.
>
> I have built a lot of SIP apps over the past 10 years, on a lot of different networks, and I have found that it is critical to be able to tweak SIP headers to make these interop. ?I wish it weren't the case, but it has been and still remains so. ?FS seems to only allow true custom headers (e.g., X- headers) to be manipulated at this level, and I find it to be a shortcoming.
>

-- 
Kristian Kielhofner


From dave at dchorton.com  Thu Apr 14 23:23:54 2011
From: dave at dchorton.com (Dave Horton)
Date: Thu, 14 Apr 2011 15:23:54 -0400
Subject: [Freeswitch-users] Relay Proxy-Authentication requests to the
	caller
In-Reply-To: <BANLkTim9Z6Smc8NPK82AOLJ-TShwcsH8JQ@mail.gmail.com>
References: <mailman.7437.1302799201.24555.freeswitch-users@lists.freeswitch.org>
	<4058C325-0AC0-4878-B86E-95EF11329086@dchorton.com>
	<BANLkTinDXVz=c2BXoR37wUbYx3MNPU+mvw@mail.gmail.com>
	<8207E7C4-ECD5-415D-BA16-5AA9AC843C89@dchorton.com>
	<BANLkTim9Z6Smc8NPK82AOLJ-TShwcsH8JQ@mail.gmail.com>
Message-ID: <9BCB4EA5-C859-42F0-80BB-E77DF2E76B8C@dchorton.com>

Hi Kristian - 

I appreciate your thoughts/feedback/guidance, as well as the manner in which they were delivered.  
At the risk of prolonging a thread that perhaps has reached the point of maximal usefulness, I'd like to 
respond briefly to a few points:

1) I don't intend to submit a patch, because I agree it seems to go against what FS wants to do -- 
at least, as best I can understand what FS wants to do since as a newbie I'm sort of puzzling that out as I go
by means of this forum, reading the book, etc.
However, FS seems to somewhat have the burden of success, in that "what it wants to do" and 
"what it can do" seem to be overlapping but not exactly equal sets (the second being larger) -- 
due to its flexibility it seems that FS can do quite a bit, as in my case of needing to whip up
a simple transcoding server.

2) I don't think there is much to re-think about the overall design and architecture in my case, 
simply because I need a transcoding server to sit a certain point in a service provider's network
and that's that.  In my case, I have multiple remote FS PBXs around the globe transmitting
to a gateway in a service provider in the US for PSTN access, over a bandwidth constrained
satellite link.  Therefore speex 8ks make sense over the satellite link, but then I need to 
transcode back to PCMU once in the service providers network.  Authentication is unfortunately
inextricably tied up with establishing the media path since credentials are carried in the INVITE.

3) I am personally mostly interested in building applications that run in a service provider core network, not out on the edge.
I'd be interested to know if this is considered in the target of what FS wants to be?  Obviously, 
FS is first and foremost a PBX, but of course its not limited to such.  Asterisk, as an example, was 
adopted by some service providers for various "application server" types of applications, 
sometimes with success, sometimes without.

4) In service provider networks, almost all "interesting" applications are B2BUA.
And I would disagree that SIP has a well-defined set of constraints on what
a B2BUA can and should be.  To the contrary, I think its pretty wide open
(at least by comparison to specifications for proxies, e.g.)

5) This question of handling (or not) SIP headers doesn't seem to be 
documented consistently.  In the FS book, there is a note that suggests 
you *should* only mess with X- headers, but implies you are not limited
to doing so.  In fact, I see no way to get at the other headers (other than
the obvious suspects -- to, from, contact, etc -- and one or two special cases
-- alert-info -- that seemed to have been plucked out over time).  It 
just doesn't seem like it would be all that difficult to provide access
to all the sip headers for developers that might need them).

Finally, as a newbie to FS, but with some experience building SIP platforms
and applications for a bit, I would be remiss in not congratulating the developers
on the great product they have created.  Well done!

On Apr 14, 2011, at 2:47 PM, Kristian Kielhofner wrote:

Kamailio is making strides in handling media and transcoding but
practically speaking that's still some ways away.

It is true that "tweaking SIP headers" for interop is very much a fact
of life.  URI formats, URI parameters, caller id presentation types,
etc, etc.  However, at the point you're asking a B2BUA to pass a
Proxy-Auth header from one leg to the next I think you should rethink
your overall design and architecture...

Of course all of these projects are open source and you're free to
tweak/modify/mangle/break them any way you see fit and that's the
beauty of open source.  You just can't expect the developers that have
to support this code for eternity to accept a patch that so clearly
goes against the fundamental design of the software.

FreeSWITCH is a lot of things but when it comes to bridging calls it's
*always* a SIP B2BUA and that comes along with a fairly well (for SIP)
defined definition for how multiple legs are handled.  This is exactly
why the descriptive term for Kamailio has moved from "sip proxy" to
"sip server" over the years.

On Thu, Apr 14, 2011 at 2:18 PM, Dave Horton <dave at dchorton.com> wrote:
> 
> In my case, I am trying to use it as a simple transcoding server.  And the rich media support of codecs, and the ease of managing multiple sip profiles and setting up a back-to-back user agent scenario lend themselves quite well to the task IMO.  I don't think the other solutions handle media or transcoding at all (correct me if I'm wrong) so trying to build the solution using those platforms would actually leave quite a lot of heavy lifting to do.
> 
> I have built a lot of SIP apps over the past 10 years, on a lot of different networks, and I have found that it is critical to be able to tweak SIP headers to make these interop.  I wish it weren't the case, but it has been and still remains so.  FS seems to only allow true custom headers (e.g., X- headers) to be manipulated at this level, and I find it to be a shortcoming.
> 

-- 
Kristian Kielhofner




From philipp.kempgen at amooma.de  Thu Apr 14 23:15:24 2011
From: philipp.kempgen at amooma.de (Philipp Kempgen)
Date: Thu, 14 Apr 2011 21:15:24 +0200
Subject: [Freeswitch-users] mod_xml_curl,
	sofia_reg.c:1872 Can't find user ...
Message-ID: <4DA747CC.3080509@amooma.de>

Hi,

I'm playing with mod_xml_curl per
http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22
but having some problems.

Softphone (username "11") sends a REGISTER to FreeSwitch (1.0.6)
at 192.168.65.133.

xml_curl.conf.xml:
---------------------------------------------------------------
  <binding name="users directory">
    <param name="gateway-url" value="http://127.0.0.1:3000/freeswitch-directory-entries/search.xml" bindings="directory" />
  </binding>
---------------------------------------------------------------

FS sends a lookup request to the web service. Parameters:
---------------------------------------------------------------
section=directory
...
key=id
user=11
domain=192.168.65.133
---------------------------------------------------------------

Fine.

The response is:
---------------------------------------------------------------
<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<document type="freeswitch/xml">
  <section name="directory">
    <domain name="192.168.65.133">
      <users>
        <user id="11" number-alias="1111">
          <params>
            <param name="dial-string" value="${sofia_contact(11 at 192.168.65.133)}" />
            <param name="password" value="123" />
            <param name="vm-password" value="417906717460606223" />
          </params>
          <variables>
            <variable name="accountcode" value="11" />
            <variable name="effective_caller_id_number" value="foo" />
            <variable name="effective_caller_id_name" value="Foo, Foo" />
            <variable name="outbound_caller_id_number" value="foo" />
            <variable name="outbound_caller_id_name" value="Foo, Foo" />
            <variable name="user_context" value="default" />
            <variable name="toll_allow" value="domestic,international,local" />
            <variable name="callgroup" value="users" />
          </variables>
        </user>
      </users>
    </domain>
  </section>
</document>
---------------------------------------------------------------

But I get a complaint on the console:
[WARNING] sofia_reg.c:1872 Can't find user [11 at 192.168.65.133]
You must define a domain called '192.168.65.133' in your directory
and add a user with the id="11" attribute and you must configure
your device to use the proper domain in it's authentication
credentials.

What can it be?

A domain with name="192.168.65.133" and a user with id="11" is in
the response, so why doesn't sofia find it?


    Philipp



From msc at freeswitch.org  Thu Apr 14 23:46:46 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Apr 2011 12:46:46 -0700
Subject: [Freeswitch-users] mod_event_socket Whether can get freeswitch
 error messages
In-Reply-To: <8slc8n$of1hij@irxd5-201.sinamail.sina.com.cn>
References: <8slc8n$of1hij@irxd5-201.sinamail.sina.com.cn>
Message-ID: <BANLkTinrcHwnEUbQ_DAcaf4XOuY6hWun0A@mail.gmail.com>

I am not sure if I understand your question. What kinds of error messages
are you hoping to see? Also, if you are more comfortable you can try asking
your question in your native language (Mandarin?) as we might have some
people who are fluent and can help.

-MC

2011/4/13 ?? <sc_zhangming at sina.com>

> freeswitch-users?
>
>                  1. mod_event_socket Whether can get freeswitch error
> messages
>                          2. how to get
>
> ?
> ??
>
>
> sc_zhangming at sina.com
> 2011-04-14
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 15 00:29:08 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 14 Apr 2011 13:29:08 -0700
Subject: [Freeswitch-users] Relay Proxy-Authentication requests to the
	caller
In-Reply-To: <9BCB4EA5-C859-42F0-80BB-E77DF2E76B8C@dchorton.com>
References: <mailman.7437.1302799201.24555.freeswitch-users@lists.freeswitch.org>
	<4058C325-0AC0-4878-B86E-95EF11329086@dchorton.com>
	<BANLkTinDXVz=c2BXoR37wUbYx3MNPU+mvw@mail.gmail.com>
	<8207E7C4-ECD5-415D-BA16-5AA9AC843C89@dchorton.com>
	<BANLkTim9Z6Smc8NPK82AOLJ-TShwcsH8JQ@mail.gmail.com>
	<9BCB4EA5-C859-42F0-80BB-E77DF2E76B8C@dchorton.com>
Message-ID: <BANLkTi=HgajucWHeiQRAbqHVtqLUofpXbA@mail.gmail.com>

Dave,

Welcome to the FS community! Thanks for your input. We always appreciate it
when experienced newbies come in and start using the software. Thanks for
the feedback. I think you'll find that everyone here listens and responds to
feedback respectfully, whether they agree or disagree. Oh, and if someone is
blunt or rude please don't take it personally. Everybody has a bad day now
and then. :)

Now, on to your main issue: tweaking SIP headers and similar actions. I can
hear Tony (main dev) cringe when he sees that phrase. If you've been doing
SIP for 10 years then you know better than most just how crazy the RFCs are
and how loosely some people interpret them. That's one of the core reasons
why interop is such a challenge. We have a bit of a mantra around here, and
I suppose it goes something like this: we don't pander to people who do
stupid things with SIP just for the sake of interop. It's also been put this
way: just say "no". (Tip 'o the hat to Nancy Reagan ;)

The FS devs have no desire to dig one character deeper into the SIP stack
than they absolutely have to. It may seem that you are asking for a simple
feature. However, allowing FreeSWITCH (a B2BUA) to "tweak SIP headers" is
ridiculously complicated. If you don't believe me then go look at
mod_sofia.c and sofia.c and sofia_glue.c and sofia_reg.c and
sofia_presence.c and sofia_sla.c. That's just so FreeSWITCH can talk to the
SIP stack! libsofia is another monster. So the bottom line from the
development viewpoint is that we have to say "no" to some things. I know
that may seem arbitrary and maybe even capricious but I can assure you that
it is not. One of the reasons that FreeSWITCH is so stable and efficient is
that it doesn't not try to be "all things SIP" to everyone. I hope you can
see our viewpoint.

In your specific use case it sounds like you really do need a true proxy
server in addition to a media server. Others may have some thoughts. Are
there any other OSS communications software packages that can do all this,
i.e. be a SIP proxy and media server? (And do it reliably?)

-MC
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From steveayre at gmail.com  Fri Apr 15 00:46:07 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 14 Apr 2011 21:46:07 +0100
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <07CC1DBD9F314CABB6E6735822053C55@e1705>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
	<1B19ABD72889C245AE8EEE08AC24103A28C423231C@exmachina.office.kapper.net>
	<BANLkTimrvA=Lgva-FAdXxWRhS-_UDuTgTA@mail.gmail.com>
	<1BDDB8AD-4CBD-4515-A7AD-693A5E875523@freeswitch.org>
	<BANLkTi=YM4x3O6Xc-JDtXVKymKdVbcmP4Q@mail.gmail.com>
	<86CDDEC506B5411AA8227253257519A6@e1705>
	<BANLkTiksX56jd7zxWjrS1gwWc_J9_QT4nw@mail.gmail.com>
	<07CC1DBD9F314CABB6E6735822053C55@e1705>
Message-ID: <BANLkTi=Ras3d=qBqD3VvYnQnRRsPY-SD9w@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Proxy_Media#Common_misconceptions_.28READ_THIS.29
http://wiki.freeswitch.org/wiki/Proxy_Media#Why_you_almost_certainly_don.27t_want_to_use_it


On 14 April 2011 17:01, Madovsky <infos at madovsky.org> wrote:

>  ok, so maybe remove the word "proxy" that confused a lot of people here
> (even me at the start) :)
>
> ----- Original Message -----
> *From:* Steven Ayre <steveayre at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Thursday, April 14, 2011 2:45 AM
> *Subject:* Re: [Freeswitch-users] proxy SDP
>
> You misunderstand I think... it *isn't* needed for T38 any longer. Only in
> very old versions.
>
> -Steve
>
>
> On 13 April 2011 23:17, Madovsky <infos at madovsky.org> wrote:
>
>> maybe change the param name to proxy_media_t38 ;)
>>
>> ----- Original Message -----
>> From: "Kristian Kielhofner" <kris at kriskinc.com>
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Sent: Wednesday, April 13, 2011 2:10 PM
>> Subject: Re: [Freeswitch-users] proxy SDP
>>
>>
>> Brian,
>>
>>  For all of the confusion proxy media creates I still see cases where
>> it is useful...  It shouldn't be removed completely.
>>
>> On Tue, Apr 12, 2011 at 8:55 PM, Brian West <brian at freeswitch.org> wrote:
>> > We aren't a proxy... we have transcended into this quasi proxy in some
>> > scenarios which mostly involve t.38... as for proxy media DO NOT USE IT.
>> > Just saying it might go away since the purpose of it is now not needed
>> > since we have full t.38.
>> >
>> > Thanks,
>> > Brian
>> >
>>
>> --
>> 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
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>  ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Apr 15 00:55:39 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 14 Apr 2011 21:55:39 +0100
Subject: [Freeswitch-users] FreeSwitch Debug Full Log(version 1.0.6)
In-Reply-To: <4DA663CB.2040107@163.com>
References: <4DA663CB.2040107@163.com>
Message-ID: <BANLkTinFvuXijBH1di=fuRu2KJVDetuSzA@mail.gmail.com>

>From fs_cli? You can't.

-Steve

2011/4/14 vivid <vivid333 at 163.com>

>  how can I use FreeSwitch to trace *the Function call Procedure* in *
> fs_cli*.
> (using Command? configuration?)
>
>
> [DEBUG] *switch_ivr_bridge.c*:911 switch_ivr_multi_threaded_bridge()
> sofia/external/1XXX4951027 receive message [BRIDGE]
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Apr 15 00:57:47 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 14 Apr 2011 21:57:47 +0100
Subject: [Freeswitch-users] mod_event_socket Whether can get freeswitch
 error messages
In-Reply-To: <BANLkTinrcHwnEUbQ_DAcaf4XOuY6hWun0A@mail.gmail.com>
References: <8slc8n$of1hij@irxd5-201.sinamail.sina.com.cn>
	<BANLkTinrcHwnEUbQ_DAcaf4XOuY6hWun0A@mail.gmail.com>
Message-ID: <BANLkTi=GhdNqDgoV6vMQCmpfK1EhQ68ZRQ@mail.gmail.com>

I think he means the log messages...

fs_cli has code in it that gets those - perhaps you can use that as a
sample.

-Steve


2011/4/14 Michael Collins <msc at freeswitch.org>

> I am not sure if I understand your question. What kinds of error messages
> are you hoping to see? Also, if you are more comfortable you can try asking
> your question in your native language (Mandarin?) as we might have some
> people who are fluent and can help.
>
> -MC
>
>
> 2011/4/13 ?? <sc_zhangming at sina.com>
>
>> freeswitch-users?
>>
>>                  1. mod_event_socket Whether can get freeswitch error
>> messages
>>                          2. how to get
>>
>> ?
>> ??
>>
>>
>> sc_zhangming at sina.com
>> 2011-04-14
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Apr 15 01:01:30 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 14 Apr 2011 22:01:30 +0100
Subject: [Freeswitch-users] FreeSwitch Debug Full Log(version 1.0.6)
In-Reply-To: <BANLkTinFvuXijBH1di=fuRu2KJVDetuSzA@mail.gmail.com>
References: <4DA663CB.2040107@163.com>
	<BANLkTinFvuXijBH1di=fuRu2KJVDetuSzA@mail.gmail.com>
Message-ID: <BANLkTimYQQDR08na1SrAb4SUZYGbWpz3dA@mail.gmail.com>

Think about it a little...
fs_cli is the client to FS. Anything send to a client causes function calls.
Logging those would cause more function calls... which'd cause more. You'd
either enter an infinite loop or be unable to log some function calls.

You're better off running FreeSWITCH with a tool like gdb or gprof from the
local machine. Note that doing so will mean you're running slower.

-Steve

On 14 April 2011 21:55, Steven Ayre <steveayre at gmail.com> wrote:

> From fs_cli? You can't.
>
> -Steve
>
> 2011/4/14 vivid <vivid333 at 163.com>
>
>>  how can I use FreeSwitch to trace *the Function call Procedure* in *
>> fs_cli*.
>> (using Command? configuration?)
>>
>>
>> [DEBUG] *switch_ivr_bridge.c*:911 switch_ivr_multi_threaded_bridge()
>> sofia/external/1XXX4951027 receive message [BRIDGE]
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 nick.rosier at gmail.com  Fri Apr 15 01:39:25 2011
From: nick.rosier at gmail.com (Nick Rosier)
Date: Thu, 14 Apr 2011 23:39:25 +0200
Subject: [Freeswitch-users] Gateway with dynamic IP address
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDC7@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>
	<BANLkTi=4wX_cc9hPU0S1-O0UT0RE_Tom0Q@mail.gmail.com>
	<BANLkTi=8LfL-CMkczXgJg8FOF1_FTYtxdQ@mail.gmail.com>
	<BANLkTims24tpVuEmq1r84XA1bZd02ebfsw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDC7@cooper>
Message-ID: <BANLkTikzmRaCAq1eXbG-J0SEhacW1ZFYzg@mail.gmail.com>

Thanks, this was a step in the right direction :-)
Outbound dialing is working but I don't have a ringtone during dialing
and when I answer the phone I don't hear anything. This might be due
to the NAT'ed SPA3102.

2011-04-14 23:35:53.986767 [DEBUG] sofia_glue.c:2971 AUDIO RTP
[sofia/sipinterface_1/sip:trunk at 192.168.1.153:5063] xxx.xxx.xxx.xxx
port 25902 -> 192.168.1.153 port 16426 codec: 0 ms: 20

Any idea how to fix that?

On 12 April 2011 22:55, Peter Olsson <peter.olsson at visionutveckling.se> wrote:
> So, it's a user (from FS point of view).. A gateway (from somewhere) registers to your FS instance. Ypu want to dial to this gateway, just dial to it as a normal user. But just try to override the invite uri;
>
> {sip_invite_to_uri=<sip:REAL_NUMBER@${domain_name}>}user/registered_user@${domain_name}
>
> I just wrote this from my head, so it might have errors, but the big idea is to dial the user, but at the same time override the sip invite uri, to a different number.
>
> At least this is what I do in this situation.. :)
>
> /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 12 april 2011 20:26
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] Gateway with dynamic IP address
>
> The problem is this isn't a user, it's a gateway that registers itself
> as a user (so FS does not register to it but the other way around). I
> need to dial to it as a user and specify the number it has to dial.
>
> On 12 April 2011 12:20, Steven Ayre <steveayre at gmail.com> wrote:
>> 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
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:4da49a0e32765712911065!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Fri Apr 15 01:50:53 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Apr 2011 16:50:53 -0500
Subject: [Freeswitch-users] Gateway with dynamic IP address
In-Reply-To: <BANLkTikzmRaCAq1eXbG-J0SEhacW1ZFYzg@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>
	<BANLkTi=8LfL-CMkczXgJg8FOF1_FTYtxdQ@mail.gmail.com>
	<BANLkTims24tpVuEmq1r84XA1bZd02ebfsw@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58C4E9CDC7@cooper>
	<BANLkTikzmRaCAq1eXbG-J0SEhacW1ZFYzg@mail.gmail.com>
Message-ID: <404411C8-EAB4-4C09-B05C-D9390F4130E6@freeswitch.org>

Get a nat that does Symmetric mappings and it'll go away.

Otherwise ignore it.  Since we are overcoming it.

/b

On Apr 14, 2011, at 4:39 PM, Nick Rosier wrote:

> 2011-04-14 23:35:53.986767 [DEBUG] sofia_glue.c:2971 AUDIO RTP
> [sofia/sipinterface_1/sip:trunk at 192.168.1.153:5063] xxx.xxx.xxx.xxx
> port 25902 -> 192.168.1.153 port 16426 codec: 0 ms: 20
> 
> Any idea how to fix that?



From roger.castaldo at gmail.com  Fri Apr 15 01:28:39 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Thu, 14 Apr 2011 17:28:39 -0400
Subject: [Freeswitch-users] mod_event_socket Whether can get freeswitch
 error messages
In-Reply-To: <BANLkTi=GhdNqDgoV6vMQCmpfK1EhQ68ZRQ@mail.gmail.com>
References: <8slc8n$of1hij@irxd5-201.sinamail.sina.com.cn>
	<BANLkTinrcHwnEUbQ_DAcaf4XOuY6hWun0A@mail.gmail.com>
	<BANLkTi=GhdNqDgoV6vMQCmpfK1EhQ68ZRQ@mail.gmail.com>
Message-ID: <BANLkTikb1+11iAhGOyfKuw=HsCDrOhZxgQ@mail.gmail.com>

If he is referring to log messages, as long as he is using 1.0.7 there is
the ability to pass the comman api log (log_level) through the even socket
which will cause freeswitch to send out log messages through the event
socket as event messages.  Those levels being the same as the ones you can
specify in the command line interface.  Warning though that the log messages
show up as full even messages with all the headers required.

2011/4/14 Steven Ayre <steveayre at gmail.com>

> I think he means the log messages...
>
> fs_cli has code in it that gets those - perhaps you can use that as a
> sample.
>
> -Steve
>
>
>
> 2011/4/14 Michael Collins <msc at freeswitch.org>
>
>> I am not sure if I understand your question. What kinds of error messages
>> are you hoping to see? Also, if you are more comfortable you can try asking
>> your question in your native language (Mandarin?) as we might have some
>> people who are fluent and can help.
>>
>> -MC
>>
>>
>> 2011/4/13 ?? <sc_zhangming at sina.com>
>>
>>> freeswitch-users?
>>>
>>>                  1. mod_event_socket Whether can get freeswitch error
>>> messages
>>>                          2. how to get
>>>
>>> ?
>>> ??
>>>
>>>
>>> sc_zhangming at sina.com
>>> 2011-04-14
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 freeswitch at aastral.net  Fri Apr 15 02:52:46 2011
From: freeswitch at aastral.net (Bill W.)
Date: Thu, 14 Apr 2011 18:52:46 -0400
Subject: [Freeswitch-users] Detect T.38 reinvite and transcode?
Message-ID: <1QAVP9-0001Iq-SY@mail.aastral.net>

Hi there,

Is there a way to have FreeSWITCH detect a T.38 reinvite, accept it, and
transcode that into ULAW before sending it on to the B-leg?

The fax is neither originated nor terminated on the FS gateway.  FS
would simply transcode T.38 to ULAW.

I looked at the mod_spandsp documentation and it wasn't readily apparent.

To be clear
host_a -> FS -> host_b

host_a sends an INVITE
host_b responds with 100, 180, 200
then host_a sends another INVITE with SDP for T.38

Can FS ack that RE-INVITE and transcode, so the legs look like this?
host_a -> T.38 -> FS -> ULAW -> host_b


Thanks!
Bill


From brian at freeswitch.org  Fri Apr 15 03:04:02 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 14 Apr 2011 18:04:02 -0500
Subject: [Freeswitch-users] Detect T.38 reinvite and transcode?
In-Reply-To: <1QAVP9-0001Iq-SY@mail.aastral.net>
References: <1QAVP9-0001Iq-SY@mail.aastral.net>
Message-ID: <D41893DE-0EEC-4DF0-BBB9-22CE63D392AE@freeswitch.org>

that is exactly what t38_gateway does.

/b

On Apr 14, 2011, at 5:52 PM, Bill W. wrote:

> Hi there,
> 
> Is there a way to have FreeSWITCH detect a T.38 reinvite, accept it, and
> transcode that into ULAW before sending it on to the B-leg?
> 
> The fax is neither originated nor terminated on the FS gateway.  FS
> would simply transcode T.38 to ULAW.
> 
> I looked at the mod_spandsp documentation and it wasn't readily apparent.
> 
> To be clear
> host_a -> FS -> host_b
> 
> host_a sends an INVITE
> host_b responds with 100, 180, 200
> then host_a sends another INVITE with SDP for T.38
> 
> Can FS ack that RE-INVITE and transcode, so the legs look like this?
> host_a -> T.38 -> FS -> ULAW -> host_b
> 
> 
> Thanks!
> Bill
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From bwibowo at gmail.com  Fri Apr 15 04:00:23 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Fri, 15 Apr 2011 07:00:23 +0700
Subject: [Freeswitch-users] outbound_caller_id_number
Message-ID: <BANLkTinRaALCY1i9sfwtyQVRQUnzZ3PDSQ@mail.gmail.com>

hi
i setup outbound_caller_id_number in my FS, config looks like this


   <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
      <variable name="outbound_caller_id_number" value="811856609"/>
      <variable name="callgroup" value="techsupport"/>

FS connect via sip to mera switch. mera switch to partner connect via H323.
on H323 i see calling number is just blank.
any idea how set outgoing number for FS?


br

budi
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From sunwood360 at gmail.com  Fri Apr 15 04:26:29 2011
From: sunwood360 at gmail.com (envelopes envelopes)
Date: Thu, 14 Apr 2011 17:26:29 -0700
Subject: [Freeswitch-users] FreeSwitch Debug Full Log(version 1.0.6)
In-Reply-To: <4DA663CB.2040107@163.com>
References: <4DA663CB.2040107@163.com>
Message-ID: <BANLkTikN=g2Y+8H-WYmt2JP8Wq0UoqR6wg@mail.gmail.com>

Run fs in windows vs debugger.
On Apr 14, 2011 9:40 AM, "vivid" <vivid333 at 163.com> wrote:
> how can I use FreeSwitch to trace *the Function call Procedure* in
*fs_cli*.
> (using Command? configuration?)
>
>
> [DEBUG] *switch_ivr_bridge.c*:911 switch_ivr_multi_threaded_bridge()
> sofia/external/1XXX4951027 receive message [BRIDGE]
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From vivid333 at 163.com  Fri Apr 15 04:52:57 2011
From: vivid333 at 163.com (vivid333)
Date: Fri, 15 Apr 2011 08:52:57 +0800 (CST)
Subject: [Freeswitch-users] FreeSwitch Debug Full Log(version 1.0.6)
In-Reply-To: <BANLkTikN=g2Y+8H-WYmt2JP8Wq0UoqR6wg@mail.gmail.com>
References: <BANLkTikN=g2Y+8H-WYmt2JP8Wq0UoqR6wg@mail.gmail.com>
	<4DA663CB.2040107@163.com>
Message-ID: <5165f3de.1420.12f56a581da.Coremail.vivid333@163.com>

Thanks all. I solved the problem.

First, My FS runs in Linux, I use fs_cli to trace FS running.


I add " #define SWITCH_FUNC_IN_LOG " in switch_log.c ,  make the project. at last I can see the function call procedure in log.





At 2011-04-15 08:26:29?"envelopes envelopes" <sunwood360 at gmail.com> wrote:


Run fs in windows vs debugger.

On Apr 14, 2011 9:40 AM, "vivid" <vivid333 at 163.com> wrote:
> how can I use FreeSwitch to trace *the Function call Procedure* in *fs_cli*.
> (using Command? configuration?)
>
>
> [DEBUG] *switch_ivr_bridge.c*:911 switch_ivr_multi_threaded_bridge()
> sofia/external/1XXX4951027 receive message [BRIDGE]
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From flyingnn at gmail.com  Fri Apr 15 05:47:35 2011
From: flyingnn at gmail.com (gmail)
Date: Fri, 15 Apr 2011 09:47:35 +0800
Subject: [Freeswitch-users] can not call between two profile
Message-ID: <BANLkTimcqAf+Yj0UUH=KiaW3L6iRWgaufg@mail.gmail.com>

hi!

1000  register to profile internal.xml, 1010 register to profile
external.xml
1000 and 1010 register successfully.

1000 can not call 1010, [ERR] switch_ivr_originate.c:2638 Cannot create
outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
1010 call 1000 ok.
that is why?

thanks
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From valery.kalinin at gmail.com  Fri Apr 15 05:52:09 2011
From: valery.kalinin at gmail.com (Valery Kalinin)
Date: Fri, 15 Apr 2011 07:52:09 +0600
Subject: [Freeswitch-users]  Cannot compile freetdm!
Message-ID: <BANLkTinntgEpU9L-QSciqC+zt4Y6JrxbqQ@mail.gmail.com>

I download latest version (2011-03-10) libisdn from:
http://oss.axsentis.de/gitweb/?p=libisdn.git;a=summary

Stefan Knoblich cannot help me...:-(

Does FreeSWITCH work only with Sangoma cards?!
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From freeswitch at aastral.net  Fri Apr 15 07:33:49 2011
From: freeswitch at aastral.net (Bill W.)
Date: Thu, 14 Apr 2011 23:33:49 -0400
Subject: [Freeswitch-users] Detect T.38 reinvite and transcode?
In-Reply-To: <D41893DE-0EEC-4DF0-BBB9-22CE63D392AE@freeswitch.org>
References: <1QAVP9-0001Iq-SY@mail.aastral.net>
	<D41893DE-0EEC-4DF0-BBB9-22CE63D392AE@freeswitch.org>
Message-ID: <1QAZn8-0003ZJ-Fu@mail.aastral.net>

Thanks so much Brian!

Seriously, the whole FreeSWITCH team deserves a big round of applause.
 It's such a pleasure to get the answer "Oh yeah, it already does that".

I documented this in the Wiki too.

For posterity:
<extension name="t38_reinvite">
  <condition>
     <action application="set" data="fax_enable_t38=true"/>
    <action application="set" data="fax_enable_t38_request=true"/>
    <action application="set" data="t38_gateway_format=udptl"/>
    <action application="set" data="execute_on_answer=t38_gateway self"/>
    <action application="bridge" data="sofia/external/1234 at host"/>
  </condition>
</extension>


Aastral Communications
bweidman at aastral.net
Tel: 786-522-0440
Tel: 888-404-3280
Cell: 954-540-4931

On 4/14/11 7:04 PM, Brian West wrote:
> that is exactly what t38_gateway does.
> 
> /b
> 
> On Apr 14, 2011, at 5:52 PM, Bill W. wrote:
> 
>> Hi there,
>>
>> Is there a way to have FreeSWITCH detect a T.38 reinvite, accept it, and
>> transcode that into ULAW before sending it on to the B-leg?
>>
>> The fax is neither originated nor terminated on the FS gateway.  FS
>> would simply transcode T.38 to ULAW.
>>
>> I looked at the mod_spandsp documentation and it wasn't readily apparent.
>>
>> To be clear
>> host_a -> FS -> host_b
>>
>> host_a sends an INVITE
>> host_b responds with 100, 180, 200
>> then host_a sends another INVITE with SDP for T.38
>>
>> Can FS ack that RE-INVITE and transcode, so the legs look like this?
>> host_a -> T.38 -> FS -> ULAW -> host_b
>>
>>
>> Thanks!
>> Bill
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From boris at tagnet.ru  Fri Apr 15 07:33:54 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 15 Apr 2011 09:33:54 +0600
Subject: [Freeswitch-users] acl by ANI
Message-ID: <4DA7BCA2.8010309@tagnet.ru>

Hello!

     Is this possible to check incoming calls from gateways for specific 
ANIs without writing script or specific extension? By acl for example?

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




From dome at tel.co.th  Fri Apr 15 09:00:16 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Fri, 15 Apr 2011 12:00:16 +0700
Subject: [Freeswitch-users] acl by ANI
In-Reply-To: <4DA7BCA2.8010309@tagnet.ru>
References: <4DA7BCA2.8010309@tagnet.ru>
Message-ID: <BANLkTikqAbpL_HJ_jds5HOQ7rHeuZjJTuw@mail.gmail.com>

On Fri, Apr 15, 2011 at 10:33 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ? ? Is this possible to check incoming calls from gateways for specific
> ANIs without writing script or specific extension? By acl for example?
why not lua script ?
>
> --
> ? ?????????,
> ? ????? ?????????
> ? ??? "??????"
> ? (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 helmut.kuper at ewetel.de  Fri Apr 15 10:39:45 2011
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Fri, 15 Apr 2011 08:39:45 +0200
Subject: [Freeswitch-users] fax and jpg to tiff
In-Reply-To: <194F3FCAB32848889F2ACD51D7658607@e1705>
References: <1C10456422324E24A1C205759F4EFE1A@e1705>	<4DA6F5D9.1020108@coppice.org><A6A5793A27324ECE935D3D62C04DEF84@e1705>	<4DA719CD.1080007@coppice.org><F15A7C0F6BC648DBBB939260F7BC21F5@e1705>	<4DA72345.6060807@ewetel.de>
	<194F3FCAB32848889F2ACD51D7658607@e1705>
Message-ID: <4DA7E831.7060004@ewetel.de>

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

Hi Franck,

thank you very much. Will put your reply in my email signature ;D


Am 14.04.2011 18:50, schrieb Madovsky:
> Gold medal for Helmut !! :D
> I invite you to the next oktober feist ! :)
> 
> Thanks
> 
> Franck
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Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/

iEYEARECAAYFAk2n6DEACgkQ4tZeNddg3dwhDACgoADtaZHmEnn6+KkPYEO7YDFe
aGEAn3VEqXh3wJTsfs0VZPJ85bkWc4te
=OQPs
-----END PGP SIGNATURE-----


From avi at avimarcus.net  Fri Apr 15 10:56:03 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 15 Apr 2011 09:56:03 +0300
Subject: [Freeswitch-users] outbound_caller_id_number
In-Reply-To: <BANLkTinRaALCY1i9sfwtyQVRQUnzZ3PDSQ@mail.gmail.com>
References: <BANLkTinRaALCY1i9sfwtyQVRQUnzZ3PDSQ@mail.gmail.com>
Message-ID: <BANLkTimQRp6Lc4nmuvcnjpg-aEZdE_D6wg@mail.gmail.com>

Hi, are you using FusionPBX?
FusionPBX stores outbound_caller_id_* to let you set a different internal
and external caller ID. When you set up an extension to make external calls
with a gateway, however, you'll see FusionPBX adds:
<variable name="effective_caller_id_name" value="$${
outbound_caller_id_name}"/>
<variable name="effective_caller_id_number" value="$${
outbound_caller_id_number}"/>

So basically, you have your variable name wrong. You need to be setting
effective.

(Also, the gateway may want a different CID type, e.g. rpid,
pid,<http://wiki.freeswitch.org/wiki/Variable_sip_cid_type>or set
caller-id-in-from to true in your gateway, but this is your first
issue.)

-Avi


On Fri, Apr 15, 2011 at 3:00 AM, budi wibowo <bwibowo at gmail.com> wrote:

> hi
> i setup outbound_caller_id_number in my FS, config looks like this
>
>
>    <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>       <variable name="outbound_caller_id_number" value="811856609"/>
>       <variable name="callgroup" value="techsupport"/>
>
> FS connect via sip to mera switch. mera switch to partner connect via
> H323.
> on H323 i see calling number is just blank.
> any idea how set outgoing number for FS?
>
>
> br
>
> budi
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Apr 15 13:48:25 2011
From: u2nsam at gmail.com (Sam)
Date: Fri, 15 Apr 2011 15:18:25 +0530
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <BANLkTikeGaHpPO4Md_ieQhMzkzpr0n8+7w@mail.gmail.com>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
	<201104140622.52563.sos@sokhapkin.dyndns.org>
	<BANLkTi=0PMw9wVsTaFu-VSZ5EPo=mTR9NQ@mail.gmail.com>
	<201104140828.12059.sos@sokhapkin.dyndns.org>
	<BANLkTikeGaHpPO4Md_ieQhMzkzpr0n8+7w@mail.gmail.com>
Message-ID: <BANLkTim4QO6m9gf=zXA2Xh-MhpP8wrtD_w@mail.gmail.com>

Hello,

AS freeswitch modifies SDP ( Media Description & Media Attributes) so is
there any way to just bypass that process.

Regards
Sam


On Thu, Apr 14, 2011 at 6:32 PM, Steven Ayre <steveayre at gmail.com> wrote:

> I don't mean they're equivalent, because proxy_media has other sideaffects
> such as disabling many features.
>
> I mean there are different better ways to avoid transcoding.
>
> What you suggested with bridge should indeed avoid it. Without the
> absolute_codec_string the bleg would be offered all codecs supported by FS
> which may be more than supported by the caller, however with that setting it
> would only offer those supported by both the caller and FS. The callee will
> pick one it supports (or return 488 Not Acceptable Here). The aleg will then
> use the same codec as the bleg, so at that point both legs will be using the
> same codec so no transcoding will take place (unless you try using something
> like eavesdrop/record). You could then if you wanted handle the 488 Not
> Acceptable Here by redialing without absolute_codec_string to try again,
> this time the bleg could use a different codec than the aleg (transcoding).
>
> -Steve
>
>
>
> On 14 April 2011 13:28, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:
>
>> Do you mean
>>
>> set proxy_media=true
>>
>> is the equivalent of
>>
>> set late_negotiation=true
>> bridge {absolute_codec_string=${ep_codec_string}}blah
>>
>> and transcoding will never happen?
>>
>> On Thursday 14 April 2011, Steven Ayre wrote:
>> > Proxy media is not required for 2 NAT clients to talk to each other.
>> >
>> > Transcoding won't be used if both legs use the same codec, and you can
>> use
>> > late-negotation/absolute_codec_string to encourage that.
>> >
>> > -Steve
>> >
>> > On 14 April 2011 11:22, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
>> wrote:
>> > > Proxy media is useful not for T38 only. It's a way for 2 NATed clients
>> to
>> > > communicate with each other if there are common codecs, no transcoding
>> > > involved.
>> > >
>> > > On Thursday 14 April 2011, Steven Ayre wrote:
>> > > > You misunderstand I think... it *isn't* needed for T38 any longer.
>> Only
>> > >
>> > > in
>> > >
>> > > > very old versions.
>> > > >
>> > > > -Steve
>> > > >
>> > > > On 13 April 2011 23:17, Madovsky <infos at madovsky.org> wrote:
>> > > > > maybe change the param name to proxy_media_t38 ;)
>> > > > >
>> > > > > ----- Original Message -----
>> > > > > From: "Kristian Kielhofner" <kris at kriskinc.com>
>> > > > > To: "FreeSWITCH Users Help" <
>> freeswitch-users at lists.freeswitch.org>
>> > > > > Sent: Wednesday, April 13, 2011 2:10 PM
>> > > > > Subject: Re: [Freeswitch-users] proxy SDP
>> > > > >
>> > > > >
>> > > > > Brian,
>> > > > >
>> > > > >  For all of the confusion proxy media creates I still see cases
>> where
>> > > > >
>> > > > > it is useful...  It shouldn't be removed completely.
>> > > > >
>> > > > > On Tue, Apr 12, 2011 at 8:55 PM, Brian West <brian at freeswitch.org
>> >
>> > >
>> > > wrote:
>> > > > > > We aren't a proxy... we have transcended into this quasi proxy
>> in
>> > >
>> > > some
>> > >
>> > > > > > scenarios which mostly involve t.38... as for proxy media DO NOT
>> > > > > > USE IT. Just saying it might go away since the purpose of it is
>> > > > > > now not needed since we have full t.38.
>> > > > > >
>> > > > > > Thanks,
>> > > > > > Brian
>> > > > >
>> > > > > --
>> > > > > 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
>> > > > >
>> > > > >
>> > > > > _______________________________________________
>> > > > > FreeSWITCH-users mailing list
>> > > > > FreeSWITCH-users at lists.freeswitch.org
>> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > >
>> > > > > UNSUBSCRIBE:
>> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > >
>> > > > > http://www.freeswitch.org
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 benkokakao at gmail.com  Fri Apr 15 15:12:42 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Fri, 15 Apr 2011 13:12:42 +0200
Subject: [Freeswitch-users] Cannot compile freetdm!
In-Reply-To: <BANLkTinntgEpU9L-QSciqC+zt4Y6JrxbqQ@mail.gmail.com>
References: <BANLkTinntgEpU9L-QSciqC+zt4Y6JrxbqQ@mail.gmail.com>
Message-ID: <BANLkTimQb+nwq8epZGjdRp8U5pN-FOGsrA@mail.gmail.com>

Hi!

Which hardware exactly are you trying to get to run?

Best regards
Christian


From kris at kriskinc.com  Fri Apr 15 17:58:56 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 15 Apr 2011 09:58:56 -0400
Subject: [Freeswitch-users] Relay Proxy-Authentication requests to the
	caller
In-Reply-To: <9BCB4EA5-C859-42F0-80BB-E77DF2E76B8C@dchorton.com>
References: <mailman.7437.1302799201.24555.freeswitch-users@lists.freeswitch.org>
	<4058C325-0AC0-4878-B86E-95EF11329086@dchorton.com>
	<BANLkTinDXVz=c2BXoR37wUbYx3MNPU+mvw@mail.gmail.com>
	<8207E7C4-ECD5-415D-BA16-5AA9AC843C89@dchorton.com>
	<BANLkTim9Z6Smc8NPK82AOLJ-TShwcsH8JQ@mail.gmail.com>
	<9BCB4EA5-C859-42F0-80BB-E77DF2E76B8C@dchorton.com>
Message-ID: <BANLkTimFu6ziAqvwtHkNPF-chv=nVb2h9g@mail.gmail.com>

Dave,

  With a project as large and complex as FreeSWITCH I highly recommend
you don't try to maintain a patch separately.  That's the double edge
sword of open source - it will bite you eventually.  I've found that
the best approach is to make your goals align with those of the
developers so your patches (if necessary at all) are accepted.

  When the only tool you have is a hammer every problem looks like a
nail...  You've already decided you need "a simple transcoding server"
and that's what you're looking for from FreeSWITCH even though the way
you've decided to implement it doesn't fit very well with FreeSWITCH.
Instead of forcing a square peg into a round hole perhaps this could
be viewed from 30,000 ft again and re-evaluated.  That's what I was
suggesting with my previous response.

  You said you have multiple remote FreeSWITCH PBXs deployed around
the globe.  I assume you have control over these PBXs?  Why not deploy
another system (FreeSWITCH and/or Kamailio) and configure it as an
outbound proxy to do your transcoding?  No code changes in FreeSWITCH
(the remote PBX or otherwise) and you maintain compatibility with
every system and piece of software out there.  FreeSWITCH, Asterisk,
dumb ATAs, phones, etc, etc - all can be configured to use an outbound
proxy and you even have more freedom over which software/hardware you
select for your "transcoding server".

  The selection of speex is curious too.  While certainly offering
bandwidth savings it's almost certainly not supported by your
endpoints on the PBX or your provider.  What codec do they use?  What
does your codec path currently look like? PCMU <-> Speex <-> PCMU?
Why not use G729, iLBC, GSM, or potentially some other low bandwidth
codec that may be supported by both your endpoints and your IP
provider (G729 being the most likely) thus eliminating the need for a
transcoding server at all?  Less media proxying, less delay/quality
issues due to transcoding, and you save some CPU cycles.

  While there is some debate over the role/definition of a B2BUA in
SIP most descriptions contain "splitting the call into two halves" and
the concept of the B2BUA being both the UAS and the UAC.  Passing
Proxy-Auth from one leg to the other is pretty clearly, in my opinion,
outside of these descriptors.

  More than anything (as you can probably tell by now) I enjoy
debating SIP architecture especially when there can be an especially
civil and respectful discourse.  This is unfortunately becoming
exceedingly rare on the internets these days (damn kids).

On Thu, Apr 14, 2011 at 3:23 PM, Dave Horton <dave at dchorton.com> wrote:
> Hi Kristian -
>
> I appreciate your thoughts/feedback/guidance, as well as the manner in which they were delivered.
> At the risk of prolonging a thread that perhaps has reached the point of maximal usefulness, I'd like to
> respond briefly to a few points:
>
> 1) I don't intend to submit a patch, because I agree it seems to go against what FS wants to do --
> at least, as best I can understand what FS wants to do since as a newbie I'm sort of puzzling that out as I go
> by means of this forum, reading the book, etc.
> However, FS seems to somewhat have the burden of success, in that "what it wants to do" and
> "what it can do" seem to be overlapping but not exactly equal sets (the second being larger) --
> due to its flexibility it seems that FS can do quite a bit, as in my case of needing to whip up
> a simple transcoding server.
>
> 2) I don't think there is much to re-think about the overall design and architecture in my case,
> simply because I need a transcoding server to sit a certain point in a service provider's network
> and that's that. ?In my case, I have multiple remote FS PBXs around the globe transmitting
> to a gateway in a service provider in the US for PSTN access, over a bandwidth constrained
> satellite link. ?Therefore speex 8ks make sense over the satellite link, but then I need to
> transcode back to PCMU once in the service providers network. ?Authentication is unfortunately
> inextricably tied up with establishing the media path since credentials are carried in the INVITE.
>
> 3) I am personally mostly interested in building applications that run in a service provider core network, not out on the edge.
> I'd be interested to know if this is considered in the target of what FS wants to be? ?Obviously,
> FS is first and foremost a PBX, but of course its not limited to such. ?Asterisk, as an example, was
> adopted by some service providers for various "application server" types of applications,
> sometimes with success, sometimes without.
>
> 4) In service provider networks, almost all "interesting" applications are B2BUA.
> And I would disagree that SIP has a well-defined set of constraints on what
> a B2BUA can and should be. ?To the contrary, I think its pretty wide open
> (at least by comparison to specifications for proxies, e.g.)
>
> 5) This question of handling (or not) SIP headers doesn't seem to be
> documented consistently. ?In the FS book, there is a note that suggests
> you *should* only mess with X- headers, but implies you are not limited
> to doing so. ?In fact, I see no way to get at the other headers (other than
> the obvious suspects -- to, from, contact, etc -- and one or two special cases
> -- alert-info -- that seemed to have been plucked out over time). ?It
> just doesn't seem like it would be all that difficult to provide access
> to all the sip headers for developers that might need them).
>
> Finally, as a newbie to FS, but with some experience building SIP platforms
> and applications for a bit, I would be remiss in not congratulating the developers
> on the great product they have created. ?Well done!
>

-- 
Kristian Kielhofner


From kris at kriskinc.com  Fri Apr 15 18:05:14 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 15 Apr 2011 10:05:14 -0400
Subject: [Freeswitch-users] proxy SDP
In-Reply-To: <BANLkTim4QO6m9gf=zXA2Xh-MhpP8wrtD_w@mail.gmail.com>
References: <BANLkTimwa54dFBRc31y+FMJXsOXfHRKJow@mail.gmail.com>
	<201104140622.52563.sos@sokhapkin.dyndns.org>
	<BANLkTi=0PMw9wVsTaFu-VSZ5EPo=mTR9NQ@mail.gmail.com>
	<201104140828.12059.sos@sokhapkin.dyndns.org>
	<BANLkTikeGaHpPO4Md_ieQhMzkzpr0n8+7w@mail.gmail.com>
	<BANLkTim4QO6m9gf=zXA2Xh-MhpP8wrtD_w@mail.gmail.com>
Message-ID: <BANLkTinBRvOaA7P0feb8y5f-7-RqXGU2EQ@mail.gmail.com>

I'm really starting to think that no amount of documentation will ever
make this clear:

FreeSWITCH default media mode:
- RTP proxied by FreeSWITCH
- FreeSWITCH controls codec negotiation
- If endpoints agree on same codec, no transcoding is performed
- All features enabled - recording, DTMF interception, etc, etc

Proxy media mode:
- RTP proxied by FreeSWITCH (c= modified, that's it)
- FreeSWITCH has no control or even understanding of other SDP
parameters (pass through)
- Endpoints *MUST* agree on same codec because FreeSWITCH can't help them
- Virtually no features available

Bypass media mode:
- RTP *NOT* proxied by FreeSWITCH
- FreeSWITCH has no control over anything SDP related - it's
completely pass through from one leg to next (including c=)
- Endpoints must agree on same codec because FreeSWITCH doesn't even
see the media or SDP
- Virtually no features available

On Fri, Apr 15, 2011 at 5:48 AM, Sam <u2nsam at gmail.com> wrote:
> Hello,
>
> AS freeswitch modifies SDP ( Media Description & Media Attributes) so is
> there any way to just bypass that process.
>
> Regards
> Sam
>
>
> On Thu, Apr 14, 2011 at 6:32 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> I don't mean they're equivalent, because proxy_media has other sideaffects
>> such as disabling many features.
>>
>> I mean there are different better ways to avoid transcoding.
>>
>> What you suggested with bridge should indeed avoid it. Without the
>> absolute_codec_string the bleg would be offered all codecs supported by FS
>> which may be more than supported by the caller, however with that setting it
>> would only offer those supported by both the caller and FS. The callee will
>> pick one it supports (or return 488 Not Acceptable Here). The aleg will then
>> use the same codec as the bleg, so at that point both legs will be using the
>> same codec so no transcoding will take place (unless you try using something
>> like eavesdrop/record). You could then if you wanted handle the 488 Not
>> Acceptable Here by redialing without absolute_codec_string to try again,
>> this time the bleg could use a different codec than the aleg (transcoding).
>>
>> -Steve
>>
>>
>> On 14 April 2011 13:28, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:
>>>
>>> Do you mean
>>>
>>> set proxy_media=true
>>>
>>> is the equivalent of
>>>
>>> set late_negotiation=true
>>> bridge {absolute_codec_string=${ep_codec_string}}blah
>>>
>>> and transcoding will never happen?
>>>
>>> On Thursday 14 April 2011, Steven Ayre wrote:
>>> > Proxy media is not required for 2 NAT clients to talk to each other.
>>> >
>>> > Transcoding won't be used if both legs use the same codec, and you can
>>> > use
>>> > late-negotation/absolute_codec_string to encourage that.
>>> >
>>> > -Steve
>>> >
>>> > On 14 April 2011 11:22, Sergey Okhapkin <sos at sokhapkin.dyndns.org>
>>> > wrote:
>>> > > Proxy media is useful not for T38 only. It's a way for 2 NATed
>>> > > clients to
>>> > > communicate with each other if there are common codecs, no
>>> > > transcoding
>>> > > involved.
>>> > >
>>> > > On Thursday 14 April 2011, Steven Ayre wrote:
>>> > > > You misunderstand I think... it *isn't* needed for T38 any longer.
>>> > > > Only
>>> > >
>>> > > in
>>> > >
>>> > > > very old versions.
>>> > > >
>>> > > > -Steve
>>> > > >
>>> > > > On 13 April 2011 23:17, Madovsky <infos at madovsky.org> wrote:
>>> > > > > maybe change the param name to proxy_media_t38 ;)
>>> > > > >
>>> > > > > ----- Original Message -----
>>> > > > > From: "Kristian Kielhofner" <kris at kriskinc.com>
>>> > > > > To: "FreeSWITCH Users Help"
>>> > > > > <freeswitch-users at lists.freeswitch.org>
>>> > > > > Sent: Wednesday, April 13, 2011 2:10 PM
>>> > > > > Subject: Re: [Freeswitch-users] proxy SDP
>>> > > > >
>>> > > > >
>>> > > > > Brian,
>>> > > > >
>>> > > > > ?For all of the confusion proxy media creates I still see cases
>>> > > > > where
>>> > > > >
>>> > > > > it is useful... ?It shouldn't be removed completely.
>>> > > > >
>>> > > > > On Tue, Apr 12, 2011 at 8:55 PM, Brian West
>>> > > > > <brian at freeswitch.org>
>>> > >
>>> > > wrote:
>>> > > > > > We aren't a proxy... we have transcended into this quasi proxy
>>> > > > > > in
>>> > >
>>> > > some
>>> > >
>>> > > > > > scenarios which mostly involve t.38... as for proxy media DO
>>> > > > > > NOT
>>> > > > > > USE IT. Just saying it might go away since the purpose of it is
>>> > > > > > now not needed since we have full t.38.
>>> > > > > >
>>> > > > > > Thanks,
>>> > > > > > Brian
>>> > > > >
>>> > > > > --
>>> > > > > 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
>>> > > > >
>>> > > > >
>>> > > > > _______________________________________________
>>> > > > > FreeSWITCH-users mailing list
>>> > > > > FreeSWITCH-users at lists.freeswitch.org
>>> > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > >
>>> > > > > UNSUBSCRIBE:
>>> > > http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > >
>>> > > > > http://www.freeswitch.org
>>> > >
>>> > > _______________________________________________
>>> > > FreeSWITCH-users mailing list
>>> > > FreeSWITCH-users at lists.freeswitch.org
>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > >
>>> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > > http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 dave at dchorton.com  Fri Apr 15 18:38:34 2011
From: dave at dchorton.com (Dave Horton)
Date: Fri, 15 Apr 2011 10:38:34 -0400
Subject: [Freeswitch-users] Relay Proxy-Authentication requests to the
	caller
In-Reply-To: <BANLkTimFu6ziAqvwtHkNPF-chv=nVb2h9g@mail.gmail.com>
References: <mailman.7437.1302799201.24555.freeswitch-users@lists.freeswitch.org>
	<4058C325-0AC0-4878-B86E-95EF11329086@dchorton.com>
	<BANLkTinDXVz=c2BXoR37wUbYx3MNPU+mvw@mail.gmail.com>
	<8207E7C4-ECD5-415D-BA16-5AA9AC843C89@dchorton.com>
	<BANLkTim9Z6Smc8NPK82AOLJ-TShwcsH8JQ@mail.gmail.com>
	<9BCB4EA5-C859-42F0-80BB-E77DF2E76B8C@dchorton.com>
	<BANLkTimFu6ziAqvwtHkNPF-chv=nVb2h9g@mail.gmail.com>
Message-ID: <DE472EA4-7E39-492C-863F-6BFAA604CAE6@dchorton.com>

Hi Kristian - 

I'm not quite grasping the alternative solution you are describing, and I 
certainly would like to if there is an alternative.  In hopes of clarifying
my understanding, let me respond to some of your questions on the
architecture:

>>You said you have multiple remote FreeSWITCH PBXs deployed around
the globe.  I assume you have control over these PBXs?  

[DH] Yes, I have control over the configuration of these FS PBXs.  Of course, 
to minimize operational headaches, I want to simplify and standardize their
configuration as much as possible.  My current solution uses a very vanilla
configuration, with pretty much only two changes out of the box, 
which is the (a) addition of a gateway to point to the NA-based service provider, 
and (b) enabling speex in the list of supported codecs.

Why not deploy another system (FreeSWITCH and/or Kamailio) and configure it as an
outbound proxy to do your transcoding?  

[DH]  Hmm.  This is where I'm not clear.  I understand the use of an outbound proxy, 
but usually I am familiar with those being deployed at the customer premise, 
to funnel all traffic through to the internet and provide any associated features
(e.g., billing, limiting, and yes transcoding if needed).  However, the  transcoding 
that I need to do at the customer premise side is being done by the FS PBX natively, 
because FS already supports speex.  So there is no need for an additional transcoding
resource at the customer side. I am simply going to have the media stream encoded 
in speex by FS as leaves (no changes needed to FS config for this, other than those above), 
and the point of transcoding back to PCMU needs to happen on the other end, 
once it gets in the service provider network.  Any calls originating from the FS PBX that use
the gateway may offer multiple codecs in the INVITE (the list will include speex by virtue of
the config change mentioned above), but the 1xx/200 responses coming
back from the service provider gateway will only offer speex.  

....No code changes in FreeSWITCH (the remote PBX or otherwise) 

[DH] Right.  I definitely want no changes to FS code at remote PBXs, and
my current solution does achieve that.  The only thing I am doing
is plopping a transcoding server in the service provider core network.  The
PBXs are plain vanilla FS with simple configurations.

As far as the no transcoding option, G.729 *is* a possibility (the others
are not because they are not supported by the service provider
endpoints, though the service provider customer has some
gear that does not support G.729 and prefers PCMU within
his network.

So in the end, I'm mostly a pragmatic (read: lazy) kind of guy, 
and since I was able to solve my problem in about 4 hours
by building a simple transcoding server using FS (positioned
in the core service provider's network) I guess I'm pretty satisified at this point.

Dave

On Apr 15, 2011, at 9:58 AM, Kristian Kielhofner wrote:

Dave,

 With a project as large and complex as FreeSWITCH I highly recommend
you don't try to maintain a patch separately.  That's the double edge
sword of open source - it will bite you eventually.  I've found that
the best approach is to make your goals align with those of the
developers so your patches (if necessary at all) are accepted.

 When the only tool you have is a hammer every problem looks like a
nail...  You've already decided you need "a simple transcoding server"
and that's what you're looking for from FreeSWITCH even though the way
you've decided to implement it doesn't fit very well with FreeSWITCH.
Instead of forcing a square peg into a round hole perhaps this could
be viewed from 30,000 ft again and re-evaluated.  That's what I was
suggesting with my previous response.

 You said you have multiple remote FreeSWITCH PBXs deployed around
the globe.  I assume you have control over these PBXs?  Why not deploy
another system (FreeSWITCH and/or Kamailio) and configure it as an
outbound proxy to do your transcoding?  No code changes in FreeSWITCH
(the remote PBX or otherwise) and you maintain compatibility with
every system and piece of software out there.  FreeSWITCH, Asterisk,
dumb ATAs, phones, etc, etc - all can be configured to use an outbound
proxy and you even have more freedom over which software/hardware you
select for your "transcoding server".

 The selection of speex is curious too.  While certainly offering
bandwidth savings it's almost certainly not supported by your
endpoints on the PBX or your provider.  What codec do they use?  What
does your codec path currently look like? PCMU <-> Speex <-> PCMU?
Why not use G729, iLBC, GSM, or potentially some other low bandwidth
codec that may be supported by both your endpoints and your IP
provider (G729 being the most likely) thus eliminating the need for a
transcoding server at all?  Less media proxying, less delay/quality
issues due to transcoding, and you save some CPU cycles.

 While there is some debate over the role/definition of a B2BUA in
SIP most descriptions contain "splitting the call into two halves" and
the concept of the B2BUA being both the UAS and the UAC.  Passing
Proxy-Auth from one leg to the other is pretty clearly, in my opinion,
outside of these descriptors.

 More than anything (as you can probably tell by now) I enjoy
debating SIP architecture especially when there can be an especially
civil and respectful discourse.  This is unfortunately becoming
exceedingly rare on the internets these days (damn kids).

On Thu, Apr 14, 2011 at 3:23 PM, Dave Horton <dave at dchorton.com> wrote:
> Hi Kristian -
> 
> I appreciate your thoughts/feedback/guidance, as well as the manner in which they were delivered.
> At the risk of prolonging a thread that perhaps has reached the point of maximal usefulness, I'd like to
> respond briefly to a few points:
> 
> 1) I don't intend to submit a patch, because I agree it seems to go against what FS wants to do --
> at least, as best I can understand what FS wants to do since as a newbie I'm sort of puzzling that out as I go
> by means of this forum, reading the book, etc.
> However, FS seems to somewhat have the burden of success, in that "what it wants to do" and
> "what it can do" seem to be overlapping but not exactly equal sets (the second being larger) --
> due to its flexibility it seems that FS can do quite a bit, as in my case of needing to whip up
> a simple transcoding server.
> 
> 2) I don't think there is much to re-think about the overall design and architecture in my case,
> simply because I need a transcoding server to sit a certain point in a service provider's network
> and that's that.  In my case, I have multiple remote FS PBXs around the globe transmitting
> to a gateway in a service provider in the US for PSTN access, over a bandwidth constrained
> satellite link.  Therefore speex 8ks make sense over the satellite link, but then I need to
> transcode back to PCMU once in the service providers network.  Authentication is unfortunately
> inextricably tied up with establishing the media path since credentials are carried in the INVITE.
> 
> 3) I am personally mostly interested in building applications that run in a service provider core network, not out on the edge.
> I'd be interested to know if this is considered in the target of what FS wants to be?  Obviously,
> FS is first and foremost a PBX, but of course its not limited to such.  Asterisk, as an example, was
> adopted by some service providers for various "application server" types of applications,
> sometimes with success, sometimes without.
> 
> 4) In service provider networks, almost all "interesting" applications are B2BUA.
> And I would disagree that SIP has a well-defined set of constraints on what
> a B2BUA can and should be.  To the contrary, I think its pretty wide open
> (at least by comparison to specifications for proxies, e.g.)
> 
> 5) This question of handling (or not) SIP headers doesn't seem to be
> documented consistently.  In the FS book, there is a note that suggests
> you *should* only mess with X- headers, but implies you are not limited
> to doing so.  In fact, I see no way to get at the other headers (other than
> the obvious suspects -- to, from, contact, etc -- and one or two special cases
> -- alert-info -- that seemed to have been plucked out over time).  It
> just doesn't seem like it would be all that difficult to provide access
> to all the sip headers for developers that might need them).
> 
> Finally, as a newbie to FS, but with some experience building SIP platforms
> and applications for a bit, I would be remiss in not congratulating the developers
> on the great product they have created.  Well done!
> 

-- 
Kristian Kielhofner




From manieq at wp.eu  Fri Apr 15 16:23:01 2011
From: manieq at wp.eu (Mariusz Czulada)
Date: Fri, 15 Apr 2011 14:23:01 +0200
Subject: [Freeswitch-users] bind_meta_app blocks DTMFs?
In-Reply-To: <4d307a65a43472.49337517@wp.pl>
	<AANLkTinAx-cxv_a5b8+VA89duWY=xHjOOmMcZ604Z8R8@mail.gmail.com>
	<4d35cee3110bc5.51489825@wp.pl>
References: <4d307a65a43472.49337517@wp.pl>
	<AANLkTinAx-cxv_a5b8+VA89duWY=xHjOOmMcZ604Z8R8@mail.gmail.com>
	<4d35cee3110bc5.51489825@wp.pl>
Message-ID: <4da838a525c533.15554765@wp.pl>

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From omdivo at gmail.com  Fri Apr 15 17:16:10 2011
From: omdivo at gmail.com (omdivo)
Date: Fri, 15 Apr 2011 15:16:10 +0200
Subject: [Freeswitch-users] Integration of SIP-T or MGCP in FreeSwitch
Message-ID: <4DA8451A.3070801@gmail.com>

Hello,

Is somebody tried to connect two Cisco's MGC using Freeswitch using MGCP 
or SIP-T/SIP-I?

Regards.


From sadhika at gmail.com  Fri Apr 15 18:06:54 2011
From: sadhika at gmail.com (Sadhika Sharma)
Date: Fri, 15 Apr 2011 19:36:54 +0530
Subject: [Freeswitch-users] Help with rxfax debug "T4 expires in phase
	T30_PHASE_B_RX"
Message-ID: <BANLkTimW6WXQ+_Ge+ASG5u1bVe_j36pTHw@mail.gmail.com>

Hi,

I need help in debugging hangup cause 548, 'Disconnected after permitted
retries' while attempting rxfax().

Logs show that T4 expires in phase T30_PHASE_B_RX.
T30 PHASE B concerns exchange of DIS/DCS messages. After sending DIS, mod
spandsp keeps waiting for DCS and exhausts its retries.

Please guide me how to debug the root cause of loss of DCS message. Logs are
mentioned below:

2011-03-07 15:53:48.675180 [DEBUG] switch_ivr.c:557 OpenZAP/4:1/43851609
Command Execute rxfax(/srv/fax/in/fcf15a4c-48a4-11e0-919e-7bf53bcc3f81.tiff)
2011-03-07 15:53:53.334614 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send
complete in phase T30_PHASE_B_TX, state 17
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 DIS:
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ....
...0= Store and forward Internet fax (T.37): Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ....
.0..= Real-time Internet fax (T.38): Not set
:
:
2011-03-07 15:53:53.875238 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Start T4
2011-03-07 15:53:53.895315 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC
signal status is Carrier up (-2) in state 17
2011-03-07 15:53:53.915402 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC
signal status is Carrier down (-1) in state 17
2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 T4
expired in phase T30_PHASE_B_RX, state 17
2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Retry
number 1


Please guide me how to debug the root cause of loss of DCS message.
Thanks,
Sadhika
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From kris at kriskinc.com  Fri Apr 15 19:35:14 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 15 Apr 2011 11:35:14 -0400
Subject: [Freeswitch-users] Integration of SIP-T or MGCP in FreeSwitch
In-Reply-To: <4DA8451A.3070801@gmail.com>
References: <4DA8451A.3070801@gmail.com>
Message-ID: <BANLkTikpyVFhCfaFW_bbfjehO2LFh+kU+g@mail.gmail.com>

Sounds like a job for YATE:

http://yate.null.ro/pmwiki/

FreeSWITCH doesn't have support for any sort of encapsulated ISUP in
SIP.  It can do passthrough of multipart bodies but it can't currently
decode them in any way.

On Fri, Apr 15, 2011 at 9:16 AM, omdivo <omdivo at gmail.com> wrote:
> Hello,
>
> Is somebody tried to connect two Cisco's MGC using Freeswitch using MGCP
> or SIP-T/SIP-I?
>
> 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
>



-- 
Kristian Kielhofner


From msc at freeswitch.org  Fri Apr 15 19:54:54 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Apr 2011 08:54:54 -0700
Subject: [Freeswitch-users] Non-transcoding G729 calls using Sangoma D100
In-Reply-To: <BANLkTimJiR=NnUNL4KtYZOSuQpLt9Jru7w@mail.gmail.com>
References: <BANLkTimJiR=NnUNL4KtYZOSuQpLt9Jru7w@mail.gmail.com>
Message-ID: <BANLkTi=h5n2kgKpTi0jmKPfAcunof+VeLw@mail.gmail.com>

Are you attempting to record the calls that are passing through the D100?
The reason I ask is that if you are doing anything with media, like
recording the call or playing MOH, etc. to either leg then it will need to
do transcoding.

Let's ask Moises Silva at Sangoma. Moy: can you comment on this?
Thanks,
MC

On Thu, Apr 14, 2011 at 4:32 AM, David Notivol <dnotivol at gmail.com> wrote:

> Hi all,
>
> I'm using a Sangoma D100 card for transcoding with FreeSwitch. It seems the
> normal behavior is a call not transcoding (g729-g729) is using a session in
> the sangoma card, although it reports the call is not encoding nor decoding.
>
> Since we can't load mod_g729 and mod_sangoma_g729 at the same time; do you
> know if is there any way to avoid g729-g729 calls using Sangoma sessions?
>
> Thanks in advance.
>
>
> Regards,
> David Notivol
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 15 19:58:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Apr 2011 08:58:18 -0700
Subject: [Freeswitch-users] The Usage of uuid_preprocess && echo
	cancellation
In-Reply-To: <6224958b.125d6.12f51993b3d.Coremail.vivid333@163.com>
References: <6224958b.125d6.12f51993b3d.Coremail.vivid333@163.com>
Message-ID: <BANLkTikzAJtMYpNZQZj0c3k_ssZvDx9bzA@mail.gmail.com>

Do you know what is causing the echo? Is it acoustic echo? Do the endpoint
devices have any echo cancelation abilities?
-MC

2011/4/13 vivid333 <vivid333 at 163.com>

> Hi:
>           1. can any one know the usage of uuid_preprocess?
>           2. according to the FreeSwitch code, this command would add a bug
> to the uuid session, initialize echo parameters, and in functions
> switch_core_session_read_frame/switch_core_session_write_frame do echo
> handle. but  the echo still exist. do any one help me ? thanks..
>
>       Echo Scenes:
>       Telephone 1000, 1001 registered the FreeSwitch, 1000 call 1001, 1001
> answered( two of them are handfree, producing scary sound);
>
> Try Solution:
> Launch the command-line ulility:
> #> fs_cli
> #>: show channels
> uuid1, uuid2
>
> #>:uuid_preprocess uuid1 recho_cancel=true;
> #>:uuid_preprocess uuid2 recho_cancel=true;
>
> but the echo still exist. can any one help me, thanks!
>
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From msc at freeswitch.org  Fri Apr 15 20:08:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Apr 2011 09:08:49 -0700
Subject: [Freeswitch-users] mod_xml_curl,
	sofia_reg.c:1872 Can't find user ...
In-Reply-To: <4DA747CC.3080509@amooma.de>
References: <4DA747CC.3080509@amooma.de>
Message-ID: <BANLkTimBad6ogmWhWDcokpSKqGcC5_C=7Q@mail.gmail.com>

Philipp,

Have you tried to add the <groups> and <group> XML elements? The wiki
mentions those as part of a minimal request:


<document type="freeswitch/xml">
  <section name="directory">
    <domain name="domain1.awesomevoipdomain.faketld">
      <params>
        <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
      </params>*      <groups>
        <group name="default">*         <users>
          <user id="1004">
            <params>
              <param name="password" value="some_password"/>
            </params>
          </user>
         </users>*        </group>
      </groups>*    </domain>
  </section>
</document>

Try that and see.
-MC

On Thu, Apr 14, 2011 at 12:15 PM, Philipp Kempgen <philipp.kempgen at amooma.de
> wrote:

> Hi,
>
> I'm playing with mod_xml_curl per
> http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22
> but having some problems.
>
> Softphone (username "11") sends a REGISTER to FreeSwitch (1.0.6)
> at 192.168.65.133.
>
> xml_curl.conf.xml:
> ---------------------------------------------------------------
>  <binding name="users directory">
>    <param name="gateway-url" value="
> http://127.0.0.1:3000/freeswitch-directory-entries/search.xml"
> bindings="directory" />
>  </binding>
> ---------------------------------------------------------------
>
> FS sends a lookup request to the web service. Parameters:
> ---------------------------------------------------------------
> section=directory
> ...
> key=id
> user=11
> domain=192.168.65.133
> ---------------------------------------------------------------
>
> Fine.
>
> The response is:
> ---------------------------------------------------------------
> <?xml version="1.0" encoding="UTF-8" standalone="yes"?>
> <document type="freeswitch/xml">
>  <section name="directory">
>    <domain name="192.168.65.133">
>      <users>
>        <user id="11" number-alias="1111">
>          <params>
>            <param name="dial-string" value="${sofia_contact(
> 11 at 192.168.65.133)}" />
>            <param name="password" value="123" />
>            <param name="vm-password" value="417906717460606223" />
>          </params>
>          <variables>
>            <variable name="accountcode" value="11" />
>            <variable name="effective_caller_id_number" value="foo" />
>            <variable name="effective_caller_id_name" value="Foo, Foo" />
>            <variable name="outbound_caller_id_number" value="foo" />
>            <variable name="outbound_caller_id_name" value="Foo, Foo" />
>            <variable name="user_context" value="default" />
>            <variable name="toll_allow" value="domestic,international,local"
> />
>            <variable name="callgroup" value="users" />
>          </variables>
>        </user>
>      </users>
>    </domain>
>  </section>
> </document>
> ---------------------------------------------------------------
>
> But I get a complaint on the console:
> [WARNING] sofia_reg.c:1872 Can't find user [11 at 192.168.65.133]
> You must define a domain called '192.168.65.133' in your directory
> and add a user with the id="11" attribute and you must configure
> your device to use the proper domain in it's authentication
> credentials.
>
> What can it be?
>
> A domain with name="192.168.65.133" and a user with id="11" is in
> the response, so why doesn't sofia find it?
>
>
>    Philipp
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 15 20:15:02 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Apr 2011 09:15:02 -0700
Subject: [Freeswitch-users] Detect T.38 reinvite and transcode?
In-Reply-To: <1QAZn8-0003ZJ-Fu@mail.aastral.net>
References: <1QAVP9-0001Iq-SY@mail.aastral.net>
	<D41893DE-0EEC-4DF0-BBB9-22CE63D392AE@freeswitch.org>
	<1QAZn8-0003ZJ-Fu@mail.aastral.net>
Message-ID: <BANLkTik2qG33Z=RR5bcafzcdG0BEY+HZkA@mail.gmail.com>

Thanks for paying the wiki tax! Now you just need to sign up for cluecon...
;)

-MC

On Thu, Apr 14, 2011 at 8:33 PM, Bill W. <freeswitch at aastral.net> wrote:

> Thanks so much Brian!
>
> Seriously, the whole FreeSWITCH team deserves a big round of applause.
>  It's such a pleasure to get the answer "Oh yeah, it already does that".
>
> I documented this in the Wiki too.
>
> For posterity:
> <extension name="t38_reinvite">
>  <condition>
>     <action application="set" data="fax_enable_t38=true"/>
>    <action application="set" data="fax_enable_t38_request=true"/>
>    <action application="set" data="t38_gateway_format=udptl"/>
>    <action application="set" data="execute_on_answer=t38_gateway self"/>
>    <action application="bridge" data="sofia/external/1234 at host"/>
>  </condition>
> </extension>
>
>
> Aastral Communications
> bweidman at aastral.net
> Tel: 786-522-0440
> Tel: 888-404-3280
> Cell: 954-540-4931
>
> On 4/14/11 7:04 PM, Brian West wrote:
> > that is exactly what t38_gateway does.
> >
> > /b
> >
> > On Apr 14, 2011, at 5:52 PM, Bill W. wrote:
> >
> >> Hi there,
> >>
> >> Is there a way to have FreeSWITCH detect a T.38 reinvite, accept it, and
> >> transcode that into ULAW before sending it on to the B-leg?
> >>
> >> The fax is neither originated nor terminated on the FS gateway.  FS
> >> would simply transcode T.38 to ULAW.
> >>
> >> I looked at the mod_spandsp documentation and it wasn't readily
> apparent.
> >>
> >> To be clear
> >> host_a -> FS -> host_b
> >>
> >> host_a sends an INVITE
> >> host_b responds with 100, 180, 200
> >> then host_a sends another INVITE with SDP for T.38
> >>
> >> Can FS ack that RE-INVITE and transcode, so the legs look like this?
> >> host_a -> T.38 -> FS -> ULAW -> host_b
> >>
> >>
> >> Thanks!
> >> Bill
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 15 20:51:43 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 15 Apr 2011 12:51:43 -0400
Subject: [Freeswitch-users] bind_meta_app blocks DTMFs?
References: <4d307a65a43472.49337517@wp.pl><AANLkTinAx-cxv_a5b8+VA89duWY=xHjOOmMcZ604Z8R8@mail.gmail.com><4d35cee3110bc5.51489825@wp.pl>
	<4da838a525c533.15554765@wp.pl>
Message-ID: <782BB906C55E474AA91DAB8BFB2DF6B1@e1705>

      <action application="set" data="hangup_after_bridge=false"/>
      <action application="set" data="continue_on_fail=true"/>
you don't need to specify continue_on_fail=tru since you don't hangup after bridge


  ----- Original Message ----- 
  From: Mariusz Czulada 
  To: FreeSWITCH Users Help 
  Sent: Friday, April 15, 2011 8:23 AM
  Subject: Re: [Freeswitch-users] bind_meta_app blocks DTMFs?


  Hi all,



  I finally solved the problem (bind_meta_app do not block DTMF), so I'd like to share the solution here, in case someone else is also interested in similiar function. What I needed was to replace direct call to conference app:

  <action application="conference" data="confX_${confX_num}_2"/>


  with call via loopback:

  <action application="bridge" data="loopback/app=conference:cx_${cx_num}_2"/>

  Now my subX_2 extension looks like this:



    <extension name="subX_2">


      <condition field="destination_number" expression="^subX_2$">
        <action application="set" data="pass_rfc2833=true"/>
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="bridge_terminate_key=2"/>
        <action application="bridge" data="loopback/app=conference:cx_${cx_num}_2"/>
      </condition>
    </extension>




  Now after pressing "*2":

  * all parties which joined subconference can speak to/hear each other,

  * after pressing "2" they leave sub and return to main conference,

  * other participants of main conference hear no beeps - joining subconf is like "mute" and returning to main conf like "unmute"



  Regards,



  Mariusz



  Dnia 18-01-2011 o godz. 18:33 Mariusz Czulada napisa?(a):

    Michael, Brian,



    The "hacked" version i used was not hacked indeed. I just tried to correct the patch I send to Jira, but was totally unrelated to this case. Now I use 1.0.7 with default configuration. Then, in conf/dialplan/default folder I added following file (subtest.xml):



    <include>
        <extension name="test_subconf">
          <condition field="destination_number" expression="^(34\d{2})$">
            <action application="set" data="confX_num=$1"/>
            <action application="answer"/>
            <action application="transfer" data="confX_join"/>
          </condition>
        </extension>
        <extension name="confX_join">
          <condition field="destination_number" expression="^confX_join$">
            <action application="bind_meta_app" data="1 a s execute_extension::subX_1"/>
            <action application="bind_meta_app" data="2 a s execute_extension::subX_2"/>
            <action application="conference" data="confX_${confX_num}"/>
          </condition>
        </extension>
        <extension name="subX_1">
          <condition field="destination_number" expression="^subX_1$">
            <action application="set" data="pass_rfc2833=true"/>
            <action application="conference" data="confX_${confX_num}_1"/>
          </condition>
        </extension>
        <extension name="subX_2">
          <condition field="destination_number" expression="^subX_2$">
            <action application="set" data="pass_rfc2833=true"/>
            <action application="conference" data="confX_${confX_num}_2"/>
          </condition>
        </extension>
    </include>




    Feature I expected to achieve was to allow some users to enter sub conference (i.e. "confX_3400_1") to discuss something on side, then get back to the main one ("confX_3400"). During this time other participants should treat them as they are still present in the conference, but probably muted.



    What I observed was:

    * while joining "confX_3400_1" first user hears "you are the only..." message, then MOH

    * when second user joins "confX_3400_1", both hear entry beep, and MOH stops

    * while in sub conf, no DTMF codes are processed; no messages like "switch_rtp.c:3113 RTP RECV DTMF 2:1600" appear on FSW console; no one is able to leave using "#"

    * while in "confX_3400_1" no one can be hear; just like they were muted

    * while in "confX_3400_1" they are still listed as "confX_3400"participants

    * after they're kicked from "confX_3400_1" from  the console, they are again audible by other participants of "confX_3400", without any propt or beep, just like the were unmuted; and can use *1 again to enter sub conf.



    So at this moment I have two problems:

    * User cannot use "#" even if it is defined as a "hangup" action in "default" configuration profile (used here).

    * Users cannot hear each other (or they cannot talk to each other, but the result is the same).



    My question is: is there something I missed in a dialplan or this is the way freeswitch works now?



    Regards,



    Mariusz



    Dnia 14-01-2011 o godz. 19:39 Michael Collins napisa?(a):

      Mariusz, 


      After moving from the main conference to the second conference, when the user presses digits, do any of them work? The way to test would be to call in to the first conference then press #1 to go to the sub conference, then press 0 a few times to see if you mute/unmute. Also, try pressing  1, 2, 3, etc. and watch the FS console. Be sure that you are in debug mode on the console. (If you use fs_cli then you will be at loglevel debug by default.)


      Question - what customizations have you made? I see that your version is "hacked" so most likely you've done some tinkering. One thing you can do to test is to update to latest git and don't do any "hacking" to the code. Test your dialplan and see if the DTMFs work as expected. If your DTMFs work on a plain install then you know that your customizations are doing something. If not, then pastebin your output (pastebin.freeswitch.org) and we'll take a look.


      -MC


      On Fri, Jan 14, 2011 at 8:31 AM, Mariusz Czulada <manieq at wp.eu> wrote:

        Hi All,

        I'm trying to implement subconference feature, to allow some
        participants go (or rather "gosub") to a another conference room to
        discuss something then get back to the main conference. I use FreeSWITCH
        Version 1.0.head (hacked-20110112T172836Z). Bellow some configure
        extracts:

        >From conference.conf.xml:
         ...
         <caller-controls>
           ...
           <group name="just_exit">
             <control action="hangup" digits="#"/>
             <control action="hangup" digits="1"/>
             <control action="mute" digits="0"/>
           </group>
           ...
         </caller-controls>
         ...
         <profiles>
           ..
           <profile name="confX_M">
             ...
             <param name="caller-controls" value="just_exit"/>
             ...
           </profile>
           <profile name="confX_S">
             ...
             <param name="caller-controls" value="just_exit"/>
             ...
           </profile>
         </profiles>
         ...


        >From dialplan (own) public/confX.conf.xml:
           ...
           <extension name="confX_join">
             <condition field="destination_number" expression="^confX_join$"/>
             <condition field="${confX_num}" expression="^1(\d{4})$">
               <action application="bind_meta_app" data="1 a s
        execute_extension::subX_1"/>
               <action application="bind_meta_app" data="2 a s
        execute_extension::subX_2"/>
               <action application="bind_meta_app" data="3 a s
        execute_extension::subX_3"/>
               <action application="conference" data="confX_${cpnfX_num}@confX_M"/>
             </condition>
           </extension>
           ...
           <extension name="subX_1">
             <condition field="destination_number" expression="^subX_1$">
               <action application="set" data="pass_rfc2833=true"/>
               <action application="conference" data="confX_${cpnfX_num}_1 at confX_S"/>
             </condition>
           </extension>
           <extension name="subX_2">
             <condition field="destination_number" expression="^subX_2$">
               <action application="set" data="pass_rfc2833=true"/>
               <action application="conference" data="confX_${cpnfX_num}_2 at confX_S"/>
             </condition>
           </extension>
           <extension name="subX_3">
             <condition field="destination_number" expression="^subX_3$">
               <action application="set" data="pass_rfc2833=true"/>
               <action application="conference" data="confX_${cpnfX_num}_3 at confX_S"/>
             </condition>
           </extension>
           ...


        During connection user is first handled by IVR, where he enters
        conference number (assume: "1234") which is stored in "confX_num"
        variable. Then call is transfered to "confX_join". Here I assign actions
        for *1, *2 and *3 DTMF sequences then join main conference
        ("confX_1234_M"). When some user presses i.e. "*2" then he also enters
        subconf 2 ("confX_1234_2"). Those users attached to "confX_1234_2" are
        at the same time still attached also to "confX_1234_M", but found muted
        and deaf to other participants. Then, leaving subconference they will be
        back "alive" in master conference, which I tested by kicking them from
        "confX_1234_2" manually.

        Now my problem begins. Then user is on "confX_1234_M" dialing "#" forces
        user to leave conference, as defined in "just-exit" controls. But when
        user enters also "confX_1234_2" DTMF codes are no longer processed,
        neither by master conference nor by subconference. I set
        "pass_rfc2833=true" but it is not helpful. Did I missed to set something
        [1] in the dialplan or other config files, or [2] actually no DTMF codes
        can be processed while in "*2" leg? And if [2]==true: would it be
        possible to develop such DTMF handling?

        BTW: DTMFs are not processed while in subconference but if user is
        kicked off from subconf back to main conference, one can use DTMF codes
        again to (un)mute (0), join sub (*2) or leave (# or 1).

        TIA for your help or tips.

        Regards,
        Mariusz



        _______________________________________________
        FreeSWITCH-users mailing list
        FreeSWITCH-users at 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 15 21:47:06 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 15 Apr 2011 13:47:06 -0400
Subject: [Freeswitch-users] rtp-timeout-sec
Message-ID: <7A239E3512DD4B679BA49F7694D3AF1E@e1705>

Is it possible to set rtp_timeout_sec in dialplan and if yes to have a different rtp_timeout_sec value for the originator leg
and the callee leg ?

thanks

Franck
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From dnotivol at gmail.com  Fri Apr 15 22:11:04 2011
From: dnotivol at gmail.com (David Notivol)
Date: Fri, 15 Apr 2011 20:11:04 +0200
Subject: [Freeswitch-users] Non-transcoding G729 calls using Sangoma D100
In-Reply-To: <BANLkTi=h5n2kgKpTi0jmKPfAcunof+VeLw@mail.gmail.com>
References: <BANLkTimJiR=NnUNL4KtYZOSuQpLt9Jru7w@mail.gmail.com>
	<BANLkTi=h5n2kgKpTi0jmKPfAcunof+VeLw@mail.gmail.com>
Message-ID: <BANLkTin8cJrDw3q=BmjZcA=FcPw7PPSAyw@mail.gmail.com>

Thanks Michael,

We're not recording the calls passing through the D100. But we're using the
FS to play some IVR's, so we're playing audios and listening for DTMF tones.

The point is, if a G729 call comes into the FS, and we play some G729 audio
files, no transcoding should be needed. But, as far as I know, mod_g729 and
mod_sangoma_g729 can't be loaded at the same time, so this G729-G729 call
uses a Sangoma D100 session.

I don't know if there's a way of configuring FS to avoid wasting D100
resources.

Regards,
David Notivol

2011/4/15 Michael Collins <msc at freeswitch.org>

> Are you attempting to record the calls that are passing through the D100?
> The reason I ask is that if you are doing anything with media, like
> recording the call or playing MOH, etc. to either leg then it will need to
> do transcoding.
>
> Let's ask Moises Silva at Sangoma. Moy: can you comment on this?
> Thanks,
> MC
>
> On Thu, Apr 14, 2011 at 4:32 AM, David Notivol <dnotivol at gmail.com> wrote:
>
>> Hi all,
>>
>> I'm using a Sangoma D100 card for transcoding with FreeSwitch. It seems
>> the normal behavior is a call not transcoding (g729-g729) is using a session
>> in the sangoma card, although it reports the call is not encoding nor
>> decoding.
>>
>> Since we can't load mod_g729 and mod_sangoma_g729 at the same time; do you
>> know if is there any way to avoid g729-g729 calls using Sangoma sessions?
>>
>> Thanks in advance.
>>
>>
>> Regards,
>> David Notivol
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 15 22:17:52 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 15 Apr 2011 14:17:52 -0400
Subject: [Freeswitch-users] Non-transcoding G729 calls using Sangoma D100
In-Reply-To: <BANLkTin8cJrDw3q=BmjZcA=FcPw7PPSAyw@mail.gmail.com>
References: <BANLkTimJiR=NnUNL4KtYZOSuQpLt9Jru7w@mail.gmail.com>
	<BANLkTi=h5n2kgKpTi0jmKPfAcunof+VeLw@mail.gmail.com>
	<BANLkTin8cJrDw3q=BmjZcA=FcPw7PPSAyw@mail.gmail.com>
Message-ID: <BANLkTin7WcG7vhv_PeG=fGVNqq2SporUYQ@mail.gmail.com>

On Fri, Apr 15, 2011 at 2:11 PM, David Notivol <dnotivol at gmail.com> wrote:

> Thanks Michael,
>
> We're not recording the calls passing through the D100. But we're using the
> FS to play some IVR's, so we're playing audios and listening for DTMF tones.
>
> The point is, if a G729 call comes into the FS, and we play some G729 audio
> files, no transcoding should be needed. But, as far as I know, mod_g729 and
> mod_sangoma_g729 can't be loaded at the same time, so this G729-G729 call
> uses a Sangoma D100 session.
>
> I don't know if there's a way of configuring FS to avoid wasting D100
> resources.
>

As long as the decode/encode count is 0, no resources are wasted. The
hardware resource is allocated only on first use. Even when you can see
"sessions" created, no hardware resource has been created until FreeSWITCH
requires something to encode or decode.

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
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From dnotivol at gmail.com  Fri Apr 15 22:29:45 2011
From: dnotivol at gmail.com (David Notivol)
Date: Fri, 15 Apr 2011 20:29:45 +0200
Subject: [Freeswitch-users] Non-transcoding G729 calls using Sangoma D100
In-Reply-To: <BANLkTin7WcG7vhv_PeG=fGVNqq2SporUYQ@mail.gmail.com>
References: <BANLkTimJiR=NnUNL4KtYZOSuQpLt9Jru7w@mail.gmail.com>
	<BANLkTi=h5n2kgKpTi0jmKPfAcunof+VeLw@mail.gmail.com>
	<BANLkTin8cJrDw3q=BmjZcA=FcPw7PPSAyw@mail.gmail.com>
	<BANLkTin7WcG7vhv_PeG=fGVNqq2SporUYQ@mail.gmail.com>
Message-ID: <BANLkTini20b9EdAkk7WZhXRWrcW6zswj-g@mail.gmail.com>

Thanks Moises for the clarification. That sounds good.

I saw the call was neither decoding nor encoding, but I thought it was
counted for the resources limit.

Regards,
David Notivol.

2011/4/15 Moises Silva <moises.silva at gmail.com>

> On Fri, Apr 15, 2011 at 2:11 PM, David Notivol <dnotivol at gmail.com> wrote:
>
>> Thanks Michael,
>>
>> We're not recording the calls passing through the D100. But we're using
>> the FS to play some IVR's, so we're playing audios and listening for DTMF
>> tones.
>>
>> The point is, if a G729 call comes into the FS, and we play some G729
>> audio files, no transcoding should be needed. But, as far as I know,
>> mod_g729 and mod_sangoma_g729 can't be loaded at the same time, so this
>> G729-G729 call uses a Sangoma D100 session.
>>
>> I don't know if there's a way of configuring FS to avoid wasting D100
>> resources.
>>
>
> As long as the decode/encode count is 0, no resources are wasted. The
> hardware resource is allocated only on first use. Even when you can see
> "sessions" created, no hardware resource has been created until FreeSWITCH
> requires something to encode or decode.
>
> 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
>
>
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From msc at freeswitch.org  Fri Apr 15 22:23:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 15 Apr 2011 11:23:33 -0700
Subject: [Freeswitch-users] rtp-timeout-sec
In-Reply-To: <7A239E3512DD4B679BA49F7694D3AF1E@e1705>
References: <7A239E3512DD4B679BA49F7694D3AF1E@e1705>
Message-ID: <BANLkTinOc-Wu2MkF2Td4Ob2Xyc1UJ34HLQ@mail.gmail.com>

Good question. Try it and tell us if it works! ;) Be sure to export with
nolocal: or set it inside {} on your bridge when you are testing it on the B
leg.
-MC

On Fri, Apr 15, 2011 at 10:47 AM, Madovsky <infos at madovsky.org> wrote:

>  Is it possible to set rtp_timeout_sec in dialplan and if yes to have a
> different rtp_timeout_sec value for the originator leg
> and the callee leg ?
>
> thanks
>
> Franck
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>
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From infos at madovsky.org  Fri Apr 15 23:28:12 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 15 Apr 2011 15:28:12 -0400
Subject: [Freeswitch-users] rtp-timeout-sec
References: <7A239E3512DD4B679BA49F7694D3AF1E@e1705>
	<BANLkTinOc-Wu2MkF2Td4Ob2Xyc1UJ34HLQ@mail.gmail.com>
Message-ID: <787FF2161C6644B9AF3A28F39DC18541@e1705>

Hi Mike,

doesnt' seem to work on B leg, but works on A leg.

Thanks

  ----- Original Message ----- 
  From: Michael Collins 
  To: FreeSWITCH Users Help 
  Sent: Friday, April 15, 2011 2:23 PM
  Subject: Re: [Freeswitch-users] rtp-timeout-sec


  Good question. Try it and tell us if it works! ;) Be sure to export with nolocal: or set it inside {} on your bridge when you are testing it on the B leg.
  -MC


  On Fri, Apr 15, 2011 at 10:47 AM, Madovsky <infos at madovsky.org> wrote:

    Is it possible to set rtp_timeout_sec in dialplan and if yes to have a different rtp_timeout_sec value for the originator leg
    and the callee leg ?

    thanks

    Franck

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






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From philipp.kempgen at amooma.de  Sat Apr 16 00:42:30 2011
From: philipp.kempgen at amooma.de (Philipp Kempgen)
Date: Fri, 15 Apr 2011 22:42:30 +0200
Subject: [Freeswitch-users] mod_xml_curl,
 sofia_reg.c:1872 Can't find user ...
In-Reply-To: <BANLkTimBad6ogmWhWDcokpSKqGcC5_C=7Q@mail.gmail.com>
References: <4DA747CC.3080509@amooma.de>
	<BANLkTimBad6ogmWhWDcokpSKqGcC5_C=7Q@mail.gmail.com>
Message-ID: <4DA8ADB6.7050500@amooma.de>

Michael,

Michael Collins wrote:
> Have you tried to add the <groups> and <group> XML elements?

Thanks a ton! That seems to do the trick. I knew it was something
simple that I was missing.

I was under the impression that groups were optional because in
the example at
http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Basic_User
there are no groups at all, and Mod-XML-Curl seemed to require the
same XML structure as in static XML files.

http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups
mentions groups and XML-Curl but also says "Using groups is optional
-- you can put your users straight into the domain section if you
desire."

Thanks again!
  Philipp


> On Thu, Apr 14, 2011 at 12:15 PM, Philipp Kempgen <philipp.kempgen at amooma.de
>> wrote:
> 
>> Hi,
>>
>> I'm playing with mod_xml_curl per
>> http://wiki.freeswitch.org/wiki/Mod_xml_curl#bindings.3D.22directory.22
>> but having some problems.
>>
>> Softphone (username "11") sends a REGISTER to FreeSwitch (1.0.6)
>> at 192.168.65.133.
>>
>> xml_curl.conf.xml:
>> ---------------------------------------------------------------
>>  <binding name="users directory">
>>    <param name="gateway-url" value="
>> http://127.0.0.1:3000/freeswitch-directory-entries/search.xml"
>> bindings="directory" />
>>  </binding>
>> ---------------------------------------------------------------
>>
>> FS sends a lookup request to the web service. Parameters:
>> ---------------------------------------------------------------
>> section=directory
>> ...
>> key=id
>> user=11
>> domain=192.168.65.133
>> ---------------------------------------------------------------
>>
>> Fine.
>>
>> The response is:
>> ---------------------------------------------------------------
>> <?xml version="1.0" encoding="UTF-8" standalone="yes"?>
>> <document type="freeswitch/xml">
>>  <section name="directory">
>>    <domain name="192.168.65.133">
>>      <users>
>>        <user id="11" number-alias="1111">
>>          <params>
>>            <param name="dial-string" value="${sofia_contact(
>> 11 at 192.168.65.133)}" />
>>            <param name="password" value="123" />
>>            <param name="vm-password" value="417906717460606223" />
>>          </params>
>>          <variables>
>>            <variable name="accountcode" value="11" />
>>            <variable name="effective_caller_id_number" value="foo" />
>>            <variable name="effective_caller_id_name" value="Foo, Foo" />
>>            <variable name="outbound_caller_id_number" value="foo" />
>>            <variable name="outbound_caller_id_name" value="Foo, Foo" />
>>            <variable name="user_context" value="default" />
>>            <variable name="toll_allow" value="domestic,international,local"
>> />
>>            <variable name="callgroup" value="users" />
>>          </variables>
>>        </user>
>>      </users>
>>    </domain>
>>  </section>
>> </document>
>> ---------------------------------------------------------------
>>
>> But I get a complaint on the console:
>> [WARNING] sofia_reg.c:1872 Can't find user [11 at 192.168.65.133]
>> You must define a domain called '192.168.65.133' in your directory
>> and add a user with the id="11" attribute and you must configure
>> your device to use the proper domain in it's authentication
>> credentials.
>>
>> What can it be?
>>
>> A domain with name="192.168.65.133" and a user with id="11" is in
>> the response, so why doesn't sofia find it?



From infos at madovsky.org  Sat Apr 16 00:57:36 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 15 Apr 2011 16:57:36 -0400
Subject: [Freeswitch-users] txfax problems last git
Message-ID: <302AAC028FEB4B84965D022F1322010A@e1705>

before update it was working well

here is the log

2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 6
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Retry number 2
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 0
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 4
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 DCS:
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... 0...= 3G mobile network: Not set
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..1.= Receive fax: Set
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ..10 00..= Selected data signalling rate: V.17 14400bps
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   1... ....= 2-D coding: Set
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..00= Recording width: 215mm +- 1%
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... 10..= Recording length: Unlimited
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .111 ....= Minimum scan line time: 0ms
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   0... ....= Extension indicator: Not set
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from state 6 to 4
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Sending ident 'boophone.com'
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Tx:  TSI without final frame tag
2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Tx:  ff 03 43 6d 6f 63 2e 65 6e 6f 68 70 6f 6f 62 20 20 20 20 20 20 20 20
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 4
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 DCS:
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... 0...= 3G mobile network: Not set
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..1.= Receive fax: Set
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ..10 00..= Selected data signalling rate: V.17 14400bps
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   1... ....= 2-D coding: Set
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..00= Recording width: 215mm +- 1%
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... 10..= Recording length: Unlimited
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .111 ....= Minimum scan line time: 0ms
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   0... ....= Extension indicator: Not set
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Tx:  DCS with final frame tag
2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Tx:  ff 13 83 00 e2 78
2011-04-15 16:54:09.546048 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 4
2011-04-15 16:54:09.626076 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 4
2011-04-15 16:54:09.626076 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from state 4 to 5
2011-04-15 16:54:09.626076 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_C_NON_ECM_TX
2011-04-15 16:54:09.626076 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 0
2011-04-15 16:54:09.626076 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 0
2011-04-15 16:54:09.626076 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 7
2011-04-15 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_C_NON_ECM_TX, state 5
2011-04-15 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from phase T30_PHASE_C_NON_ECM_TX to T30_PHASE_B_RX
2011-04-15 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 4
2011-04-15 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 0
2011-04-15 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Start T4
2011-04-15 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from state 5 to 6
2011-04-15 16:54:13.166347 [DEBUG] mod_nibblebill.c:572 Received request via SESSION_HEARTBEAT!
2011-04-15 16:54:13.166347 [DEBUG] mod_nibblebill.c:433 Attempting to bill at $0.17304 per minute to account 9999999999999
2011-04-15 16:54:13.166347 [DEBUG] mod_nibblebill.c:491 30 seconds passed since last bill time of 2011-04-15 16:53:43
2011-04-15 16:54:13.166347 [DEBUG] mod_nibblebill.c:498 Billing $0.086548 to 9999999999999 (Call: 1132765120 at 67.205.80.135 / 0.000009 so far)
2011-04-15 16:54:13.166347 [DEBUG] mod_nibblebill.c:321 Doing update query
[UPDATE accounts SET cash=cash-0.086548 WHERE id='9999999999999']
2011-04-15 16:54:13.288037 [DEBUG] mod_nibblebill.c:366 Doing lookup query
[SELECT cash AS nibble_balance FROM accounts WHERE id='9999999999999']
2011-04-15 16:54:13.293107 [DEBUG] mod_nibblebill.c:376 Retrieved current balance for account 9999999999999 (balance = 6.655555)
2011-04-15 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 6
2011-04-15 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Too many retries. Giving up.
2011-04-15 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX
2011-04-15 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 0
2011-04-15 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 4
2011-04-15 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from state 6 to 3
2011-04-15 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Tx:  DCN with final frame tag
2011-04-15 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Tx:  ff 13 fb
2011-04-15 16:54:17.165849 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2011-04-15 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2011-04-15 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Disconnecting
2011-04-15 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E
2011-04-15 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 0
2011-04-15 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 1
2011-04-15 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from state 3 to 2
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_E, state 2
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:333 ==============================================================================
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:345 Fax processing not successful - result (20) Received no response to DCS or TCF.
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:350 Remote station id: 33456809528
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:351 Local station id:  boophone.com
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:352 Pages transferred: 0
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:354 Total fax pages:   1
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:355 Image resolution:  8031x7700
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:356 Transfer Rate:     14400
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:358 ECM status         off
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:359 remote country:
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:360 remote vendor:
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:361 remote model:
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:363 ==============================================================================
2011-04-15 16:54:18.245661 [DEBUG] switch_core_session.c:954 Send signal sofia/external/0033456809528 [BREAK]
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from state 2 to 32
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from phase T30_PHASE_E to T30_PHASE_CALL_FINISHED
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 9
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX FAX exchange complete
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 9
2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX FAX exchange complete
2011-04-15 16:54:18.265918 [DEBUG] switch_core_codec.c:141 sofia/external/0033456809528 Restore previous codec PCMU:0.



my dialplan

                <extension name="sendfax">
                        <condition field="destination_number" expression="^sendfax$">
                                <action application="set" data="fax_enable_t38_request=false"/>
                                <action application="set" data="fax_enable_t38=true"/>
                                <action application="txfax" data="${fax_path}"/>
                                <action application="hangup"/>
                        </condition>
                </extension>


Thanks !

Franck



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From errotan at elder.hu  Sat Apr 16 01:13:37 2011
From: errotan at elder.hu (=?iso-8859-1?q?Pusk=E1s_Zsolt?=)
Date: Fri, 15 Apr 2011 23:13:37 +0200
Subject: [Freeswitch-users] txfax problems last git
In-Reply-To: <302AAC028FEB4B84965D022F1322010A@e1705>
References: <302AAC028FEB4B84965D022F1322010A@e1705>
Message-ID: <201104152313.37393.errotan@elder.hu>

Hi Madovsky!

Please use http://jira.freeswitch.org/ to report problems.

Thank you.

2011. ?prilis 15. 22:57:36 d?tummal Madovsky az al?bbiakat ?rta:
> before update it was working well
> 
> here is the log
> 
> 2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 T4
> expired in phase T30_PHASE_B_RX, state 6 2011-04-15 16:54:07.685484
> [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Retry number 2 2011-04-15
> 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from
> phase T30_PHASE_B_RX to T30_PHASE_B_TX 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW FAX Set rx type 0 2011-04-15 16:54:07.685484
> [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 4 2011-04-15
> 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 DCS: 2011-04-15
> 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ...0= Store
> and forward Internet fax (T.37): Not set 2011-04-15 16:54:07.685484
> [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... .0..= Real-time Internet
> fax (T.38): Not set 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   .... 0...= 3G mobile network: Not set
> 2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ....
> ..1.= Receive fax: Set 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   ..10 00..= Selected data signalling
> rate: V.17 14400bps 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or
> 200x200pels/25.4mm: Set 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   1... ....= 2-D coding: Set 2011-04-15
> 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..00=
> Recording width: 215mm +- 1% 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   .... 10..= Recording length: Unlimited
> 2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .111
> ....= Minimum scan line time: 0ms 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   0... ....= Extension indicator: Not set
> 2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
> Changing from state 6 to 4 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Sending ident 'boophone.com' 2011-04-15
> 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Tx:  TSI without
> final frame tag 2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296
> FLOW T.30 Tx:  ff 03 43 6d 6f 63 2e 65 6e 6f 68 70 6f 6f 62 20 20 20 20 20
> 20 20 20 2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW
> T.30 Send complete in phase T30_PHASE_B_TX, state 4 2011-04-15
> 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 DCS: 2011-04-15
> 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ...0= Store
> and forward Internet fax (T.37): Not set 2011-04-15 16:54:09.265684
> [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... .0..= Real-time Internet
> fax (T.38): Not set 2011-04-15 16:54:09.265684 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   .... 0...= 3G mobile network: Not set
> 2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ....
> ..1.= Receive fax: Set 2011-04-15 16:54:09.265684 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   ..10 00..= Selected data signalling
> rate: V.17 14400bps 2011-04-15 16:54:09.265684 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or
> 200x200pels/25.4mm: Set 2011-04-15 16:54:09.265684 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   1... ....= 2-D coding: Set 2011-04-15
> 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..00=
> Recording width: 215mm +- 1% 2011-04-15 16:54:09.265684 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   .... 10..= Recording length: Unlimited
> 2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .111
> ....= Minimum scan line time: 0ms 2011-04-15 16:54:09.265684 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   0... ....= Extension indicator: Not set
> 2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Tx: 
> DCS with final frame tag 2011-04-15 16:54:09.265684 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Tx:  ff 13 83 00 e2 78 2011-04-15
> 16:54:09.546048 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send complete in
> phase T30_PHASE_B_TX, state 4 2011-04-15 16:54:09.626076 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_B_TX,
> state 4 2011-04-15 16:54:09.626076 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
> Changing from state 4 to 5 2011-04-15 16:54:09.626076 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Changing from phase T30_PHASE_B_TX to
> T30_PHASE_C_NON_ECM_TX 2011-04-15 16:54:09.626076 [DEBUG]
> mod_spandsp_fax.c:296 FLOW FAX Set rx type 0 2011-04-15 16:54:09.626076
> [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 0 2011-04-15
> 16:54:09.626076 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 7
> 2011-04-15 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send
> complete in phase T30_PHASE_C_NON_ECM_TX, state 5 2011-04-15
> 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from
> phase T30_PHASE_C_NON_ECM_TX to T30_PHASE_B_RX 2011-04-15 16:54:12.645314
> [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 4 2011-04-15
> 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 0
> 2011-04-15 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Start
> T4 2011-04-15 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
> Changing from state 5 to 6 2011-04-15 16:54:13.166347 [DEBUG]
> mod_nibblebill.c:572 Received request via SESSION_HEARTBEAT! 2011-04-15
> 16:54:13.166347 [DEBUG] mod_nibblebill.c:433 Attempting to bill at
> $0.17304 per minute to account 9999999999999 2011-04-15 16:54:13.166347
> [DEBUG] mod_nibblebill.c:491 30 seconds passed since last bill time of
> 2011-04-15 16:53:43 2011-04-15 16:54:13.166347 [DEBUG]
> mod_nibblebill.c:498 Billing $0.086548 to 9999999999999 (Call:
> 1132765120 at 67.205.80.135 / 0.000009 so far) 2011-04-15 16:54:13.166347
> [DEBUG] mod_nibblebill.c:321 Doing update query [UPDATE accounts SET
> cash=cash-0.086548 WHERE id='9999999999999']
> 2011-04-15 16:54:13.288037 [DEBUG] mod_nibblebill.c:366 Doing lookup query
> [SELECT cash AS nibble_balance FROM accounts WHERE id='9999999999999']
> 2011-04-15 16:54:13.293107 [DEBUG] mod_nibblebill.c:376 Retrieved current
> balance for account 9999999999999 (balance = 6.655555) 2011-04-15
> 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 T4 expired in
> phase T30_PHASE_B_RX, state 6 2011-04-15 16:54:16.105756 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Too many retries. Giving up. 2011-04-15
> 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from
> phase T30_PHASE_B_RX to T30_PHASE_D_TX 2011-04-15 16:54:16.105756 [DEBUG]
> mod_spandsp_fax.c:296 FLOW FAX Set rx type 0 2011-04-15 16:54:16.105756
> [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 4 2011-04-15
> 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from
> state 6 to 3 2011-04-15 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW
> T.30 Tx:  DCN with final frame tag 2011-04-15 16:54:16.105756 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Tx:  ff 13 fb 2011-04-15 16:54:17.165849
> [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase
> T30_PHASE_D_TX, state 3 2011-04-15 16:54:17.245860 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_D_TX,
> state 3 2011-04-15 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
> Disconnecting 2011-04-15 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296
> FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E 2011-04-15
> 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 0
> 2011-04-15 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx
> type 1 2011-04-15 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
> Changing from state 3 to 2 2011-04-15 16:54:18.245661 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_E, state
> 2 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:333
> ==========================================================================
> ==== 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:345 Fax
> processing not successful - result (20) Received no response to DCS or
> TCF. 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:350 Remote
> station id: 33456809528 2011-04-15 16:54:18.245661 [DEBUG]
> mod_spandsp_fax.c:351 Local station id:  boophone.com 2011-04-15
> 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:352 Pages transferred: 0
> 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:354 Total fax pages: 
>  1 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:355 Image
> resolution:  8031x7700 2011-04-15 16:54:18.245661 [DEBUG]
> mod_spandsp_fax.c:356 Transfer Rate:     14400 2011-04-15 16:54:18.245661
> [DEBUG] mod_spandsp_fax.c:358 ECM status         off 2011-04-15
> 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:359 remote country: 2011-04-15
> 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:360 remote vendor: 2011-04-15
> 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:361 remote model: 2011-04-15
> 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:363
> ==========================================================================
> ==== 2011-04-15 16:54:18.245661 [DEBUG] switch_core_session.c:954 Send
> signal sofia/external/0033456809528 [BREAK] 2011-04-15 16:54:18.245661
> [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from state 2 to 32
> 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
> Changing from phase T30_PHASE_E to T30_PHASE_CALL_FINISHED 2011-04-15
> 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 9
> 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX FAX
> exchange complete 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296
> FLOW FAX Set tx type 9 2011-04-15 16:54:18.245661 [DEBUG]
> mod_spandsp_fax.c:296 FLOW FAX FAX exchange complete 2011-04-15
> 16:54:18.265918 [DEBUG] switch_core_codec.c:141
> sofia/external/0033456809528 Restore previous codec PCMU:0.
> 
> 
> 
> my dialplan
> 
>                 <extension name="sendfax">
>                         <condition field="destination_number"
> expression="^sendfax$"> <action application="set"
> data="fax_enable_t38_request=false"/> <action application="set"
> data="fax_enable_t38=true"/> <action application="txfax"
> data="${fax_path}"/> <action application="hangup"/>
>                         </condition>
>                 </extension>
> 
> 
> Thanks !
> 
> Franck


From infos at madovsky.org  Sat Apr 16 01:25:23 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 15 Apr 2011 17:25:23 -0400
Subject: [Freeswitch-users] txfax problems last git
References: <302AAC028FEB4B84965D022F1322010A@e1705>
	<201104152313.37393.errotan@elder.hu>
Message-ID: <0BED76B766B94D8BA6FCD9BD39E0CC65@e1705>

done !

----- Original Message ----- 
From: "Pusk?s Zsolt" <errotan at elder.hu>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, April 15, 2011 5:13 PM
Subject: Re: [Freeswitch-users] txfax problems last git


Hi Madovsky!

Please use http://jira.freeswitch.org/ to report problems.

Thank you.

2011. ?prilis 15. 22:57:36 d?tummal Madovsky az al?bbiakat ?rta:
> before update it was working well
>
> here is the log
>
> 2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 T4
> expired in phase T30_PHASE_B_RX, state 6 2011-04-15 16:54:07.685484
> [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Retry number 2 2011-04-15
> 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from
> phase T30_PHASE_B_RX to T30_PHASE_B_TX 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW FAX Set rx type 0 2011-04-15 16:54:07.685484
> [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 4 2011-04-15
> 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 DCS: 2011-04-15
> 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ...0= Store
> and forward Internet fax (T.37): Not set 2011-04-15 16:54:07.685484
> [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... .0..= Real-time Internet
> fax (T.38): Not set 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   .... 0...= 3G mobile network: Not set
> 2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ....
> ..1.= Receive fax: Set 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   ..10 00..= Selected data signalling
> rate: V.17 14400bps 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or
> 200x200pels/25.4mm: Set 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   1... ....= 2-D coding: Set 2011-04-15
> 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..00=
> Recording width: 215mm +- 1% 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   .... 10..= Recording length: Unlimited
> 2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .111
> ....= Minimum scan line time: 0ms 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   0... ....= Extension indicator: Not set
> 2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
> Changing from state 6 to 4 2011-04-15 16:54:07.685484 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Sending ident 'boophone.com' 2011-04-15
> 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Tx:  TSI without
> final frame tag 2011-04-15 16:54:07.685484 [DEBUG] mod_spandsp_fax.c:296
> FLOW T.30 Tx:  ff 03 43 6d 6f 63 2e 65 6e 6f 68 70 6f 6f 62 20 20 20 20 20
> 20 20 20 2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW
> T.30 Send complete in phase T30_PHASE_B_TX, state 4 2011-04-15
> 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 DCS: 2011-04-15
> 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ...0= Store
> and forward Internet fax (T.37): Not set 2011-04-15 16:54:09.265684
> [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... .0..= Real-time Internet
> fax (T.38): Not set 2011-04-15 16:54:09.265684 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   .... 0...= 3G mobile network: Not set
> 2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   ....
> ..1.= Receive fax: Set 2011-04-15 16:54:09.265684 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   ..10 00..= Selected data signalling
> rate: V.17 14400bps 2011-04-15 16:54:09.265684 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or
> 200x200pels/25.4mm: Set 2011-04-15 16:54:09.265684 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   1... ....= 2-D coding: Set 2011-04-15
> 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .... ..00=
> Recording width: 215mm +- 1% 2011-04-15 16:54:09.265684 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   .... 10..= Recording length: Unlimited
> 2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30   .111
> ....= Minimum scan line time: 0ms 2011-04-15 16:54:09.265684 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30   0... ....= Extension indicator: Not set
> 2011-04-15 16:54:09.265684 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Tx:
> DCS with final frame tag 2011-04-15 16:54:09.265684 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Tx:  ff 13 83 00 e2 78 2011-04-15
> 16:54:09.546048 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send complete in
> phase T30_PHASE_B_TX, state 4 2011-04-15 16:54:09.626076 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_B_TX,
> state 4 2011-04-15 16:54:09.626076 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
> Changing from state 4 to 5 2011-04-15 16:54:09.626076 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Changing from phase T30_PHASE_B_TX to
> T30_PHASE_C_NON_ECM_TX 2011-04-15 16:54:09.626076 [DEBUG]
> mod_spandsp_fax.c:296 FLOW FAX Set rx type 0 2011-04-15 16:54:09.626076
> [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 0 2011-04-15
> 16:54:09.626076 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 7
> 2011-04-15 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send
> complete in phase T30_PHASE_C_NON_ECM_TX, state 5 2011-04-15
> 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from
> phase T30_PHASE_C_NON_ECM_TX to T30_PHASE_B_RX 2011-04-15 16:54:12.645314
> [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 4 2011-04-15
> 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 0
> 2011-04-15 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Start
> T4 2011-04-15 16:54:12.645314 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
> Changing from state 5 to 6 2011-04-15 16:54:13.166347 [DEBUG]
> mod_nibblebill.c:572 Received request via SESSION_HEARTBEAT! 2011-04-15
> 16:54:13.166347 [DEBUG] mod_nibblebill.c:433 Attempting to bill at
> $0.17304 per minute to account 9999999999999 2011-04-15 16:54:13.166347
> [DEBUG] mod_nibblebill.c:491 30 seconds passed since last bill time of
> 2011-04-15 16:53:43 2011-04-15 16:54:13.166347 [DEBUG]
> mod_nibblebill.c:498 Billing $0.086548 to 9999999999999 (Call:
> 1132765120 at 67.205.80.135 / 0.000009 so far) 2011-04-15 16:54:13.166347
> [DEBUG] mod_nibblebill.c:321 Doing update query [UPDATE accounts SET
> cash=cash-0.086548 WHERE id='9999999999999']
> 2011-04-15 16:54:13.288037 [DEBUG] mod_nibblebill.c:366 Doing lookup query
> [SELECT cash AS nibble_balance FROM accounts WHERE id='9999999999999']
> 2011-04-15 16:54:13.293107 [DEBUG] mod_nibblebill.c:376 Retrieved current
> balance for account 9999999999999 (balance = 6.655555) 2011-04-15
> 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 T4 expired in
> phase T30_PHASE_B_RX, state 6 2011-04-15 16:54:16.105756 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Too many retries. Giving up. 2011-04-15
> 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from
> phase T30_PHASE_B_RX to T30_PHASE_D_TX 2011-04-15 16:54:16.105756 [DEBUG]
> mod_spandsp_fax.c:296 FLOW FAX Set rx type 0 2011-04-15 16:54:16.105756
> [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx type 4 2011-04-15
> 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from
> state 6 to 3 2011-04-15 16:54:16.105756 [DEBUG] mod_spandsp_fax.c:296 FLOW
> T.30 Tx:  DCN with final frame tag 2011-04-15 16:54:16.105756 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Tx:  ff 13 fb 2011-04-15 16:54:17.165849
> [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase
> T30_PHASE_D_TX, state 3 2011-04-15 16:54:17.245860 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_D_TX,
> state 3 2011-04-15 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
> Disconnecting 2011-04-15 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296
> FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E 2011-04-15
> 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 0
> 2011-04-15 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set tx
> type 1 2011-04-15 16:54:17.245860 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
> Changing from state 3 to 2 2011-04-15 16:54:18.245661 [DEBUG]
> mod_spandsp_fax.c:296 FLOW T.30 Send complete in phase T30_PHASE_E, state
> 2 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:333
> ==========================================================================
> ==== 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:345 Fax
> processing not successful - result (20) Received no response to DCS or
> TCF. 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:350 Remote
> station id: 33456809528 2011-04-15 16:54:18.245661 [DEBUG]
> mod_spandsp_fax.c:351 Local station id:  boophone.com 2011-04-15
> 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:352 Pages transferred: 0
> 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:354 Total fax pages:
>  1 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:355 Image
> resolution:  8031x7700 2011-04-15 16:54:18.245661 [DEBUG]
> mod_spandsp_fax.c:356 Transfer Rate:     14400 2011-04-15 16:54:18.245661
> [DEBUG] mod_spandsp_fax.c:358 ECM status         off 2011-04-15
> 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:359 remote country: 2011-04-15
> 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:360 remote vendor: 2011-04-15
> 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:361 remote model: 2011-04-15
> 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:363
> ==========================================================================
> ==== 2011-04-15 16:54:18.245661 [DEBUG] switch_core_session.c:954 Send
> signal sofia/external/0033456809528 [BREAK] 2011-04-15 16:54:18.245661
> [DEBUG] mod_spandsp_fax.c:296 FLOW T.30 Changing from state 2 to 32
> 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296 FLOW T.30
> Changing from phase T30_PHASE_E to T30_PHASE_CALL_FINISHED 2011-04-15
> 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX Set rx type 9
> 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296 FLOW FAX FAX
> exchange complete 2011-04-15 16:54:18.245661 [DEBUG] mod_spandsp_fax.c:296
> FLOW FAX Set tx type 9 2011-04-15 16:54:18.245661 [DEBUG]
> mod_spandsp_fax.c:296 FLOW FAX FAX exchange complete 2011-04-15
> 16:54:18.265918 [DEBUG] switch_core_codec.c:141
> sofia/external/0033456809528 Restore previous codec PCMU:0.
>
>
>
> my dialplan
>
>                 <extension name="sendfax">
>                         <condition field="destination_number"
> expression="^sendfax$"> <action application="set"
> data="fax_enable_t38_request=false"/> <action application="set"
> data="fax_enable_t38=true"/> <action application="txfax"
> data="${fax_path}"/> <action application="hangup"/>
>                         </condition>
>                 </extension>
>
>
> Thanks !
>
> Franck

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



From singhujjwal at gmail.com  Sat Apr 16 00:17:17 2011
From: singhujjwal at gmail.com (Ujjwal SIngh)
Date: Sat, 16 Apr 2011 01:47:17 +0530
Subject: [Freeswitch-users] Noise heard when a user holds a conference in
	SRTP mode
In-Reply-To: <BANLkTinp3JQkJw3ze4_dEHOT+BzDsEMPAQ@mail.gmail.com>
References: <BANLkTinp3JQkJw3ze4_dEHOT+BzDsEMPAQ@mail.gmail.com>
Message-ID: <BANLkTikhmk+HGwV=kNe7UvpazwmpaJ3YrA@mail.gmail.com>

Hi,

I have FreeSwitch configured as a conference bridge, the
sip_secure_media=true in dialplan and vars.xml.
User A, B and C are in a conference, the media is established in SRTP mode,
when any user presses hold, other members hear noise, while in RTP mode
normal Music on Hold is heard.

Can anybody please help what is going wrong in SRTP case.
I have uploaded FreeSwitch log in pastebin below

http://pastebin.freeswitch.org/16103


Thanks for the help.
Regards,
Ujjwal
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From freeswitch at peely.com  Sat Apr 16 15:57:46 2011
From: freeswitch at peely.com (peely)
Date: Sat, 16 Apr 2011 04:57:46 -0700 (PDT)
Subject: [Freeswitch-users] gentls_cert: Specifying the Country Name?
Message-ID: <1302955066042-6278826.post@n2.nabble.com>

Hi,

I'm trying to generate a certificate request using gentls_cert, only when I
attempt to use the contents of careq.pem getting a commercial certificate I
am instructed the country code is missing.

Is there a way of passing the country code into gentls_cert?


Thanks,


Neil.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/gentls-cert-Specifying-the-Country-Name-tp6278826p6278826.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From vetali100 at gmail.com  Sat Apr 16 16:54:12 2011
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Sat, 16 Apr 2011 15:54:12 +0300
Subject: [Freeswitch-users] can not call between two profile
In-Reply-To: <BANLkTimcqAf+Yj0UUH=KiaW3L6iRWgaufg@mail.gmail.com>
References: <BANLkTimcqAf+Yj0UUH=KiaW3L6iRWgaufg@mail.gmail.com>
Message-ID: <BANLkTinThK7iHddP63aXXUCbiKUXqOKTqQ@mail.gmail.com>

In general, external profile is not for the purpose of registration of your
own subscribers, but to receive calls from OUTSIDE your network.
But to answer you question anyway -
Check your dialplan, probably you are calling from 1000 to the same internal
profile, but you must call to the external profile (since 1010 is registered
there).

Please paste your dialplan to pastebin, and provide the link. Don't paste it
to this mail thread.

use one of the available:

http://pastebin.freeswitch.org/

<http://pastebin.freeswitch.org/>http://www.pastebin.com

http://www.pastebin.ca



2011/4/15 gmail <flyingnn at gmail.com>

> hi!
>
> 1000  register to profile internal.xml, 1010 register to profile
> external.xml
> 1000 and 1010 register successfully.
>
> 1000 can not call 1010, [ERR] switch_ivr_originate.c:2638 Cannot create
> outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
> 1010 call 1000 ok.
> that is why?
>
> 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 daletrub at gmail.com  Sat Apr 16 22:33:09 2011
From: daletrub at gmail.com (Dale Trub)
Date: Sat, 16 Apr 2011 11:33:09 -0700
Subject: [Freeswitch-users] conference play / stop / PAUSE/RESUME?
Message-ID: <BANLkTi=9672+WLZON0teCJLa-CFsg4E88w@mail.gmail.com>

We let the conference host play a file to the conference and want to give
them a control to "pause" such a file, and then resume it.

Has anyone found a way to do this?

Would someone create something like that for a modest bounty?  We could
incorp in the code we use, as well as submit to the project overall.

-Dale
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From bwibowo at gmail.com  Sun Apr 17 02:11:08 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Sun, 17 Apr 2011 05:11:08 +0700
Subject: [Freeswitch-users] webphone app
In-Reply-To: <BANLkTi=73yyZ4Chh8OwfiJhNXAVXnrZ8MA@mail.gmail.com>
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
	<32EF1A658EFC4E5393D8D1A3A486DA31@e1705>
	<BANLkTikdW-5-hDz9A+0mDid_KNEQNhzf6Q@mail.gmail.com>
	<BANLkTi=73yyZ4Chh8OwfiJhNXAVXnrZ8MA@mail.gmail.com>
Message-ID: <BANLkTi=_tM67uu0MJBsWq5NLnVOEbPHm0w@mail.gmail.com>

i try that, but got confussion on  installing the application.
http://code.google.com/p/bigbluebutton/wiki/Red5Phone also not work.
please give detail installation method


On Thu, Apr 7, 2011 at 8:17 AM, Moe Navid <manavid at gmail.com> wrote:

> 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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Sun Apr 17 04:07:28 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 16 Apr 2011 20:07:28 -0400
Subject: [Freeswitch-users] last git voicemail problem
Message-ID: <6BBA040CD70F424886D6E1969F0BADC5@e1705>

the voicemail doesn't repeat that the message is too short
when no voice is coming. instead there's undefinetly a silence until
it reaches the max lenght

Thanks

Franck
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From anthony.minessale at gmail.com  Sun Apr 17 04:30:07 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 16 Apr 2011 19:30:07 -0500
Subject: [Freeswitch-users] last git voicemail problem
In-Reply-To: <6BBA040CD70F424886D6E1969F0BADC5@e1705>
References: <6BBA040CD70F424886D6E1969F0BADC5@e1705>
Message-ID: <BANLkTi=072QB4V5hRJq5BdXpderO-8gDMw@mail.gmail.com>

Jira
On Apr 16, 2011 7:08 PM, "Madovsky" <infos at madovsky.org> wrote:
> the voicemail doesn't repeat that the message is too short
> when no voice is coming. instead there's undefinetly a silence until
> it reaches the max lenght
>
> Thanks
>
> Franck
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From jcasale at activenetwerx.com  Sun Apr 17 05:45:44 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sun, 17 Apr 2011 01:45:44 +0000
Subject: [Freeswitch-users] show registrations missing
Message-ID: <assp.00880913a1.92097A6A775D5147B1078E3F15430B9239C458@prato.activenetwerx.local>

In my openwrt setup, I don't have a `show registrations` in the cli?
Anyone know what provides that functionality?

thanks,
jlc


From fraserredmond at gmail.com  Sun Apr 17 08:34:39 2011
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Sun, 17 Apr 2011 00:34:39 -0400
Subject: [Freeswitch-users] one-way audio problem on some inbound gateways,
 but not others (and no outbound gateways)
Message-ID: <BANLkTinuTg+DwXQ5_fU1K-mE+oqYgH7apQ@mail.gmail.com>

I'm getting a one-way-audio problem on 2 of my inbound gateways.

I have 3 gateways (1800 numbers), all directing into the same dialplan,
which then directs the caller to an extn (answered by a softphone.) One of
the gateways works fine, the other two do not (they used to several months
ago, I'm not exactly sure when they stopped working.)

One of the gateways that doesn't work is used for outbound calls without any
problem.

The pstn caller's audio arrives at the softphone, but the softphone's audio
doesn't make it to the caller. It also doesn't show up in call-recordings.

I've tried 3 different softphones to make sure it wasn't a problem with
that. (The microphone indicators on them indicate they're picking up audio.
Sending a dtmf does go through.)

I've just upgraded to the latest version of freeswitch to make sure it
wasn't that.

Any ideas of what to try?

Cheers,
Fraser
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From brian at freeswitch.org  Sun Apr 17 08:39:55 2011
From: brian at freeswitch.org (Brian Weat)
Date: Sat, 16 Apr 2011 23:39:55 -0500
Subject: [Freeswitch-users] one-way audio problem on some inbound
	gateways, but not others (and no outbound gateways)
In-Reply-To: <BANLkTinuTg+DwXQ5_fU1K-mE+oqYgH7apQ@mail.gmail.com>
References: <BANLkTinuTg+DwXQ5_fU1K-mE+oqYgH7apQ@mail.gmail.com>
Message-ID: <4A1D272B-E12C-4CF5-9149-4DED32E88CCB@freeswitch.org>

have a sip trace?

Sent from my iPhone

On Apr 16, 2011, at 11:34 PM, Fraser Redmond <fraserredmond at gmail.com> wrote:

> Any ideas of what to try?
> 



From brian at freeswitch.org  Sun Apr 17 08:41:26 2011
From: brian at freeswitch.org (Brian Weat)
Date: Sat, 16 Apr 2011 23:41:26 -0500
Subject: [Freeswitch-users] show registrations missing
In-Reply-To: <assp.00880913a1.92097A6A775D5147B1078E3F15430B9239C458@prato.activenetwerx.local>
References: <assp.00880913a1.92097A6A775D5147B1078E3F15430B9239C458@prato.activenetwerx.local>
Message-ID: <BC86D31A-5406-408D-9E0F-DFCD17CBD9E2@freeswitch.org>

sofia status profile xxxx

replace xxxx with profile name.  'sofia status' show all profiles and gateway status.

/b

Sent from my iPhone

On Apr 16, 2011, at 8:45 PM, "Joseph L. Casale" <jcasale at activenetwerx.com> wrote:

> In my openwrt setup, I don't have a `show registrations` in the cli?
> Anyone know what provides that functionality?
> 
> thanks,
> jlc
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From jcasale at activenetwerx.com  Sun Apr 17 08:55:57 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sun, 17 Apr 2011 04:55:57 +0000
Subject: [Freeswitch-users] show registrations missing
In-Reply-To: <BC86D31A-5406-408D-9E0F-DFCD17CBD9E2@freeswitch.org>
References: <92097A6A775D5147B1078E3F15430B9239C458@prato.activenetwerx.local>
	<BC86D31A-5406-408D-9E0F-DFCD17CBD9E2@freeswitch.org>
Message-ID: <assp.008808a7f3.92097A6A775D5147B1078E3F15430B9239EA4D@prato.activenetwerx.local>

>sofia status profile xxxx

Thanks Brian!
jlc


From jcasale at activenetwerx.com  Sun Apr 17 20:45:40 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sun, 17 Apr 2011 16:45:40 +0000
Subject: [Freeswitch-users] Beep on hold
Message-ID: <assp.0088b98154.92097A6A775D5147B1078E3F15430B923A10FD@prato.activenetwerx.local>

Working on an OpenWRT install and space is limited so moh files are not an option.
I looked through the wiki but didn't see anything and I am sure its trivial but how can
I simply play a beep every 10 seconds in place of moh?

Thanks!
jlc


From steveayre at gmail.com  Mon Apr 18 00:38:45 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 17 Apr 2011 21:38:45 +0100
Subject: [Freeswitch-users] Beep on hold
In-Reply-To: <assp.0088b98154.92097A6A775D5147B1078E3F15430B923A10FD@prato.activenetwerx.local>
References: <assp.0088b98154.92097A6A775D5147B1078E3F15430B923A10FD@prato.activenetwerx.local>
Message-ID: <BANLkTinz0oJx1dKut4qMbhHw28=0as4L3Q@mail.gmail.com>

Hold music can be set using the hold_music channel variable:
http://wiki.freeswitch.org/wiki/Channel_Variables#hold_music

One option is that you could either create a music file that contains a beep
followed by 10s of silence (I believe it loops so just record a single
pass). Wave would be best as any other format will require the additonal
step of decoding the file before sending it to the client, which'd use more
CPU.

The other option is to use Tonestream (
http://wiki.freeswitch.org/wiki/Tone_stream) to generate a stream that
contains a beep and 10s of silence which loops.

-Steve



On 17 April 2011 17:45, Joseph L. Casale <jcasale at activenetwerx.com> wrote:

> Working on an OpenWRT install and space is limited so moh files are not an
> option.
> I looked through the wiki but didn't see anything and I am sure its trivial
> but how can
> I simply play a beep every 10 seconds in place of moh?
>
> Thanks!
> jlc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon Apr 18 03:51:27 2011
From: bwibowo at gmail.com (Budi wibowo)
Date: Sun, 17 Apr 2011 23:51:27 +0000
Subject: [Freeswitch-users] webphone app
In-Reply-To: <BANLkTi=_tM67uu0MJBsWq5NLnVOEbPHm0w@mail.gmail.com>
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com><32EF1A658EFC4E5393D8D1A3A486DA31@e1705><BANLkTikdW-5-hDz9A+0mDid_KNEQNhzf6Q@mail.gmail.com><BANLkTi=73yyZ4Chh8OwfiJhNXAVXnrZ8MA@mail.gmail.com><BANLkTi=_tM67uu0MJBsWq5NLnVOEbPHm0w@mail.gmail.com>
Message-ID: <1481137344-1303084287-cardhu_decombobulator_blackberry.rim.net-323628945-@b26.c2.bise3.blackberry>

The phone is working now thx

-----Original Message-----
From: budi wibowo <bwibowo at gmail.com>
Date: Sun, 17 Apr 2011 05:11:08 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] webphone app

i try that, but got confussion on  installing the application.
http://code.google.com/p/bigbluebutton/wiki/Red5Phone also not work.
please give detail installation method


On Thu, Apr 7, 2011 at 8:17 AM, Moe Navid <manavid at gmail.com> wrote:

> 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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 curriegrad2004 at gmail.com  Mon Apr 18 05:54:26 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Sun, 17 Apr 2011 18:54:26 -0700
Subject: [Freeswitch-users] gentls_cert: Specifying the Country Name?
In-Reply-To: <1302955066042-6278826.post@n2.nabble.com>
References: <1302955066042-6278826.post@n2.nabble.com>
Message-ID: <BANLkTik+UJxiv-mivZK+u-csaa0WWi_sGg@mail.gmail.com>

You can always use the OpenSSL toolkit to manually generate the
required certificates

On Sat, Apr 16, 2011 at 4:57 AM, peely <freeswitch at peely.com> wrote:
> Hi,
>
> I'm trying to generate a certificate request using gentls_cert, only when I
> attempt to use the contents of careq.pem getting a commercial certificate I
> am instructed the country code is missing.
>
> Is there a way of passing the country code into gentls_cert?
>
>
> Thanks,
>
>
> Neil.
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/gentls-cert-Specifying-the-Country-Name-tp6278826p6278826.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From Nabble at slickdeals.endjunk.com  Mon Apr 18 05:59:34 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 17 Apr 2011 18:59:34 -0700 (PDT)
Subject: [Freeswitch-users] DNS SRV Failover - How about incoming?
In-Reply-To: <AANLkTim06KmPqqMNYgXnyCVc7cVwjCc60=D=ZF686neS@mail.gmail.com>
References: <AANLkTim06KmPqqMNYgXnyCVc7cVwjCc60=D=ZF686neS@mail.gmail.com>
Message-ID: <1303091974704-6282254.post@n2.nabble.com>


Avi Marcus-2 wrote:
> What about incoming to the phone, via FS? If devices are behind NAT then
> is there any way to successfully get the calls back to them? Would it make
> sense to have the registration mirrored across the servers that may be
> failed over to?
I hope we get some feedback on this because I would like to know, too.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/DNS-SRV-Failover-How-about-incoming-tp6183293p6282254.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jcasale at activenetwerx.com  Mon Apr 18 07:11:33 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Mon, 18 Apr 2011 03:11:33 +0000
Subject: [Freeswitch-users] Beep on hold
In-Reply-To: <BANLkTinz0oJx1dKut4qMbhHw28=0as4L3Q@mail.gmail.com>
References: <92097A6A775D5147B1078E3F15430B923A10FD@prato.activenetwerx.local>
	<BANLkTinz0oJx1dKut4qMbhHw28=0as4L3Q@mail.gmail.com>
Message-ID: <assp.0089017369.92097A6A775D5147B1078E3F15430B923A2763@prato.activenetwerx.local>

>The other option is to use Tonestream 

That's perfect Steve.
Thanks!
jlc


From acrow at integrafin.co.uk  Mon Apr 18 12:09:00 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 18 Apr 2011 09:09:00 +0100
Subject: [Freeswitch-users] show registrations missing
In-Reply-To: <BC86D31A-5406-408D-9E0F-DFCD17CBD9E2@freeswitch.org>
References: <assp.00880913a1.92097A6A775D5147B1078E3F15430B9239C458@prato.activenetwerx.local>
	<BC86D31A-5406-408D-9E0F-DFCD17CBD9E2@freeswitch.org>
Message-ID: <4DABF19C.3050106@integrafin.co.uk>

Brian,

Have you changed your surname recently?

Alex

On 17/04/11 05:41, Brian Weat wrote:

-- 
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.

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Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856)



From boris at tagnet.ru  Mon Apr 18 17:06:32 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Mon, 18 Apr 2011 19:06:32 +0600
Subject: [Freeswitch-users] regexp with LUA script
Message-ID: <4DAC3758.1000405@tagnet.ru>

Hello!

     How can I use regexp with LUA scripts? Yes, I found I may use 
api:execute('regex'....) but can't understand how to get matched strings 
from it. For example my re is (\d+)#(\d+) and I want to get $1 and $2 in 
some variables.

-- 
Regards,
Boris




From roger.castaldo at gmail.com  Mon Apr 18 17:37:40 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Mon, 18 Apr 2011 09:37:40 -0400
Subject: [Freeswitch-users] Loading xml.so
Message-ID: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>

Hi I have a freshly compiled install of freeswitch that appears to be
working, but the cpu is sitting at 100% with it doing nothing, no logging
messages, nothing.  I ran some tests and tried to reload xml to see if there
was a configuration error and it throws a critical message saying
"/freeswitch/mod/xml.so: cannot open shared object file: No such file or
directory".  I have double checked my configurations and there should not be
any loading of this that I can see, but the file is missing in the mod
directory and I cannot seem to find a reference of it in the source.  I am
not sure if this is where my issue is with the application but I think this
is a good start, to find out why it was not compiled in and what is using
the library.
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From gourav at rentec.com  Mon Apr 18 17:39:15 2011
From: gourav at rentec.com (Gourav Vohra)
Date: Mon, 18 Apr 2011 09:39:15 -0400 (EDT)
Subject: [Freeswitch-users] Shared Call appearence, barging and presence
In-Reply-To: <1346872337.280695.1303133806114.JavaMail.root@zinnia1>
Message-ID: <711646127.280727.1303133955153.JavaMail.root@zinnia1>


Can someone please help me with this.  

I am having the following issue.

In my setup I have three phones connected to freeswitch.

phn1: x2908
phn2: x2995
phn3: x2996, x2995

phn2 and phn3 are configured with X2995 as a shared line with barge enabled.

In my test I place a call from phn1(x2908) to phn2(x2995) and phn3(x2995).
Next I pick up the call on phn3.
Next I barge into the call from phn2.
At this point phn1, phn2 and phn3 are all on the call.
Next I press "End Call" on phn3.

Phone3 was the phone I picked the initial call on and phn2 was the phone
that barged in.  After pressing "End Call" on phn3, the call completely
disappeared on phn3.  Phone1 and phn2 were still on the call.

After pressing "End Call" on phn3 I should have still seen the call
appearance on phn3.  But in my case phn3 showed that x2995 was    
not in use at all.  

I also did a SIP trace on this and all I see from phn3 is a BYE
to the freeswitch server.

Following are the registration settings set on the phone that I am using.

        reg.1.server.1.address="192.168.126.33"
        reg.1.server.1.register="1"
        reg.1.address="2995"
        reg.1.label="2995"
        reg.1.type="shared"
        reg.1.auth.userId="2995"
        reg.1.bargeInEnabled="1"
        reg.1.callsPerLineKey="4"
        reg.1.lineKeys="1"
        reg.1.outboundProxy.address="192.168.126.33"
        reg.1.outboundProxy.port="5060"


Thanks.-


From d.eisner at minesite.com.au  Mon Apr 18 08:26:24 2011
From: d.eisner at minesite.com.au (Dirk Eisner)
Date: Mon, 18 Apr 2011 14:26:24 +1000
Subject: [Freeswitch-users] Multi-homed freeswitch want to use internal
	interface only
Message-ID: <DD5D873A5D84F949A2051D0F326B82F601462533@SYD5.minesite.com.au>

Hi everyone,

 

I have a server which has multiple network cards configured with
different IP addresses in different subnets.

I would like to use only one IP address (internal) for the registration
of phones.

 

My problem is that I'm having difficulty to stop FreeSwitch from finding
the external interface.

 

It there somewhere a easy description of the process?

 

Alternatively I'm happy to pay for some support, If someone would be so
friendly to pass on a contact. It would be great if this person is in
Australia, but I'm happy with anyone.

 

Thank you for your help 

 

 

 

 

 

   Dirk Eisner | Systems Architect
    Mine Site Technologies Pty Ltd.

     Tel: 02-9018-0940    | Fax: 02-9437-5688    |  Mobile: 0458-080-370
     25-27 Whiting St  | Artarmon, NSW  2064
     http://www.minesite.com.au   | E-mail: d.eisner at minesite.com.au

 

 

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From vedran.zeljeznak at gmail.com  Mon Apr 18 11:25:48 2011
From: vedran.zeljeznak at gmail.com (Vedran Zeljeznak)
Date: Mon, 18 Apr 2011 09:25:48 +0200
Subject: [Freeswitch-users] mod_h323
Message-ID: <BANLkTik_sVoF_hdpxPEi5Rc3JNPpKdTxHQ@mail.gmail.com>

hello,

has anybody compiled and ran mod_h323 with success lately?

I'm having trouble bridging two H.323 calls via mod_h323.
First two calls bridge relatively successfully, eg. some problems occur when
called side hangs up the call first, the calling side of the call doesn't
hangup automatically.
The third call however doesn't create any debuging messages in FS console.
Looks like something remained locked in the first call.

My FS version is: "FreeSWITCH Version 1.0.head (git-413d590 2011-04-14
19-56-45 -0500)"
OpenH.323 version: h323plus-20100525
PTLib version: 2.8.2 -
http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/tags/v2_8_2/

i've put freeswitch.log file to pastebin (
http://pastebin.freeswitch.org/16108).

Do you have any suggestions on where did i go wrong or where to start
looking for a bug?

---
Vedran Zeljeznak
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From ktaylor91 at yahoo.com  Mon Apr 18 12:49:40 2011
From: ktaylor91 at yahoo.com (Kenneth Taylor)
Date: Mon, 18 Apr 2011 01:49:40 -0700 (PDT)
Subject: [Freeswitch-users] frame per packet
Message-ID: <630958.78492.qm@web121503.mail.ne1.yahoo.com>

Hi,
I'm building a voip client for android phones over gsm and consider using FS as 
sever.
In order to save cellular costs I want to put more than one frame per Rtp 
packet.

Is there any easy solution to do it in FS? 

Or, where is the right place to add it? In mod_sofia or sofia_glue? should I go 
to lower levels?
I don't think I should do it in the codec, it may cause problems in the rtp 
sequence number and probably a lot more.

TNX,
Ken
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From habib at alexcoder.com  Mon Apr 18 16:35:41 2011
From: habib at alexcoder.com (Mohammed Habib)
Date: Mon, 18 Apr 2011 14:35:41 +0200
Subject: [Freeswitch-users] Originated session callback.
In-Reply-To: <BANLkTik81bAyQiwxJym=KH3VdAD-oTPiqA@mail.gmail.com>
References: <BANLkTinje48f3MSjB6bRWqpfgTi-xsxC8A@mail.gmail.com>
	<BANLkTik81bAyQiwxJym=KH3VdAD-oTPiqA@mail.gmail.com>
Message-ID: <BANLkTi=JeAnNfvwrb-Y2OnGQjJRNjD-vHw@mail.gmail.com>

I need help getting events from originated session.

This is my lua script:

function onInput_MainSession(s, type, obj)
   -- This one is working fine.
   freeswitch.consoleLog("info", "Callback with type " .. type .. "\n");
end

function onInput_NewSession(s, type, obj)
   -- This one is never called.
   freeswitch.consoleLog("info", "Callback with type " .. type .. "\n");
end

session:answer();
session:setInputCallback("onInput_MainSession");
session:sleep(200);
session:execute("detect_speech", "unimrcp testgrammer trestgrammer");

newsession = freeswitch.Session("user/1002");
newsession:setInputCallback("onInput_NewSession");
newsession:sleep(200);
newsession:execute("detect_speech", "unimrcp testgrammer trestgrammer");

while ((session:ready() == true) ) and (newsession:ready() == true) do
    -- Loop
    sleep(200);
end

I am unable to capture any of the new session events or dtmf.

Please help.

Thank you,
Mohammed Habib
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From kris at kriskinc.com  Mon Apr 18 18:05:44 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 18 Apr 2011 10:05:44 -0400
Subject: [Freeswitch-users] frame per packet
In-Reply-To: <630958.78492.qm@web121503.mail.ne1.yahoo.com>
References: <630958.78492.qm@web121503.mail.ne1.yahoo.com>
Message-ID: <BANLkTim2HBv=NOjSSm+gWtw0_+ujfKzR=Q@mail.gmail.com>

Why not just use a (relatively) high ptime?

On Mon, Apr 18, 2011 at 4:49 AM, Kenneth Taylor <ktaylor91 at yahoo.com> wrote:
> Hi,
> I'm building a voip client for android phones over gsm and consider using FS
> as sever.
> In order to save cellular costs I want to put more than one frame per Rtp
> packet.
> Is there any easy solution to do it in FS?
> Or, where is the right place to add it? In mod_sofia or sofia_glue? should I
> go to lower levels?
> I don't think I should do it in the codec, it may cause problems in the rtp
> sequence number and probably a lot more.
> TNX,
> Ken

-- 
Kristian Kielhofner


From roger.castaldo at gmail.com  Mon Apr 18 18:07:19 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Mon, 18 Apr 2011 10:07:19 -0400
Subject: [Freeswitch-users] Multi-homed freeswitch want to use internal
 interface only
In-Reply-To: <DD5D873A5D84F949A2051D0F326B82F601462533@SYD5.minesite.com.au>
References: <DD5D873A5D84F949A2051D0F326B82F601462533@SYD5.minesite.com.au>
Message-ID: <BANLkTikPoeZs0iq4KTFxCRzx5-oNW773tQ@mail.gmail.com>

I believe the configuration settings you are looking for would be in the
/conf/vars.xml and /conf/sip_profiles/internal.xml.  Inside here you can
specify the ips to bind the internal sip profile too as well as configure
which ips to bind freeswitch to.  I cannot recall what the settings are
called but the configuration files are quite well documented within them to
point you in the right direction.

On Mon, Apr 18, 2011 at 12:26 AM, Dirk Eisner <d.eisner at minesite.com.au>wrote:

>  Hi everyone,
>
>
>
> I have a server which has multiple network cards configured with different
> IP addresses in different subnets.
>
> I would like to use only one IP address (internal) for the registration of
> phones.
>
>
>
> My problem is that I?m having difficulty to stop FreeSwitch from finding
> the external interface.
>
>
>
> It there somewhere a easy description of the process?
>
>
>
> Alternatively I?m happy to pay for some support, If someone would be so
> friendly to pass on a contact. It would be great if this person is in
> Australia, but I?m happy with anyone.
>
>
>
> Thank you for your help
>
>
>
>
>
> **
>
> * *
>
>
>
> *   **Dirk Eisner** **|* *Systems Architect*
>     *Mine Site Technologies Pty Ltd.*
>
>      Tel: 02-9018-0940    *|* Fax: 02-9437-5688    *|*  Mobile:
> 0458-080-370
>      25-27 Whiting St * **|* Artarmon, NSW*  *2064
>      *http://www.minesite.com.au   **|* E-mail: d.eisner at minesite.com.au
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kris at kriskinc.com  Mon Apr 18 18:07:41 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Mon, 18 Apr 2011 10:07:41 -0400
Subject: [Freeswitch-users] Loading xml.so
In-Reply-To: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>
References: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>
Message-ID: <BANLkTimCDLM2L4byTCe_Xi9hZCj4-EQ=aw@mail.gmail.com>

Which module do you want to load?  What functionality are you looking for?

There is no module "xml.so" included with the FreeSWITCH source.

On Mon, Apr 18, 2011 at 9:37 AM, Roger Castaldo
<roger.castaldo at gmail.com> wrote:
> Hi I have a freshly compiled install of freeswitch that appears to be
> working, but the cpu is sitting at 100% with it doing nothing, no logging
> messages, nothing.? I ran some tests and tried to reload xml to see if there
> was a configuration error and it throws a critical message saying
> "/freeswitch/mod/xml.so: cannot open shared object file: No such file or
> directory".? I have double checked my configurations and there should not be
> any loading of this that I can see, but the file is missing in the mod
> directory and I cannot seem to find a reference of it in the source.? I am
> not sure if this is where my issue is with the application but I think this
> is a good start, to find out why it was not compiled in and what is using
> the library.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Nabble at slickdeals.endjunk.com  Mon Apr 18 18:44:25 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 18 Apr 2011 07:44:25 -0700 (PDT)
Subject: [Freeswitch-users] Loading xml.so
In-Reply-To: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>
References: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>
Message-ID: <1303137865325-6283898.post@n2.nabble.com>


Roger Castaldo wrote:
> Hi I have a freshly compiled install of freeswitch that appears to be
> working, but the cpu is sitting at 100% with it doing nothing, no logging
> messages, nothing.
Under what platform is your FS hosted on and what programs have hogged the
CPU to 100%?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Loading-xml-so-tp6283668p6283898.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From infos at madovsky.org  Mon Apr 18 18:56:39 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 18 Apr 2011 10:56:39 -0400
Subject: [Freeswitch-users] Loading xml.so
References: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>
Message-ID: <AA3959A6218748989D81EAB4CE7DE266@e1705>

try to compile your kernel with a 1000hz freqency option
  ----- Original Message ----- 
  From: Roger Castaldo 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Monday, April 18, 2011 9:37 AM
  Subject: [Freeswitch-users] Loading xml.so


  Hi I have a freshly compiled install of freeswitch that appears to be working, but the cpu is sitting at 100% with it doing nothing, no logging messages, nothing.  I ran some tests and tried to reload xml to see if there was a configuration error and it throws a critical message saying "/freeswitch/mod/xml.so: cannot open shared object file: No such file or directory".  I have double checked my configurations and there should not be any loading of this that I can see, but the file is missing in the mod directory and I cannot seem to find a reference of it in the source.  I am not sure if this is where my issue is with the application but I think this is a good start, to find out why it was not compiled in and what is using the library.



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


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  FreeSWITCH-users at lists.freeswitch.org
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  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From stkn at freeswitch.org  Mon Apr 18 19:10:12 2011
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Mon, 18 Apr 2011 17:10:12 +0200
Subject: [Freeswitch-users] Cannot compile freetdm!
In-Reply-To: <BANLkTinntgEpU9L-QSciqC+zt4Y6JrxbqQ@mail.gmail.com>
References: <BANLkTinntgEpU9L-QSciqC+zt4Y6JrxbqQ@mail.gmail.com>
Message-ID: <201104181710.20357.stkn@freeswitch.org>

Am Friday 15 April 2011 schrieb Valery Kalinin:
> I download latest version (2011-03-10) libisdn from:
> http://oss.axsentis.de/gitweb/?p=libisdn.git;a=summary
> 
> Stefan Knoblich cannot help me...:-(
> 
> Does FreeSWITCH work only with Sangoma cards?!

I fixed those warnings in ftmod_isdn, however, that module is currently
unsupported and in an unknown state (= most likely broken after the
OpenZAP -> FreeTDM switch and later changes).

The module is currently being rewritten (including parts of libisdn) and the
new code will be released when it reaches a usable state (no ETA).

- 
-------------------------------------------------------------------------------
Stefan Knoblich                        | Web:   http://www.axsentis.de/
axsentis GmbH                          |        http://oss.axsentis.de/
Eupener Str. 74, 50933 Koeln, Germany  |
Amtsgericht Koeln: HR B 56238          | Email: s.knoblich at axsentis.de
UST-ID: DE244977565                    | JID:   s.knoblich at jabber.axsentis.de
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Email: stkn at freeswitch.org
IRC:   #freeswitch-de @ irc.freenode.net
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From anthony.minessale at gmail.com  Mon Apr 18 19:30:59 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 18 Apr 2011 10:30:59 -0500
Subject: [Freeswitch-users] Loading xml.so
In-Reply-To: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>
References: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>
Message-ID: <BANLkTimGHfEXzoV-3w55=yj9TRxnM54dTQ@mail.gmail.com>

There is no such thing as what you describe, error messages were not
meant to be paraphrased.
The answer is go to the wiki and follow the instructions to build a
fresh checkout of GIT and try again and if you get the same thing
provide traces of the exact messages.




On Mon, Apr 18, 2011 at 8:37 AM, Roger Castaldo
<roger.castaldo at gmail.com> wrote:
> Hi I have a freshly compiled install of freeswitch that appears to be
> working, but the cpu is sitting at 100% with it doing nothing, no logging
> messages, nothing.? I ran some tests and tried to reload xml to see if there
> was a configuration error and it throws a critical message saying
> "/freeswitch/mod/xml.so: cannot open shared object file: No such file or
> directory".? I have double checked my configurations and there should not be
> any loading of this that I can see, but the file is missing in the mod
> directory and I cannot seem to find a reference of it in the source.? I am
> not sure if this is where my issue is with the application but I think this
> is a good start, to find out why it was not compiled in and what is using
> the library.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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  Mon Apr 18 19:43:03 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Apr 2011 08:43:03 -0700
Subject: [Freeswitch-users] regexp with LUA script
In-Reply-To: <4DAC3758.1000405@tagnet.ru>
References: <4DAC3758.1000405@tagnet.ru>
Message-ID: <BANLkTikebBSkuaTMNZQ6vpFSwVf=FjusTQ@mail.gmail.com>

I don't believe there is a way to pull multiple values back from the call to
api:execute. However, Lua has native pattern matching with captures. Here
are a few resources for you:

FreeSWITCH book, pages 149-151
http://www.lua.org/pil/20.3.html
http://www.lua.org/manual/5.1/manual.html#5.4.1

Here's a simple example that you can actually run with "luarun" if you want
to do some testing:

-- pattern.lua
data = "1234#5678";
_,_,var1,var2 = string.find(data,"(%d+)#(%d+)");
freeswitch.consoleLog("INFO","\ndata: " .. data .. "\nvar1: " .. var1 ..
"\nvar2: " .. var2 .. "\n");

Output:
freeswitch at internal> luarun pattern.lua
+OK

2011-04-18 08:28:49.242080 [INFO] switch_cpp.cpp:1197
data: 1234#5678
var1: 1234
var2: 5678

Have fun!
-MC

P.S. - I added some of this information to the wiki:
http://wiki.freeswitch.org/wiki/Mod_lua#Native_Lua_Pattern_Matching

On Mon, Apr 18, 2011 at 6:06 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     How can I use regexp with LUA scripts? Yes, I found I may use
> api:execute('regex'....) but can't understand how to get matched strings
> from it. For example my re is (\d+)#(\d+) and I want to get $1 and $2 in
> some variables.
>
> --
> 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 anthony.minessale at gmail.com  Mon Apr 18 20:30:04 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 18 Apr 2011 11:30:04 -0500
Subject: [Freeswitch-users] conference play / stop / PAUSE/RESUME?
In-Reply-To: <BANLkTi=9672+WLZON0teCJLa-CFsg4E88w@mail.gmail.com>
References: <BANLkTi=9672+WLZON0teCJLa-CFsg4E88w@mail.gmail.com>
Message-ID: <BANLkTimgE00BQ9AC9O8jOtBtKz3Mv4je=Q@mail.gmail.com>

if you need this level of granularity, I suggest instead of the
conference play stuff that is built in, that you use the originate
command to call a new channel into the conference to play the file
then use the uuid_fileman api command to manipulate the playback.


On Sat, Apr 16, 2011 at 1:33 PM, Dale Trub <daletrub at gmail.com> wrote:
> We let the conference host play a file to the conference and want to give
> them a control to "pause" such a file, and then resume it.
> Has anyone found a way to do this?
> Would someone create something like that for a modest bounty? ?We could
> incorp in the code we use, as well as submit to the project overall.
> -Dale
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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 18 20:35:24 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Mon, 18 Apr 2011 22:35:24 +0600
Subject: [Freeswitch-users] regexp with LUA script
In-Reply-To: <BANLkTikebBSkuaTMNZQ6vpFSwVf=FjusTQ@mail.gmail.com>
References: <4DAC3758.1000405@tagnet.ru>
	<BANLkTikebBSkuaTMNZQ6vpFSwVf=FjusTQ@mail.gmail.com>
Message-ID: <4DAC684C.1080907@tagnet.ru>

Hello!

     Yes, I know about LUA native patterns. Unfortunatelly I can't (or 
don't know how to) do something like:
\d{4,6} or \d{10,}

> I don't believe there is a way to pull multiple values back from the 
> call to api:execute. However, Lua has native pattern matching with 
> captures. Here are a few resources for you:
>
> FreeSWITCH book, pages 149-151
> http://www.lua.org/pil/20.3.html
> http://www.lua.org/manual/5.1/manual.html#5.4.1
>
> Here's a simple example that you can actually run with "luarun" if you 
> want to do some testing:
>
> -- pattern.lua
> data = "1234#5678";
> _,_,var1,var2 = string.find(data,"(%d+)#(%d+)");
> freeswitch.consoleLog("INFO","\ndata: " .. data .. "\nvar1: " .. var1 
> .. "\nvar2: " .. var2 .. "\n");
>
> Output:
> freeswitch at internal> luarun pattern.lua
> +OK
>
> 2011-04-18 08:28:49.242080 [INFO] switch_cpp.cpp:1197
> data: 1234#5678
> var1: 1234
> var2: 5678
>
> Have fun!
> -MC
>
> P.S. - I added some of this information to the wiki:
> http://wiki.freeswitch.org/wiki/Mod_lua#Native_Lua_Pattern_Matching
>
> On Mon, Apr 18, 2011 at 6:06 AM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         How can I use regexp with LUA scripts? Yes, I found I may use
>     api:execute('regex'....) but can't understand how to get matched
>     strings
>     from it. For example my re is (\d+)#(\d+) and I want to get $1 and
>     $2 in
>     some variables.
>
>     --
>     Regards,
>     Boris
>
>
>
>     _______________________________________________
>     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


-- 
Regards,
Boris


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From roger.castaldo at gmail.com  Mon Apr 18 21:11:50 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Mon, 18 Apr 2011 13:11:50 -0400
Subject: [Freeswitch-users] Loading xml.so
In-Reply-To: <BANLkTimGHfEXzoV-3w55=yj9TRxnM54dTQ@mail.gmail.com>
References: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>
	<BANLkTimGHfEXzoV-3w55=yj9TRxnM54dTQ@mail.gmail.com>
Message-ID: <BANLkTinVxO38Y5cZOtZJQWWJ14wLft1Xuw@mail.gmail.com>

The cpu thing apparently was being caused by the system clock thinking its
1943, guess the cmos battery died, once the time was corrected things
stoppped running at 100% cpu.  As far as the xml.so I have no idea which
module is referencing it, just quoting the message that the console gave, I
suspect one of my library references is missing.  Either way its not
affecting the system so I am no longer concerned.  Thank you for the help.

On Mon, Apr 18, 2011 at 11:30 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> There is no such thing as what you describe, error messages were not
> meant to be paraphrased.
> The answer is go to the wiki and follow the instructions to build a
> fresh checkout of GIT and try again and if you get the same thing
> provide traces of the exact messages.
>
>
>
>
> On Mon, Apr 18, 2011 at 8:37 AM, Roger Castaldo
> <roger.castaldo at gmail.com> wrote:
> > Hi I have a freshly compiled install of freeswitch that appears to be
> > working, but the cpu is sitting at 100% with it doing nothing, no logging
> > messages, nothing.  I ran some tests and tried to reload xml to see if
> there
> > was a configuration error and it throws a critical message saying
> > "/freeswitch/mod/xml.so: cannot open shared object file: No such file or
> > directory".  I have double checked my configurations and there should not
> be
> > any loading of this that I can see, but the file is missing in the mod
> > directory and I cannot seem to find a reference of it in the source.  I
> am
> > not sure if this is where my issue is with the application but I think
> this
> > is a good start, to find out why it was not compiled in and what is using
> > the library.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anthony Minessale II
>
> 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 acrow at integrafin.co.uk  Mon Apr 18 22:11:12 2011
From: acrow at integrafin.co.uk (Alex Crow)
Date: Mon, 18 Apr 2011 19:11:12 +0100
Subject: [Freeswitch-users] Mitel 3300 ICP "Multicall" Equivalent
In-Reply-To: <4D67EB2A.80409@integrafin.co.uk>
References: <4D66B5C5.7000005@integrafin.co.uk>	<9CCDC306-B282-4CC6-A648-B12A97398D9F@freeswitch.org>
	<4D67EB2A.80409@integrafin.co.uk>
Message-ID: <4DAC7EC0.4000600@integrafin.co.uk>

On 25/02/11 17:47, Alex Crow wrote:
> On 25/02/11 16:18, Brian West wrote:
>> You want Directed Call Pickup right?
>>
>> /b
>>
>> On Feb 24, 2011, at 1:47 PM, Alex Crow wrote:
>>
>>
> Hi Brian,
>
> Thanks for your reply - makes perfect sense - if it's Asterisk style
> **<multicall_number>  on a key with a SUBSCRIBE to the same number on the
> phone. I am concerned that one of the phones actually would have to be
> registered to that extension to make it work, or is that not correct?
> This is why I was looking at SLA.
>
> The other part of this is that the<multicall_number>  must not go to
> voicemail or busy tone until capacity is reached, ie. all the people
> with the BLF button configured are busy. I tried this with
> TrixBox/Asterisk but as soon as one call was live all further calls to
> the same number gave voicemail or busy.
>
> If both of the above work "just great" in FreeSwitch then you may well
> have found my solution.
>
> I haven't attempted what you describe with FS as I don't have enough
> phones set up at the moment to do it, and I currently only have 1
> Grandstream GXP2000 (needs FreeSwitch code hacking for BLF to work even
> half-properly) and a shedload of Mitel (SIP Mode 5224/5324) where the
> BLF doesn't work at all. Softphones are not under consideration at
> present. I'm quite able to get a couple of Polycoms or Snoms but I'd
> need to justify the expense to management.
>
> Cheers
>
> Alex
>
>
>
Brian, all,

For any others that might be attempting such a migration, I think I have 
the setup OK.

It seems to be sufficient to enable multiple registrations on FS, then 
at least on SNOM leave everything at default (no SLA!) other than 
disabling call waiting, and assign a "line" button for your "multicall" 
number. This appears to work as I described, incoming calls to the 
number ring on all registered phones, flash the button and pressing it 
picks up. If all the phones are busy the call goes to VM. Voila!

Cheers

Alex



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From msc at freeswitch.org  Mon Apr 18 23:09:05 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Apr 2011 12:09:05 -0700
Subject: [Freeswitch-users] regexp with LUA script
In-Reply-To: <4DAC684C.1080907@tagnet.ru>
References: <4DAC3758.1000405@tagnet.ru>
	<BANLkTikebBSkuaTMNZQ6vpFSwVf=FjusTQ@mail.gmail.com>
	<4DAC684C.1080907@tagnet.ru>
Message-ID: <BANLkTik+O40aS1MtJJjCm-F1Ccy-W+OjwQ@mail.gmail.com>

On Mon, Apr 18, 2011 at 9:35 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

>  Hello!
>
>     Yes, I know about LUA native patterns. Unfortunatelly I can't (or don't
> know how to) do something like:
> \d{4,6} or \d{10,}
>

\d{4,6} is "%d%d%d%d%d?%d?"

\d{10,} is "%d%d%d%d%d%d%d%d%d%d+"

At first glance that may seem inelegant, however, it is far more efficient
than trying to send a regex through an api:execute call.

-MC
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From steveayre at gmail.com  Mon Apr 18 23:58:42 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 18 Apr 2011 20:58:42 +0100
Subject: [Freeswitch-users] Loading xml.so
In-Reply-To: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>
References: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>
Message-ID: <68AE1930-8D2F-41F3-81E1-FEDF9A26922C@gmail.com>

Do a gcore to generate a coredump of the process while it is still running. If you load that into gdb and get a backtrace it might show the culprit.

What version are you on? Have you tried the latest got head? If not, it's possible it's a bug that is already fixed.

> freeswitch/mod/xml.so

That sounds wrong. Check your modules.conf.xml file in your config. Is it trying to load a module named xml? (there is no such module, but there is mod_xml_curl, mod_xml_cdr etc

Steve on iPhone

On 18 Apr 2011, at 14:37, Roger Castaldo <roger.castaldo at gmail.com> wrote:

> Hi I have a freshly compiled install of freeswitch that appears to be working, but the cpu is sitting at 100% with it doing nothing, no logging messages, nothing.  I ran some tests and tried to reload xml to see if there was a configuration error and it throws a critical message saying "/freeswitch/mod/xml.so: cannot open shared object file: No such file or directory".  I have double checked my configurations and there should not be any loading of this that I can see, but the file is missing in the mod directory and I cannot seem to find a reference of it in the source.  I am not sure if this is where my issue is with the application but I think this is a good start, to find out why it was not compiled in and what is using the library.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Tue Apr 19 00:04:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Apr 2011 13:04:49 -0700
Subject: [Freeswitch-users] conference play / stop / PAUSE/RESUME?
In-Reply-To: <BANLkTimgE00BQ9AC9O8jOtBtKz3Mv4je=Q@mail.gmail.com>
References: <BANLkTi=9672+WLZON0teCJLa-CFsg4E88w@mail.gmail.com>
	<BANLkTimgE00BQ9AC9O8jOtBtKz3Mv4je=Q@mail.gmail.com>
Message-ID: <BANLkTin=Ps=ri8mVyH0HTgFe2pB4oD_Sxg@mail.gmail.com>

On Mon, Apr 18, 2011 at 9:30 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> if you need this level of granularity, I suggest instead of the
> conference play stuff that is built in, that you use the originate
> command to call a new channel into the conference to play the file
> then use the uuid_fileman api command to manipulate the playback.
>

FYI,
I just added the basic uuid_fileman stuff to the wiki:
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman

I don't have actual examples yet - I just combed through the source code to
pull out the various commands that are available. For those who like to dig
into the source code: uuid_fileman in mod_commands.c is essentially a
wrapper/helper that calls switch_ivr_process_fh() from switch_ivr.c in the
core. The commands are all in that function. I was able to deduce most of
the args and their meanings by looking at the source code. If you have used
these functions and can contribute some simple examples that would be very
helpful.

Many thanks,
MC
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From steveayre at gmail.com  Tue Apr 19 00:04:40 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 18 Apr 2011 21:04:40 +0100
Subject: [Freeswitch-users] frame per packet
In-Reply-To: <630958.78492.qm@web121503.mail.ne1.yahoo.com>
References: <630958.78492.qm@web121503.mail.ne1.yahoo.com>
Message-ID: <546FE599-8C29-44AD-8D78-9711BF397544@gmail.com>

AFAIK, a rtp packet *is* a frame, so no you can't.

As the other poster said, use a higher ptime. That will make the frame store a longer period of time which will do the same as that you want.

Overhead will be lower, but quality will be worse if you drop packets or they fail to arrive in a timely manner. That's very likely on gsm, and the packet size will mean it takes longer to be delivered which'll mean more delay and probably more packets arriving too late. You'll need to experiment to find a good balance. And remember what works well in a city might fail to work in the country where coverage is poorer.

Steve on iPhone

On 18 Apr 2011, at 09:49, Kenneth Taylor <ktaylor91 at yahoo.com> wrote:

> Hi,
> I'm building a voip client for android phones over gsm and consider using FS as sever.
> In order to save cellular costs I want to put more than one frame per Rtp packet.
> 
> Is there any easy solution to do it in FS? 
> 
> Or, where is the right place to add it? In mod_sofia or sofia_glue? should I go to lower levels?
> I don't think I should do it in the codec, it may cause problems in the rtp sequence number and probably a lot more.
> 
> TNX,
> Ken
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 d.eisner at minesite.com.au  Tue Apr 19 00:21:40 2011
From: d.eisner at minesite.com.au (Dirk Eisner)
Date: Mon, 18 Apr 2011 20:21:40 -0000
Subject: [Freeswitch-users] show registrations missing
Message-ID: <02e001cbfe05$cf21fc96$0800a8c0@minesite.com.au>


No but why do ask?

Sent from my HTC

----- Reply message -----
From: "Alex Crow" <acrow at integrafin.co.uk>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] show registrations missing
Date: Mon, Apr 18, 2011 18:19




Brian,

Have you changed your surname recently?

Alex

On 17/04/11 05:41, Brian Weat wrote:

-- 
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


From roger.castaldo at gmail.com  Tue Apr 19 00:34:39 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Mon, 18 Apr 2011 16:34:39 -0400
Subject: [Freeswitch-users] Loading xml.so
In-Reply-To: <68AE1930-8D2F-41F3-81E1-FEDF9A26922C@gmail.com>
References: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>
	<68AE1930-8D2F-41F3-81E1-FEDF9A26922C@gmail.com>
Message-ID: <BANLkTinQSJwUrHYt4vLcMyXBrXgFZs8vMA@mail.gmail.com>

I am using 1.0.7, I tend to avoid straight from trunk builds.  As far as the
xml modules being loaded, the only one being loaded is dialplan_xml.  All
the other modules are disabled and were not compiled into the system as I
have no intention of using them.  I will try and do a gcore dump to see what
is trying to load the xml.so when I have a chance.

On Mon, Apr 18, 2011 at 3:58 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Do a gcore to generate a coredump of the process while it is still running.
> If you load that into gdb and get a backtrace it might show the culprit.
>
> What version are you on? Have you tried the latest got head? If not, it's
> possible it's a bug that is already fixed.
>
> > freeswitch/mod/xml.so
>
> That sounds wrong. Check your modules.conf.xml file in your config. Is it
> trying to load a module named xml? (there is no such module, but there is
> mod_xml_curl, mod_xml_cdr etc
>
> Steve on iPhone
>
> On 18 Apr 2011, at 14:37, Roger Castaldo <roger.castaldo at gmail.com> wrote:
>
> > Hi I have a freshly compiled install of freeswitch that appears to be
> working, but the cpu is sitting at 100% with it doing nothing, no logging
> messages, nothing.  I ran some tests and tried to reload xml to see if there
> was a configuration error and it throws a critical message saying
> "/freeswitch/mod/xml.so: cannot open shared object file: No such file or
> directory".  I have double checked my configurations and there should not be
> any loading of this that I can see, but the file is missing in the mod
> directory and I cannot seem to find a reference of it in the source.  I am
> not sure if this is where my issue is with the application but I think this
> is a good start, to find out why it was not compiled in and what is using
> the library.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 19 00:49:42 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Apr 2011 13:49:42 -0700
Subject: [Freeswitch-users] show registrations missing
In-Reply-To: <02e001cbfe05$cf21fc96$0800a8c0@minesite.com.au>
References: <02e001cbfe05$cf21fc96$0800a8c0@minesite.com.au>
Message-ID: <BANLkTikF9mw2bp0tNQ=mP76Ga2FNmGN13g@mail.gmail.com>

On Mon, Apr 18, 2011 at 1:22 PM, Dirk Eisner <d.eisner at minesite.com.au>wrote:

>
> No but why do ask?
>

Because Brian's iPhone email says "Brian Weat" instead of "Brian West."
Maybe he should change it to Brian Barley or Brian Hops. :)

-MC
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From msc at freeswitch.org  Tue Apr 19 01:03:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Apr 2011 14:03:29 -0700
Subject: [Freeswitch-users] Originated session callback.
In-Reply-To: <BANLkTi=JeAnNfvwrb-Y2OnGQjJRNjD-vHw@mail.gmail.com>
References: <BANLkTinje48f3MSjB6bRWqpfgTi-xsxC8A@mail.gmail.com>
	<BANLkTik81bAyQiwxJym=KH3VdAD-oTPiqA@mail.gmail.com>
	<BANLkTi=JeAnNfvwrb-Y2OnGQjJRNjD-vHw@mail.gmail.com>
Message-ID: <BANLkTik+HScKABdarAXUUJZUkHAcV1b0SQ@mail.gmail.com>

What are you trying to accomplish with this script? Why are you creating a
new session right in the middle of handling an existing session? I am
curious to know what problem you are attempting to solve.

-MC

On Mon, Apr 18, 2011 at 5:35 AM, Mohammed Habib <habib at alexcoder.com> wrote:

> I need help getting events from originated session.
>
> This is my lua script:
>
> function onInput_MainSession(s, type, obj)
>    -- This one is working fine.
>    freeswitch.consoleLog("info", "Callback with type " .. type .. "\n");
> end
>
> function onInput_NewSession(s, type, obj)
>    -- This one is never called.
>    freeswitch.consoleLog("info", "Callback with type " .. type .. "\n");
> end
>
> session:answer();
> session:setInputCallback("onInput_MainSession");
> session:sleep(200);
> session:execute("detect_speech", "unimrcp testgrammer trestgrammer");
>
> newsession = freeswitch.Session("user/1002");
> newsession:setInputCallback("onInput_NewSession");
> newsession:sleep(200);
> newsession:execute("detect_speech", "unimrcp testgrammer trestgrammer");
>
> while ((session:ready() == true) ) and (newsession:ready() == true) do
>     -- Loop
>     sleep(200);
> end
>
> I am unable to capture any of the new session events or dtmf.
>
> Please help.
>
> Thank you,
> Mohammed Habib
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 lautram.mathieu at gmail.com  Tue Apr 19 01:22:14 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Mon, 18 Apr 2011 23:22:14 +0200
Subject: [Freeswitch-users] stuck channels
In-Reply-To: <AANLkTik5ahRedWCETwPhYB8kSqtGckX-vTGA2aXL2ZG1@mail.gmail.com>
References: <AANLkTimykLH-7i1PivZK8SQHrAFRzuXDMr0QjPPX2Ky3@mail.gmail.com>
	<1301583451169-6227463.post@n2.nabble.com>
	<AANLkTik5ahRedWCETwPhYB8kSqtGckX-vTGA2aXL2ZG1@mail.gmail.com>
Message-ID: <BANLkTinNQsBgvo44Cd609DqMF4o7knEGDg@mail.gmail.com>

Hi,

I posted my problem on Jira 3 weeks ago. At first, I had many answers. But
now, there's nobody. Is there a solution for my problem? Is this an
unfixable bug?
I can't fax anything because I'm waiting for an answer or a fix. Can you
please have a look on the post and the log that I posted please?

Thank you very much :-)

2011/3/31 Mathieu Lautram <lautram.mathieu at gmail.com>

> Thank you for your answer.
> Yes, I use the last Freeswitch (FreeSWITCH Version 1.0.head (git-6e78f6f
> 2011-03-30 11-41-45 +0200)) and the problem is still there and it's always
> related to fax calls.
> I'm reporting this bug to Jira, hope it will be fix soon.
>
>
> 2011/3/31 Jeff Lenk <jeff at jefflenk.com>
>
>> So you used make current and the problem still occurs?
>>
>> Is it always related to fax calls?
>>
>> If the above is correct you should file this a Jira with all relevant
>> information.
>>
>> see http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/stuck-channels-tp6226874p6227463.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
>>
>
>
>
> --
> Mathieu LAUTRAM
> Application developer
>
> BJT Partners - FRANCE
> +33 1 79 75 99 60
> +33 6 61 59 07 25
>



-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From msc at freeswitch.org  Tue Apr 19 01:30:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Apr 2011 14:30:33 -0700
Subject: [Freeswitch-users] stuck channels
In-Reply-To: <BANLkTinNQsBgvo44Cd609DqMF4o7knEGDg@mail.gmail.com>
References: <AANLkTimykLH-7i1PivZK8SQHrAFRzuXDMr0QjPPX2Ky3@mail.gmail.com>
	<1301583451169-6227463.post@n2.nabble.com>
	<AANLkTik5ahRedWCETwPhYB8kSqtGckX-vTGA2aXL2ZG1@mail.gmail.com>
	<BANLkTinNQsBgvo44Cd609DqMF4o7knEGDg@mail.gmail.com>
Message-ID: <BANLkTikwA1EmmrOwn_zxSzFE3NCHr_EW2g@mail.gmail.com>

For those of us who don't live and breath Jira, could you paste the link or
at least the bug number?
-MC

On Mon, Apr 18, 2011 at 2:22 PM, Mathieu Lautram
<lautram.mathieu at gmail.com>wrote:

> Hi,
>
> I posted my problem on Jira 3 weeks ago. At first, I had many answers. But
> now, there's nobody. Is there a solution for my problem? Is this an
> unfixable bug?
> I can't fax anything because I'm waiting for an answer or a fix. Can you
> please have a look on the post and the log that I posted please?
>
> Thank you very much :-)
>
> 2011/3/31 Mathieu Lautram <lautram.mathieu at gmail.com>
>
>> Thank you for your answer.
>> Yes, I use the last Freeswitch (FreeSWITCH Version 1.0.head (git-6e78f6f
>> 2011-03-30 11-41-45 +0200)) and the problem is still there and it's always
>> related to fax calls.
>> I'm reporting this bug to Jira, hope it will be fix soon.
>>
>>
>> 2011/3/31 Jeff Lenk <jeff at jefflenk.com>
>>
>>> So you used make current and the problem still occurs?
>>>
>>> Is it always related to fax calls?
>>>
>>> If the above is correct you should file this a Jira with all relevant
>>> information.
>>>
>>> see http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/stuck-channels-tp6226874p6227463.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
>>>
>>
>>
>>
>> --
>> Mathieu LAUTRAM
>> Application developer
>>
>> BJT Partners - FRANCE
>> +33 1 79 75 99 60
>> +33 6 61 59 07 25
>>
>
>
>
> --
> Mathieu LAUTRAM
> Application developer
>
> BJT Partners - FRANCE
> +33 1 79 75 99 60
> +33 6 61 59 07 25
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 lautram.mathieu at gmail.com  Tue Apr 19 01:33:30 2011
From: lautram.mathieu at gmail.com (Mathieu Lautram)
Date: Mon, 18 Apr 2011 23:33:30 +0200
Subject: [Freeswitch-users] stuck channels
In-Reply-To: <BANLkTikwA1EmmrOwn_zxSzFE3NCHr_EW2g@mail.gmail.com>
References: <AANLkTimykLH-7i1PivZK8SQHrAFRzuXDMr0QjPPX2Ky3@mail.gmail.com>
	<1301583451169-6227463.post@n2.nabble.com>
	<AANLkTik5ahRedWCETwPhYB8kSqtGckX-vTGA2aXL2ZG1@mail.gmail.com>
	<BANLkTinNQsBgvo44Cd609DqMF4o7knEGDg@mail.gmail.com>
	<BANLkTikwA1EmmrOwn_zxSzFE3NCHr_EW2g@mail.gmail.com>
Message-ID: <BANLkTim7T5rwmOYBZqrWnLu=b-sfCR1S1Q@mail.gmail.com>

Oh yes sorry, here is the link:
http://jira.freeswitch.org/browse/FS-3213

<http://jira.freeswitch.org/browse/FS-3213>Thank you !

2011/4/18 Michael Collins <msc at freeswitch.org>

> For those of us who don't live and breath Jira, could you paste the link or
> at least the bug number?
> -MC
>
>
> On Mon, Apr 18, 2011 at 2:22 PM, Mathieu Lautram <
> lautram.mathieu at gmail.com> wrote:
>
>> Hi,
>>
>> I posted my problem on Jira 3 weeks ago. At first, I had many answers. But
>> now, there's nobody. Is there a solution for my problem? Is this an
>> unfixable bug?
>> I can't fax anything because I'm waiting for an answer or a fix. Can you
>> please have a look on the post and the log that I posted please?
>>
>> Thank you very much :-)
>>
>> 2011/3/31 Mathieu Lautram <lautram.mathieu at gmail.com>
>>
>>> Thank you for your answer.
>>> Yes, I use the last Freeswitch (FreeSWITCH Version 1.0.head (git-6e78f6f
>>> 2011-03-30 11-41-45 +0200)) and the problem is still there and it's always
>>> related to fax calls.
>>> I'm reporting this bug to Jira, hope it will be fix soon.
>>>
>>>
>>> 2011/3/31 Jeff Lenk <jeff at jefflenk.com>
>>>
>>>> So you used make current and the problem still occurs?
>>>>
>>>> Is it always related to fax calls?
>>>>
>>>> If the above is correct you should file this a Jira with all relevant
>>>> information.
>>>>
>>>> see http://wiki.freeswitch.org/wiki/Reporting_Bugs
>>>>
>>>> --
>>>> View this message in context:
>>>> http://freeswitch-users.2379917.n2.nabble.com/stuck-channels-tp6226874p6227463.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
>>>>
>>>
>>>
>>>
>>> --
>>> Mathieu LAUTRAM
>>> Application developer
>>>
>>> BJT Partners - FRANCE
>>> +33 1 79 75 99 60
>>> +33 6 61 59 07 25
>>>
>>
>>
>>
>> --
>> Mathieu LAUTRAM
>> Application developer
>>
>> BJT Partners - FRANCE
>> +33 1 79 75 99 60
>> +33 6 61 59 07 25
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Mathieu LAUTRAM
Application developer

BJT Partners - FRANCE
+33 1 79 75 99 60
+33 6 61 59 07 25
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From lfurrea at gmail.com  Tue Apr 19 02:04:12 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Mon, 18 Apr 2011 16:04:12 -0600
Subject: [Freeswitch-users] FreeTDM disconnect supervision
Message-ID: <BANLkTikwxtkATFaK0jQJhpnsvgOhxJp2RQ@mail.gmail.com>

Hello,

According to what I have found in regards the tones used for signaling on
FreeTDM, it seems that anything set through DAHDI is ignored and specs from
tones.conf are used instead.

However I have not been able to properly detect a busy tone to be able to
set an FXO back on hook once a busy tone is sniffed.

tones.conf has the following references for busy tones [us]:

generate-busy => v=-7;%(500,500,480,620)
detect-busy=> 480,620

I would like to know if detect-busy takes into account the time frame that
the frequencies are on (played) or off (not being played) and if there is
somehow a way to specify this. Is there any default value for time on and
time off?

I appreciate your help!

Regards
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From anthony.minessale at gmail.com  Tue Apr 19 02:16:01 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 18 Apr 2011 17:16:01 -0500
Subject: [Freeswitch-users] Loading xml.so
In-Reply-To: <BANLkTinQSJwUrHYt4vLcMyXBrXgFZs8vMA@mail.gmail.com>
References: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>
	<68AE1930-8D2F-41F3-81E1-FEDF9A26922C@gmail.com>
	<BANLkTinQSJwUrHYt4vLcMyXBrXgFZs8vMA@mail.gmail.com>
Message-ID: <BANLkTikxEbsYK=NtJTbcLz=BBUJb8m27mQ@mail.gmail.com>

FYI 1.0.7 remains unreleased it means GIT HEAD.  There is no 1.0.7

If you want to avoid doing this then you have to end this conversation
because reproducing an issue on GIT HEAD is a strict requirement for
help.  We have a big community and a small number of developers so we
need to normalize the codebase to the latest version when debugging.





On Mon, Apr 18, 2011 at 3:34 PM, Roger Castaldo
<roger.castaldo at gmail.com> wrote:
> I am using 1.0.7, I tend to avoid straight from trunk builds.? As far as the
> xml modules being loaded, the only one being loaded is dialplan_xml.? All
> the other modules are disabled and were not compiled into the system as I
> have no intention of using them.? I will try and do a gcore dump to see what
> is trying to load the xml.so when I have a chance.
>
> On Mon, Apr 18, 2011 at 3:58 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Do a gcore to generate a coredump of the process while it is still
>> running. If you load that into gdb and get a backtrace it might show the
>> culprit.
>>
>> What version are you on? Have you tried the latest got head? If not, it's
>> possible it's a bug that is already fixed.
>>
>> > freeswitch/mod/xml.so
>>
>> That sounds wrong. Check your modules.conf.xml file in your config. Is it
>> trying to load a module named xml? (there is no such module, but there is
>> mod_xml_curl, mod_xml_cdr etc
>>
>> Steve on iPhone
>>
>> On 18 Apr 2011, at 14:37, Roger Castaldo <roger.castaldo at gmail.com> wrote:
>>
>> > Hi I have a freshly compiled install of freeswitch that appears to be
>> > working, but the cpu is sitting at 100% with it doing nothing, no logging
>> > messages, nothing. ?I ran some tests and tried to reload xml to see if there
>> > was a configuration error and it throws a critical message saying
>> > "/freeswitch/mod/xml.so: cannot open shared object file: No such file or
>> > directory". ?I have double checked my configurations and there should not be
>> > any loading of this that I can see, but the file is missing in the mod
>> > directory and I cannot seem to find a reference of it in the source. ?I am
>> > not sure if this is where my issue is with the application but I think this
>> > is a good start, to find out why it was not compiled in and what is using
>> > the library.
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anthony Minessale II

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

AIM: anthm
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FreeSWITCH Developer Conference
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pstn:+19193869900


From msc at freeswitch.org  Tue Apr 19 02:35:37 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Apr 2011 15:35:37 -0700
Subject: [Freeswitch-users] FreeTDM disconnect supervision
In-Reply-To: <BANLkTikwxtkATFaK0jQJhpnsvgOhxJp2RQ@mail.gmail.com>
References: <BANLkTikwxtkATFaK0jQJhpnsvgOhxJp2RQ@mail.gmail.com>
Message-ID: <BANLkTimfov5LO4ph+HjcYcxmJz87dWe1Yw@mail.gmail.com>

In all my experience with these it has been that tones.conf is for
specifying the absence or presence of a frequency but not the cadence.

-MC

On Mon, Apr 18, 2011 at 3:04 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

> Hello,
>
> According to what I have found in regards the tones used for signaling on
> FreeTDM, it seems that anything set through DAHDI is ignored and specs from
> tones.conf are used instead.
>
> However I have not been able to properly detect a busy tone to be able to
> set an FXO back on hook once a busy tone is sniffed.
>
> tones.conf has the following references for busy tones [us]:
>
> generate-busy => v=-7;%(500,500,480,620)
> detect-busy=> 480,620
>
> I would like to know if detect-busy takes into account the time frame that
> the frequencies are on (played) or off (not being played) and if there is
> somehow a way to specify this. Is there any default value for time on and
> time off?
>
> I appreciate your help!
>
> 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 grsingh750 at gmail.com  Tue Apr 19 02:48:08 2011
From: grsingh750 at gmail.com (guru singh)
Date: Tue, 19 Apr 2011 04:18:08 +0530
Subject: [Freeswitch-users] FreeTDM disconnect supervision
In-Reply-To: <BANLkTikwxtkATFaK0jQJhpnsvgOhxJp2RQ@mail.gmail.com>
References: <BANLkTikwxtkATFaK0jQJhpnsvgOhxJp2RQ@mail.gmail.com>
Message-ID: <BANLkTinRmBORb9zO0yvt22sWW8JE2E7g3A@mail.gmail.com>

Hi Luis,

You could try the dialplan application tone_detect to detect the busy
tone and hangup.
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect

guru

On Tue, Apr 19, 2011 at 3:34 AM, Luis F Urrea <lfurrea at gmail.com> wrote:
> Hello,
> According to what I have found in regards the tones used for signaling on
> FreeTDM, it seems that anything set through DAHDI is ignored and specs from
> tones.conf are used instead.
> However I have not been able to properly detect a busy tone to be able to
> set an FXO back on hook once a busy tone is sniffed.
> tones.conf has the following references for busy tones [us]:
> generate-busy => v=-7;%(500,500,480,620)
> detect-busy=> 480,620
> I would like to know if detect-busy takes into account the time frame that
> the frequencies are on (played) or off (not being played) and if there is
> somehow a way to specify this. Is there any default value for time on and
> time off?
> I appreciate your help!
> 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
>
>


From lfurrea at gmail.com  Tue Apr 19 03:27:54 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Mon, 18 Apr 2011 17:27:54 -0600
Subject: [Freeswitch-users] FreeTDM disconnect supervision
In-Reply-To: <BANLkTinRmBORb9zO0yvt22sWW8JE2E7g3A@mail.gmail.com>
References: <BANLkTikwxtkATFaK0jQJhpnsvgOhxJp2RQ@mail.gmail.com>
	<BANLkTinRmBORb9zO0yvt22sWW8JE2E7g3A@mail.gmail.com>
Message-ID: <BANLkTimk_=m2GVZB=c7PtxqZoBJWR0ELMA@mail.gmail.com>

Yes guru,

The point is that using tone_detect I am not able to specify cadence
either.

I need to be able to detect a tone that plays 450Hz for 330ms and then
silence for 330ms.

Strange thing is that on tones.conf generate tones do seem to specify
cadence such as US busy

v=-7;(500,500,420,680)

I would interpret 420Hz for .5s and 680Hz for .5s

Is that how it is to be interpreted??


TIA


On Mon, Apr 18, 2011 at 4:48 PM, guru singh <grsingh750 at gmail.com> wrote:

> Hi Luis,
>
> You could try the dialplan application tone_detect to detect the busy
> tone and hangup.
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>
> guru
>
> On Tue, Apr 19, 2011 at 3:34 AM, Luis F Urrea <lfurrea at gmail.com> wrote:
> > Hello,
> > According to what I have found in regards the tones used for signaling on
> > FreeTDM, it seems that anything set through DAHDI is ignored and specs
> from
> > tones.conf are used instead.
> > However I have not been able to properly detect a busy tone to be able to
> > set an FXO back on hook once a busy tone is sniffed.
> > tones.conf has the following references for busy tones [us]:
> > generate-busy => v=-7;%(500,500,480,620)
> > detect-busy=> 480,620
> > I would like to know if detect-busy takes into account the time frame
> that
> > the frequencies are on (played) or off (not being played) and if there is
> > somehow a way to specify this. Is there any default value for time on and
> > time off?
> > I appreciate your help!
> > 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
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 19 03:38:06 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 18 Apr 2011 16:38:06 -0700
Subject: [Freeswitch-users] FreeTDM disconnect supervision
In-Reply-To: <BANLkTimk_=m2GVZB=c7PtxqZoBJWR0ELMA@mail.gmail.com>
References: <BANLkTikwxtkATFaK0jQJhpnsvgOhxJp2RQ@mail.gmail.com>
	<BANLkTinRmBORb9zO0yvt22sWW8JE2E7g3A@mail.gmail.com>
	<BANLkTimk_=m2GVZB=c7PtxqZoBJWR0ELMA@mail.gmail.com>
Message-ID: <BANLkTikNrdU0aFHUxvGJ=sSfPD=_SzZgJg@mail.gmail.com>

Generating a tone w/ cadence is a piece of cake. *Detecting* a tone and
cadence is a bit trickier. Nothing in FreeTDM (that I'm aware of) can detect
both tone *and* cadence, which is why it works great in the US (which uses a
combination of tones) as opposed to Mexico (which uses, I think 425Hz for
everything).

I'd ask Moises Silva if there's any plans to add the cadence detection.

-MC

On Mon, Apr 18, 2011 at 4:27 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

> Yes guru,
>
> The point is that using tone_detect I am not able to specify cadence
> either.
>
> I need to be able to detect a tone that plays 450Hz for 330ms and then
> silence for 330ms.
>
> Strange thing is that on tones.conf generate tones do seem to specify
> cadence such as US busy
>
> v=-7;(500,500,420,680)
>
> I would interpret 420Hz for .5s and 680Hz for .5s
>
> Is that how it is to be interpreted??
>
>
> TIA
>
>
> On Mon, Apr 18, 2011 at 4:48 PM, guru singh <grsingh750 at gmail.com> wrote:
>
>> Hi Luis,
>>
>> You could try the dialplan application tone_detect to detect the busy
>> tone and hangup.
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>>
>> guru
>>
>> On Tue, Apr 19, 2011 at 3:34 AM, Luis F Urrea <lfurrea at gmail.com> wrote:
>> > Hello,
>> > According to what I have found in regards the tones used for signaling
>> on
>> > FreeTDM, it seems that anything set through DAHDI is ignored and specs
>> from
>> > tones.conf are used instead.
>> > However I have not been able to properly detect a busy tone to be able
>> to
>> > set an FXO back on hook once a busy tone is sniffed.
>> > tones.conf has the following references for busy tones [us]:
>> > generate-busy => v=-7;%(500,500,480,620)
>> > detect-busy=> 480,620
>> > I would like to know if detect-busy takes into account the time frame
>> that
>> > the frequencies are on (played) or off (not being played) and if there
>> is
>> > somehow a way to specify this. Is there any default value for time on
>> and
>> > time off?
>> > I appreciate your help!
>> > 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
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 valery.kalinin at gmail.com  Tue Apr 19 05:44:48 2011
From: valery.kalinin at gmail.com (Valery Kalinin)
Date: Tue, 19 Apr 2011 07:44:48 +0600
Subject: [Freeswitch-users]  Cannot compile freetdm!
Message-ID: <BANLkTim_+KGD6QvcXGru7QZJJ4O-6vS0Ng@mail.gmail.com>

Digium 1TE121BF
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From boris at tagnet.ru  Tue Apr 19 07:15:12 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 19 Apr 2011 09:15:12 +0600
Subject: [Freeswitch-users] regexp with LUA script
In-Reply-To: <BANLkTik+O40aS1MtJJjCm-F1Ccy-W+OjwQ@mail.gmail.com>
References: <4DAC3758.1000405@tagnet.ru>	<BANLkTikebBSkuaTMNZQ6vpFSwVf=FjusTQ@mail.gmail.com>	<4DAC684C.1080907@tagnet.ru>
	<BANLkTik+O40aS1MtJJjCm-F1Ccy-W+OjwQ@mail.gmail.com>
Message-ID: <4DACFE40.6050001@tagnet.ru>

Hello!

     Thank You, Michael! I'll try your recipe.
>
>
> On Mon, Apr 18, 2011 at 9:35 AM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         Yes, I know about LUA native patterns. Unfortunatelly I can't
>     (or don't know how to) do something like:
>     \d{4,6} or \d{10,}
>
>
> \d{4,6} is "%d%d%d%d%d?%d?"
>
> \d{10,} is "%d%d%d%d%d%d%d%d%d%d+"
>
> At first glance that may seem inelegant, however, it is far more 
> efficient than trying to send a regex through an api:execute call.
>
> -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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005

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From boris at tagnet.ru  Tue Apr 19 08:34:50 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 19 Apr 2011 10:34:50 +0600
Subject: [Freeswitch-users] mod_lua and require
Message-ID: <4DAD10EA.2070007@tagnet.ru>

Hello!

     I want to include LUA script into another LUA script. I have 
modified lua.conf.xml and set <param name="script-directory" 
value="$${base_dir}/scripts/?"/> and restarted freeswitch. In my script 
called test1.lua I wrote: require('test2.lua');
test2.lua is placed into the same directory as test1.lua, and directory 
is /opt/fs/scripts/. But luarun test1.lua gives an error:
  2011-04-19 10:14:40.651205 [ERR] mod_lua.cpp:182 
/opt/fs/scripts/test1.lua:3: module 'test2.lua' not found:
         no field package.preload['test2.lua']
         no file '/opt/fs/scripts/test2/lua'
         no file './test2/lua.so'
         no file './libtest2/lua51.so'
         no file '/usr/local/lib/lua/5.1/test2/lua.so'
         no file '/usr/local/lib/lua/5.1/libtest2/lua51.so'
         no file '/usr/local/lib/lua/5.1/loadall.so'
         no file './test2.so'
         no file './libtest251.so'
         no file '/usr/local/lib/lua/5.1/test2.so'
         no file '/usr/local/lib/lua/5.1/libtest251.so'
         no file '/usr/local/lib/lua/5.1/loadall.so'
stack traceback:
         [C]: in function 'require'
         /opt/fs/scripts/test1.lua:3: in main chunk


What is wrong with my setup?

-- 
Regards,
Boris




From curriegrad2004 at gmail.com  Tue Apr 19 10:01:19 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Mon, 18 Apr 2011 23:01:19 -0700
Subject: [Freeswitch-users] Multi-homed freeswitch want to use internal
 interface only
In-Reply-To: <BANLkTikPoeZs0iq4KTFxCRzx5-oNW773tQ@mail.gmail.com>
References: <DD5D873A5D84F949A2051D0F326B82F601462533@SYD5.minesite.com.au>
	<BANLkTikPoeZs0iq4KTFxCRzx5-oNW773tQ@mail.gmail.com>
Message-ID: <BANLkTimNMaj18GFRtYzpDsSvx9q+1=kYjA@mail.gmail.com>

Actually you would configure it with the sofia sip profiles instead of using
the vars.xml file

On Mon, Apr 18, 2011 at 7:07 AM, Roger Castaldo <roger.castaldo at gmail.com>wrote:

> I believe the configuration settings you are looking for would be in the
> /conf/vars.xml and /conf/sip_profiles/internal.xml.  Inside here you can
> specify the ips to bind the internal sip profile too as well as configure
> which ips to bind freeswitch to.  I cannot recall what the settings are
> called but the configuration files are quite well documented within them to
> point you in the right direction.
>
> On Mon, Apr 18, 2011 at 12:26 AM, Dirk Eisner <d.eisner at minesite.com.au>wrote:
>
>>  Hi everyone,
>>
>>
>>
>> I have a server which has multiple network cards configured with different
>> IP addresses in different subnets.
>>
>> I would like to use only one IP address (internal) for the registration of
>> phones.
>>
>>
>>
>> My problem is that I?m having difficulty to stop FreeSwitch from finding
>> the external interface.
>>
>>
>>
>> It there somewhere a easy description of the process?
>>
>>
>>
>> Alternatively I?m happy to pay for some support, If someone would be so
>> friendly to pass on a contact. It would be great if this person is in
>> Australia, but I?m happy with anyone.
>>
>>
>>
>> Thank you for your help
>>
>>
>>
>>
>>
>> **
>>
>> * *
>>
>>
>>
>> *   **Dirk Eisner** **|* *Systems Architect*
>>     *Mine Site Technologies Pty Ltd.*
>>
>>      Tel: 02-9018-0940    *|* Fax: 02-9437-5688    *|*  Mobile:
>> 0458-080-370
>>      25-27 Whiting St * **|* Artarmon, NSW*  *2064
>>      *http://www.minesite.com.au   **|* E-mail: d.eisner at minesite.com.au
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From clive at lansink.co.nz  Tue Apr 19 06:01:40 2011
From: clive at lansink.co.nz (Clive Lansink)
Date: Tue, 19 Apr 2011 14:01:40 +1200
Subject: [Freeswitch-users] Playing tones
Message-ID: <E1QC0Gp-0002ul-3e@mx5.orcon.net.nz>

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From monemran at gmail.com  Tue Apr 19 11:39:51 2011
From: monemran at gmail.com (M.Emran)
Date: Tue, 19 Apr 2011 13:39:51 +0600
Subject: [Freeswitch-users] mod_lua and require
In-Reply-To: <4DAD10EA.2070007@tagnet.ru>
References: <4DAD10EA.2070007@tagnet.ru>
Message-ID: <BANLkTikuW+aZ_P0MpKHv2Ngey-2x=769wQ@mail.gmail.com>

Before run Freeswitch you have to put this command:

export LUA_PATH=/usr/local/freeswitch/scripts/db_connect.lua


On Tue, Apr 19, 2011 at 10:34 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     I want to include LUA script into another LUA script. I have
> modified lua.conf.xml and set <param name="script-directory"
> value="$${base_dir}/scripts/?"/> and restarted freeswitch. In my script
> called test1.lua I wrote: require('test2.lua');
> test2.lua is placed into the same directory as test1.lua, and directory
> is /opt/fs/scripts/. But luarun test1.lua gives an error:
>  2011-04-19 10:14:40.651205 [ERR] mod_lua.cpp:182
> /opt/fs/scripts/test1.lua:3: module 'test2.lua' not found:
>         no field package.preload['test2.lua']
>         no file '/opt/fs/scripts/test2/lua'
>         no file './test2/lua.so'
>         no file './libtest2/lua51.so'
>         no file '/usr/local/lib/lua/5.1/test2/lua.so'
>         no file '/usr/local/lib/lua/5.1/libtest2/lua51.so'
>         no file '/usr/local/lib/lua/5.1/loadall.so'
>         no file './test2.so'
>         no file './libtest251.so'
>         no file '/usr/local/lib/lua/5.1/test2.so'
>         no file '/usr/local/lib/lua/5.1/libtest251.so'
>         no file '/usr/local/lib/lua/5.1/loadall.so'
> stack traceback:
>         [C]: in function 'require'
>         /opt/fs/scripts/test1.lua:3: in main chunk
>
>
> What is wrong with my setup?
>
> --
> 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
>



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

E-Mail: info at e-softbilling.com
Web: www.e-softbilling.com
         www.isoftswitch.com
         www.howtonix.com
         www.sipmobiledialer.com
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From steveayre at gmail.com  Tue Apr 19 11:54:00 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 19 Apr 2011 08:54:00 +0100
Subject: [Freeswitch-users] Loading xml.so
In-Reply-To: <BANLkTinQSJwUrHYt4vLcMyXBrXgFZs8vMA@mail.gmail.com>
References: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>
	<68AE1930-8D2F-41F3-81E1-FEDF9A26922C@gmail.com>
	<BANLkTinQSJwUrHYt4vLcMyXBrXgFZs8vMA@mail.gmail.com>
Message-ID: <BANLkTi=0UAzaAuCQ2AqvUSeLEa1mhYn-BQ@mail.gmail.com>

The 1.0.7 downloads a built daily from trunk.

-Steve


On 18 April 2011 21:34, Roger Castaldo <roger.castaldo at gmail.com> wrote:

> I am using 1.0.7, I tend to avoid straight from trunk builds.  As far as
> the xml modules being loaded, the only one being loaded is dialplan_xml.
> All the other modules are disabled and were not compiled into the system as
> I have no intention of using them.  I will try and do a gcore dump to see
> what is trying to load the xml.so when I have a chance.
>
>
> On Mon, Apr 18, 2011 at 3:58 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> Do a gcore to generate a coredump of the process while it is still
>> running. If you load that into gdb and get a backtrace it might show the
>> culprit.
>>
>> What version are you on? Have you tried the latest got head? If not, it's
>> possible it's a bug that is already fixed.
>>
>> > freeswitch/mod/xml.so
>>
>> That sounds wrong. Check your modules.conf.xml file in your config. Is it
>> trying to load a module named xml? (there is no such module, but there is
>> mod_xml_curl, mod_xml_cdr etc
>>
>> Steve on iPhone
>>
>> On 18 Apr 2011, at 14:37, Roger Castaldo <roger.castaldo at gmail.com>
>> wrote:
>>
>> > Hi I have a freshly compiled install of freeswitch that appears to be
>> working, but the cpu is sitting at 100% with it doing nothing, no logging
>> messages, nothing.  I ran some tests and tried to reload xml to see if there
>> was a configuration error and it throws a critical message saying
>> "/freeswitch/mod/xml.so: cannot open shared object file: No such file or
>> directory".  I have double checked my configurations and there should not be
>> any loading of this that I can see, but the file is missing in the mod
>> directory and I cannot seem to find a reference of it in the source.  I am
>> not sure if this is where my issue is with the application but I think this
>> is a good start, to find out why it was not compiled in and what is using
>> the library.
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 19 12:00:39 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 19 Apr 2011 14:00:39 +0600
Subject: [Freeswitch-users] mod_lua and require
In-Reply-To: <BANLkTikuW+aZ_P0MpKHv2Ngey-2x=769wQ@mail.gmail.com>
References: <4DAD10EA.2070007@tagnet.ru>
	<BANLkTikuW+aZ_P0MpKHv2Ngey-2x=769wQ@mail.gmail.com>
Message-ID: <4DAD4127.8060402@tagnet.ru>

Hello!

     Would You please explain why?? I have no db_connect.lua in my 
scripts directory and also no freeswitch installed in 
/usr/local/freeswitch.

> Before run Freeswitch you have to put this command:
>
> export LUA_PATH=/usr/local/freeswitch/scripts/db_connect.lua
>
>
> On Tue, Apr 19, 2011 at 10:34 AM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         I want to include LUA script into another LUA script. I have
>     modified lua.conf.xml and set <param name="script-directory"
>     value="$${base_dir}/scripts/?"/> and restarted freeswitch. In my
>     script
>     called test1.lua I wrote: require('test2.lua');
>     test2.lua is placed into the same directory as test1.lua, and
>     directory
>     is /opt/fs/scripts/. But luarun test1.lua gives an error:
>      2011-04-19 10:14:40.651205 [ERR] mod_lua.cpp:182
>     /opt/fs/scripts/test1.lua:3: module 'test2.lua' not found:
>             no field package.preload['test2.lua']
>             no file '/opt/fs/scripts/test2/lua'
>             no file './test2/lua.so'
>             no file './libtest2/lua51.so'
>             no file '/usr/local/lib/lua/5.1/test2/lua.so'
>             no file '/usr/local/lib/lua/5.1/libtest2/lua51.so'
>             no file '/usr/local/lib/lua/5.1/loadall.so'
>             no file './test2.so'
>             no file './libtest251.so'
>             no file '/usr/local/lib/lua/5.1/test2.so'
>             no file '/usr/local/lib/lua/5.1/libtest251.so'
>             no file '/usr/local/lib/lua/5.1/loadall.so'
>     stack traceback:
>             [C]: in function 'require'
>             /opt/fs/scripts/test1.lua:3: in main chunk
>
>
>     What is wrong with my setup?
>
>     --
>     Regards,
>     Boris
>
>
>
>     _______________________________________________
>     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
>
>
>
>
> -- 
> Regards
> ----------
> M Emran
>
> E-Mail: info at e-softbilling.com <mailto:info at e-softbilling.com>
> Web: www.e-softbilling.com <http://www.e-softbilling.com/>
> www.isoftswitch.com <http://www.isoftswitch.com/>
> www.howtonix.com <http://www.howtonix.com/>
> www.sipmobiledialer.com <http://www.sipmobiledialer.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


-- 
Regards,
Boris


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From monemran at gmail.com  Tue Apr 19 12:09:02 2011
From: monemran at gmail.com (M.Emran)
Date: Tue, 19 Apr 2011 14:09:02 +0600
Subject: [Freeswitch-users] mod_lua and require
In-Reply-To: <4DAD4127.8060402@tagnet.ru>
References: <4DAD10EA.2070007@tagnet.ru>
	<BANLkTikuW+aZ_P0MpKHv2Ngey-2x=769wQ@mail.gmail.com>
	<4DAD4127.8060402@tagnet.ru>
Message-ID: <BANLkTi=rPGC2FA0FdNhO-vXn-6wYBfgfZA@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_get_Lua_to_see_my_own_libraries_using_.22require.22


On Tue, Apr 19, 2011 at 2:00 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

>  Hello!
>
>     Would You please explain why?? I have no db_connect.lua in my scripts
> directory and also no freeswitch installed in /usr/local/freeswitch.
>
> Before run Freeswitch you have to put this command:
>
> export LUA_PATH=/usr/local/freeswitch/scripts/db_connect.lua
>
>
> On Tue, Apr 19, 2011 at 10:34 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
>
>> Hello!
>>
>>     I want to include LUA script into another LUA script. I have
>> modified lua.conf.xml and set <param name="script-directory"
>> value="$${base_dir}/scripts/?"/> and restarted freeswitch. In my script
>> called test1.lua I wrote: require('test2.lua');
>> test2.lua is placed into the same directory as test1.lua, and directory
>> is /opt/fs/scripts/. But luarun test1.lua gives an error:
>>  2011-04-19 10:14:40.651205 [ERR] mod_lua.cpp:182
>> /opt/fs/scripts/test1.lua:3: module 'test2.lua' not found:
>>         no field package.preload['test2.lua']
>>         no file '/opt/fs/scripts/test2/lua'
>>         no file './test2/lua.so'
>>         no file './libtest2/lua51.so'
>>         no file '/usr/local/lib/lua/5.1/test2/lua.so'
>>         no file '/usr/local/lib/lua/5.1/libtest2/lua51.so'
>>         no file '/usr/local/lib/lua/5.1/loadall.so'
>>         no file './test2.so'
>>         no file './libtest251.so'
>>         no file '/usr/local/lib/lua/5.1/test2.so'
>>         no file '/usr/local/lib/lua/5.1/libtest251.so'
>>         no file '/usr/local/lib/lua/5.1/loadall.so'
>> stack traceback:
>>         [C]: in function 'require'
>>         /opt/fs/scripts/test1.lua:3: in main chunk
>>
>>
>> What is wrong with my setup?
>>
>> --
>> 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
>>
>
>
>
> --
> Regards
> ----------
> M Emran
>
> E-Mail: info at e-softbilling.com
> Web: www.e-softbilling.com
>          www.isoftswitch.com
>          www.howtonix.com
>          www.sipmobiledialer.com
>
>
> _______________________________________________
> 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
>
>
>
> --
> 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
>
>


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

E-Mail: info at e-softbilling.com
Web: www.e-softbilling.com
         www.isoftswitch.com
         www.howtonix.com
         www.sipmobiledialer.com
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From boris at tagnet.ru  Tue Apr 19 12:24:10 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 19 Apr 2011 14:24:10 +0600
Subject: [Freeswitch-users] mod_lua and require
In-Reply-To: <BANLkTi=rPGC2FA0FdNhO-vXn-6wYBfgfZA@mail.gmail.com>
References: <4DAD10EA.2070007@tagnet.ru>	<BANLkTikuW+aZ_P0MpKHv2Ngey-2x=769wQ@mail.gmail.com>	<4DAD4127.8060402@tagnet.ru>
	<BANLkTi=rPGC2FA0FdNhO-vXn-6wYBfgfZA@mail.gmail.com>
Message-ID: <4DAD46AA.40200@tagnet.ru>

Thank You!

> http://wiki.freeswitch.org/wiki/Mod_lua#How_can_I_get_Lua_to_see_my_own_libraries_using_.22require.22
>
>
> On Tue, Apr 19, 2011 at 2:00 PM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         Would You please explain why?? I have no db_connect.lua in my
>     scripts directory and also no freeswitch installed in
>     /usr/local/freeswitch.
>
>>     Before run Freeswitch you have to put this command:
>>
>>     export LUA_PATH=/usr/local/freeswitch/scripts/db_connect.lua
>>
>>
>>     On Tue, Apr 19, 2011 at 10:34 AM, Boris Kovalenko
>>     <boris at tagnet.ru <mailto:boris at tagnet.ru>> wrote:
>>
>>         Hello!
>>
>>             I want to include LUA script into another LUA script. I have
>>         modified lua.conf.xml and set <param name="script-directory"
>>         value="$${base_dir}/scripts/?"/> and restarted freeswitch. In
>>         my script
>>         called test1.lua I wrote: require('test2.lua');
>>         test2.lua is placed into the same directory as test1.lua, and
>>         directory
>>         is /opt/fs/scripts/. But luarun test1.lua gives an error:
>>          2011-04-19 10:14:40.651205 [ERR] mod_lua.cpp:182
>>         /opt/fs/scripts/test1.lua:3: module 'test2.lua' not found:
>>                 no field package.preload['test2.lua']
>>                 no file '/opt/fs/scripts/test2/lua'
>>                 no file './test2/lua.so'
>>                 no file './libtest2/lua51.so'
>>                 no file '/usr/local/lib/lua/5.1/test2/lua.so'
>>                 no file '/usr/local/lib/lua/5.1/libtest2/lua51.so'
>>                 no file '/usr/local/lib/lua/5.1/loadall.so'
>>                 no file './test2.so'
>>                 no file './libtest251.so'
>>                 no file '/usr/local/lib/lua/5.1/test2.so'
>>                 no file '/usr/local/lib/lua/5.1/libtest251.so'
>>                 no file '/usr/local/lib/lua/5.1/loadall.so'
>>         stack traceback:
>>                 [C]: in function 'require'
>>                 /opt/fs/scripts/test1.lua:3: in main chunk
>>
>>
>>         What is wrong with my setup?
>>
>>         --
>>         Regards,
>>         Boris
>>
>>
>>
>>         _______________________________________________
>>         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
>>
>>
>>
>>
>>     -- 
>>     Regards
>>     ----------
>>     M Emran
>>
>>     E-Mail: info at e-softbilling.com <mailto:info at e-softbilling.com>
>>     Web: www.e-softbilling.com <http://www.e-softbilling.com/>
>>     www.isoftswitch.com <http://www.isoftswitch.com/>
>>     www.howtonix.com <http://www.howtonix.com/>
>>     www.sipmobiledialer.com <http://www.sipmobiledialer.com/>
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>
>     -- 
>     Regards,
>     Boris
>
>
>
>     _______________________________________________
>     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
>
>
>
>
> -- 
> Regards
> ----------
> M Emran
>
> E-Mail: info at e-softbilling.com <mailto:info at e-softbilling.com>
> Web: www.e-softbilling.com <http://www.e-softbilling.com/>
> www.isoftswitch.com <http://www.isoftswitch.com/>
> www.howtonix.com <http://www.howtonix.com/>
> www.sipmobiledialer.com <http://www.sipmobiledialer.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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005

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From peter.olsson at visionutveckling.se  Tue Apr 19 12:36:18 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 19 Apr 2011 10:36:18 +0200
Subject: [Freeswitch-users] mod_h323
In-Reply-To: <BANLkTik_sVoF_hdpxPEi5Rc3JNPpKdTxHQ@mail.gmail.com>
References: <BANLkTik_sVoF_hdpxPEi5Rc3JNPpKdTxHQ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C50EE4AA@cooper>

Yes, I'm using it - not so much though, maybe 50 calls a day.

Try using ptlib 2.8.3 and h323plus 1.22.0 - that's what I'm using here. Though it seems it might be a locking issue, what platform are you on - could you get a core next time it happens?

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Vedran Zeljeznak
Skickat: den 18 april 2011 09:26
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] mod_h323

hello,

has anybody compiled and ran mod_h323 with success lately?

I'm having trouble bridging two H.323 calls via mod_h323.
First two calls bridge relatively successfully, eg. some problems occur when called side hangs up the call first, the calling side of the call doesn't hangup automatically.
The third call however doesn't create any debuging messages in FS console. Looks like something remained locked in the first call.

My FS version is: "FreeSWITCH Version 1.0.head (git-413d590 2011-04-14 19-56-45 -0500)"
OpenH.323 version: h323plus-20100525
PTLib version: 2.8.2 - http://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/tags/v2_8_2/

i've put freeswitch.log file to pastebin (http://pastebin.freeswitch.org/16108).

Do you have any suggestions on where did i go wrong or where to start looking for a bug?

---
Vedran Zeljeznak
!DSPAM:4dac440e32761616010430!
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From sadhika at gmail.com  Tue Apr 19 13:43:09 2011
From: sadhika at gmail.com (Sadhika Sharma)
Date: Tue, 19 Apr 2011 15:13:09 +0530
Subject: [Freeswitch-users] Help with rxfax debug "T4 expires in phase
	T30_PHASE_B_RX"
In-Reply-To: <BANLkTimW6WXQ+_Ge+ASG5u1bVe_j36pTHw@mail.gmail.com>
References: <BANLkTimW6WXQ+_Ge+ASG5u1bVe_j36pTHw@mail.gmail.com>
Message-ID: <BANLkTikAz7Rcci6g_95mmJD7xAm=iHbZTA@mail.gmail.com>

Anybody having any idea about this?

On Fri, Apr 15, 2011 at 7:36 PM, Sadhika Sharma <sadhika at gmail.com> wrote:

> Hi,
>
> I need help in debugging hangup cause 548, 'Disconnected after permitted
> retries' while attempting rxfax().
>
> Logs show that T4 expires in phase T30_PHASE_B_RX.
> T30 PHASE B concerns exchange of DIS/DCS messages. After sending DIS, mod
> spandsp keeps waiting for DCS and exhausts its retries.
>
> Please guide me how to debug the root cause of loss of DCS message. Logs
> are mentioned below:
>
> 2011-03-07 15:53:48.675180 [DEBUG] switch_ivr.c:557 OpenZAP/4:1/43851609
> Command Execute rxfax(/srv/fax/in/fcf15a4c-48a4-11e0-919e-7bf53bcc3f81.tiff)
> 2011-03-07 15:53:53.334614 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send
> complete in phase T30_PHASE_B_TX, state 17
> 2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 DIS:
> 2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ....
> ...0= Store and forward Internet fax (T.37): Not set
> 2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ....
> .0..= Real-time Internet fax (T.38): Not set
> :
> :
> 2011-03-07 15:53:53.875238 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Start T4
> 2011-03-07 15:53:53.895315 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC
> signal status is Carrier up (-2) in state 17
> 2011-03-07 15:53:53.915402 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC
> signal status is Carrier down (-1) in state 17
> 2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 T4
> expired in phase T30_PHASE_B_RX, state 17
> 2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Retry
> number 1
>
>
> Please guide me how to debug the root cause of loss of DCS message.
> Thanks,
> Sadhika
>



-- 
Sadhika
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From habib at alexcoder.com  Tue Apr 19 14:07:05 2011
From: habib at alexcoder.com (Mohammed Habib)
Date: Tue, 19 Apr 2011 12:07:05 +0200
Subject: [Freeswitch-users] Originated session callback.
In-Reply-To: <BANLkTik+HScKABdarAXUUJZUkHAcV1b0SQ@mail.gmail.com>
References: <BANLkTinje48f3MSjB6bRWqpfgTi-xsxC8A@mail.gmail.com>
	<BANLkTik81bAyQiwxJym=KH3VdAD-oTPiqA@mail.gmail.com>
	<BANLkTi=JeAnNfvwrb-Y2OnGQjJRNjD-vHw@mail.gmail.com>
	<BANLkTik+HScKABdarAXUUJZUkHAcV1b0SQ@mail.gmail.com>
Message-ID: <BANLkTi=FOj_FrgTjRCGyZ-hPLM9yrdsbkA@mail.gmail.com>

I was hoping to make a scenario like this:
User 1 calls and uses ASR to set some information (like the pizza demo),
then User 2 is called and confirms User 1 using ASR too.
So, I need to have both both sessions at the same time and both working with
ASR and TTS.

Thank you,
Mohammed Habib

On Mon, Apr 18, 2011 at 11:03 PM, Michael Collins <msc at freeswitch.org>wrote:

> What are you trying to accomplish with this script? Why are you creating a
> new session right in the middle of handling an existing session? I am
> curious to know what problem you are attempting to solve.
>
> -MC
>
> On Mon, Apr 18, 2011 at 5:35 AM, Mohammed Habib <habib at alexcoder.com>wrote:
>
>> I need help getting events from originated session.
>>
>> This is my lua script:
>>
>> function onInput_MainSession(s, type, obj)
>>    -- This one is working fine.
>>    freeswitch.consoleLog("info", "Callback with type " .. type .. "\n");
>> end
>>
>> function onInput_NewSession(s, type, obj)
>>    -- This one is never called.
>>    freeswitch.consoleLog("info", "Callback with type " .. type .. "\n");
>> end
>>
>> session:answer();
>> session:setInputCallback("onInput_MainSession");
>> session:sleep(200);
>> session:execute("detect_speech", "unimrcp testgrammer trestgrammer");
>>
>> newsession = freeswitch.Session("user/1002");
>> newsession:setInputCallback("onInput_NewSession");
>> newsession:sleep(200);
>> newsession:execute("detect_speech", "unimrcp testgrammer trestgrammer");
>>
>> while ((session:ready() == true) ) and (newsession:ready() == true) do
>>     -- Loop
>>     sleep(200);
>> end
>>
>> I am unable to capture any of the new session events or dtmf.
>>
>> Please help.
>>
>> Thank you,
>> Mohammed Habib
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 roger.castaldo at gmail.com  Tue Apr 19 16:42:07 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Tue, 19 Apr 2011 08:42:07 -0400
Subject: [Freeswitch-users] Loading xml.so
In-Reply-To: <BANLkTi=0UAzaAuCQ2AqvUSeLEa1mhYn-BQ@mail.gmail.com>
References: <BANLkTikxGrNvmDZCxCkj=auYYiRedVaChQ@mail.gmail.com>
	<68AE1930-8D2F-41F3-81E1-FEDF9A26922C@gmail.com>
	<BANLkTinQSJwUrHYt4vLcMyXBrXgFZs8vMA@mail.gmail.com>
	<BANLkTi=0UAzaAuCQ2AqvUSeLEa1mhYn-BQ@mail.gmail.com>
Message-ID: <BANLkTi=RYodPHs98nBVCGRghgJPm1SV8=g@mail.gmail.com>

Okay so sorry guys apparently I am two for two on this, should have stayed
in bed, just checked through my command history and instead of runnin
reloadxml I ran reload xml, which told freeswitch to reload a module that
just doesn't exist, sorry guys, still a great product and I would offer help
as a developer but my time with C went the way of the dodo bird.

On Tue, Apr 19, 2011 at 3:54 AM, Steven Ayre <steveayre at gmail.com> wrote:

> The 1.0.7 downloads a built daily from trunk.
>
> -Steve
>
>
>
> On 18 April 2011 21:34, Roger Castaldo <roger.castaldo at gmail.com> wrote:
>
>> I am using 1.0.7, I tend to avoid straight from trunk builds.  As far as
>> the xml modules being loaded, the only one being loaded is dialplan_xml.
>> All the other modules are disabled and were not compiled into the system as
>> I have no intention of using them.  I will try and do a gcore dump to see
>> what is trying to load the xml.so when I have a chance.
>>
>>
>> On Mon, Apr 18, 2011 at 3:58 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> Do a gcore to generate a coredump of the process while it is still
>>> running. If you load that into gdb and get a backtrace it might show the
>>> culprit.
>>>
>>> What version are you on? Have you tried the latest got head? If not, it's
>>> possible it's a bug that is already fixed.
>>>
>>> > freeswitch/mod/xml.so
>>>
>>> That sounds wrong. Check your modules.conf.xml file in your config. Is it
>>> trying to load a module named xml? (there is no such module, but there is
>>> mod_xml_curl, mod_xml_cdr etc
>>>
>>> Steve on iPhone
>>>
>>> On 18 Apr 2011, at 14:37, Roger Castaldo <roger.castaldo at gmail.com>
>>> wrote:
>>>
>>> > Hi I have a freshly compiled install of freeswitch that appears to be
>>> working, but the cpu is sitting at 100% with it doing nothing, no logging
>>> messages, nothing.  I ran some tests and tried to reload xml to see if there
>>> was a configuration error and it throws a critical message saying
>>> "/freeswitch/mod/xml.so: cannot open shared object file: No such file or
>>> directory".  I have double checked my configurations and there should not be
>>> any loading of this that I can see, but the file is missing in the mod
>>> directory and I cannot seem to find a reference of it in the source.  I am
>>> not sure if this is where my issue is with the application but I think this
>>> is a good start, to find out why it was not compiled in and what is using
>>> the library.
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 19 17:01:37 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 19 Apr 2011 21:01:37 +0800
Subject: [Freeswitch-users] Help with rxfax debug "T4 expires in phase
 T30_PHASE_B_RX"
In-Reply-To: <BANLkTikAz7Rcci6g_95mmJD7xAm=iHbZTA@mail.gmail.com>
References: <BANLkTimW6WXQ+_Ge+ASG5u1bVe_j36pTHw@mail.gmail.com>
	<BANLkTikAz7Rcci6g_95mmJD7xAm=iHbZTA@mail.gmail.com>
Message-ID: <4DAD87B1.7020102@coppice.org>

Sadhika,

If you post only a fragment of a log, and describe only half of what 
happens, people will generally ignore what you post. I guess they don't 
like being treated like they are mind readers.

The error seems pretty clear. You aren't getting anything from the far 
end. Since you give no clue as to what you are doing, beyond trying to 
receive FAX, that's about as detailed an analysis as you are going to get.

Steve


On 04/19/2011 05:43 PM, Sadhika Sharma wrote:
> Anybody having any idea about this?
>
> On Fri, Apr 15, 2011 at 7:36 PM, Sadhika Sharma <sadhika at gmail.com 
> <mailto:sadhika at gmail.com>> wrote:
>
>     Hi,
>
>     I need help in debugging hangup cause 548, 'Disconnected after
>     permitted retries' while attempting rxfax().
>
>     Logs show that T4 expires in phase T30_PHASE_B_RX.
>     T30 PHASE B concerns exchange of DIS/DCS messages. After sending
>     DIS, mod spandsp keeps waiting for DCS and exhausts its retries.
>
>     Please guide me how to debug the root cause of loss of DCS
>     message. Logs are mentioned below:
>
>     2011-03-07 15:53:48.675180 [DEBUG] switch_ivr.c:557
>     OpenZAP/4:1/43851609 Command Execute
>     rxfax(/srv/fax/in/fcf15a4c-48a4-11e0-919e-7bf53bcc3f81.tiff)
>     2011-03-07 15:53:53.334614 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>     Send complete in phase T30_PHASE_B_TX, state 17
>     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>     DIS:
>     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW
>     T.30   .... ...0= Store and forward Internet fax (T.37): Not set
>     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW
>     T.30   .... .0..= Real-time Internet fax (T.38): Not set
>     :
>     :
>     2011-03-07 15:53:53.875238 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>     Start T4
>     2011-03-07 15:53:53.895315 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>     HDLC signal status is Carrier up (-2) in state 17
>     2011-03-07 15:53:53.915402 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>     HDLC signal status is Carrier down (-1) in state 17
>     2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>     T4 expired in phase T30_PHASE_B_RX, state 17
>     2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>     Retry number 1
>
>
>     Please guide me how to debug the root cause of loss of DCS message.
>     Thanks,
>     Sadhika
>
>
>
>
> -- 
> Sadhika
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From sadhika at gmail.com  Tue Apr 19 17:45:33 2011
From: sadhika at gmail.com (Sadhika Sharma)
Date: Tue, 19 Apr 2011 19:15:33 +0530
Subject: [Freeswitch-users] Help with rxfax debug "T4 expires in phase
	T30_PHASE_B_RX"
In-Reply-To: <4DAD87B1.7020102@coppice.org>
References: <BANLkTimW6WXQ+_Ge+ASG5u1bVe_j36pTHw@mail.gmail.com>
	<BANLkTikAz7Rcci6g_95mmJD7xAm=iHbZTA@mail.gmail.com>
	<4DAD87B1.7020102@coppice.org>
Message-ID: <BANLkTikjeWYgw5jJf0Gr0dyFj4_BGd01Zw@mail.gmail.com>

Thanks Steve for replying,

I will explain the problem in clearer terms. Whenever a call lands on
freeswitch machine, I exceute rxfax(). I get about 5000 fax calls every
month, out of which about 800 transmissions fail due to hangup cause 548
'Disconnected after permitted retries' Complete log is attached.

On Tue, Apr 19, 2011 at 6:31 PM, Steve Underwood <steveu at coppice.org> wrote:

> Sadhika,
>
> If you post only a fragment of a log, and describe only half of what
> happens, people will generally ignore what you post. I guess they don't
> like being treated like they are mind readers.
>
> The error seems pretty clear. You aren't getting anything from the far
> end. Since you give no clue as to what you are doing, beyond trying to
> receive FAX, that's about as detailed an analysis as you are going to get.
>
> Steve
>
>
> On 04/19/2011 05:43 PM, Sadhika Sharma wrote:
> > Anybody having any idea about this?
> >
> > On Fri, Apr 15, 2011 at 7:36 PM, Sadhika Sharma <sadhika at gmail.com
> > <mailto:sadhika at gmail.com>> wrote:
> >
> >     Hi,
> >
> >     I need help in debugging hangup cause 548, 'Disconnected after
> >     permitted retries' while attempting rxfax().
> >
> >     Logs show that T4 expires in phase T30_PHASE_B_RX.
> >     T30 PHASE B concerns exchange of DIS/DCS messages. After sending
> >     DIS, mod spandsp keeps waiting for DCS and exhausts its retries.
> >
> >     Please guide me how to debug the root cause of loss of DCS
> >     message. Logs are mentioned below:
> >
> >     2011-03-07 15:53:48.675180 [DEBUG] switch_ivr.c:557
> >     OpenZAP/4:1/43851609 Command Execute
> >     rxfax(/srv/fax/in/fcf15a4c-48a4-11e0-919e-7bf53bcc3f81.tiff)
> >     2011-03-07 15:53:53.334614 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
> >     Send complete in phase T30_PHASE_B_TX, state 17
> >     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
> >     DIS:
> >     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW
> >     T.30   .... ...0= Store and forward Internet fax (T.37): Not set
> >     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW
> >     T.30   .... .0..= Real-time Internet fax (T.38): Not set
> >     :
> >     :
> >     2011-03-07 15:53:53.875238 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
> >     Start T4
> >     2011-03-07 15:53:53.895315 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
> >     HDLC signal status is Carrier up (-2) in state 17
> >     2011-03-07 15:53:53.915402 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
> >     HDLC signal status is Carrier down (-1) in state 17
> >     2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
> >     T4 expired in phase T30_PHASE_B_RX, state 17
> >     2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
> >     Retry number 1
> >
> >
> >     Please guide me how to debug the root cause of loss of DCS message.
> >     Thanks,
> >     Sadhika
> >
> >
> >
> >
> > --
> > Sadhika
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Sadhika
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2011-03-07 15:53:47.662495 [NOTICE] ozmod_libpri.c:772 -- Ring on channel 4:1 (from 7172225287 to 43851609)
2011-03-07 15:53:47.662495 [DEBUG] ozmod_libpri.c:793 Changing state on 4:1 from DOWN to RING
2011-03-07 15:53:47.662495 [DEBUG] ozmod_libpri.c:429 4:1 STATE [RING]
2011-03-07 15:53:47.662495 [DEBUG] mod_openzap.c:1934 got clear channel sig [START]
2011-03-07 15:53:47.662495 [DEBUG] mod_openzap.c:403 Set codec PCMA 20ms
2011-03-07 15:53:47.662495 [DEBUG] mod_openzap.c:1439 Connect inbound channel OpenZAP/4:1/43851609
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [NOTICE] switch_channel.c:779 New Channel OpenZAP/4:1/43851609 [fcf15a4c-48a4-11e0-919e-7bf53bcc3f81]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] mod_openzap.c:1450 (OpenZAP/4:1/43851609) State Change CS_NEW -> CS_INIT
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_core_session.c:1047 Send signal OpenZAP/4:1/43851609 [BREAK]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/4:1/43851609) Running State Change CS_INIT
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/4:1/43851609) State INIT
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] mod_openzap.c:431 (OpenZAP/4:1/43851609) State Change CS_INIT -> CS_ROUTING
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_core_session.c:1047 Send signal OpenZAP/4:1/43851609 [BREAK]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_core_state_machine.c:338 (OpenZAP/4:1/43851609) State INIT going to sleep
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/4:1/43851609) Running State Change CS_ROUTING
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_channel.c:1598 (OpenZAP/4:1/43851609) Callstate Change DOWN -> RINGING
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/4:1/43851609) State ROUTING
2011-03-07 15:53:47.662495 [DEBUG] mod_openzap.c:454 OpenZAP/4:1/43851609 CHANNEL ROUTING
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_core_state_machine.c:77 OpenZAP/4:1/43851609 Standard ROUTING
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [INFO] mod_dialplan_xml.c:331 Processing 7172225287 <7172225287>->43851609 in context default
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 Dialplan: OpenZAP/4:1/43851609 parsing [default->unloop] continue=false
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 Dialplan: OpenZAP/4:1/43851609 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 Dialplan: OpenZAP/4:1/43851609 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 Dialplan: OpenZAP/4:1/43851609 parsing [default->knowlus] continue=false
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 Dialplan: OpenZAP/4:1/43851609 Regex (PASS) [knowlus] destination_number(43851609) =~ /^\d+$/ break=on-false
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 Dialplan: OpenZAP/4:1/43851609 Action socket(0.0.0.0:8888 async full) 
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 Dialplan: OpenZAP/4:1/43851609 Action hangup() 
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_core_state_machine.c:119 (OpenZAP/4:1/43851609) State Change CS_ROUTING -> CS_EXECUTE
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_core_session.c:1047 Send signal OpenZAP/4:1/43851609 [BREAK]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_core_state_machine.c:341 (OpenZAP/4:1/43851609) State ROUTING going to sleep
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/4:1/43851609) Running State Change CS_EXECUTE
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_core_state_machine.c:348 (OpenZAP/4:1/43851609) State EXECUTE
2011-03-07 15:53:47.662495 [DEBUG] mod_openzap.c:471 OpenZAP/4:1/43851609 CHANNEL EXECUTE
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:47.662495 [DEBUG] switch_core_state_machine.c:157 OpenZAP/4:1/43851609 Standard EXECUTE
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 EXECUTE OpenZAP/4:1/43851609 socket(0.0.0.0:8888 async full)
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 DIS:
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= 3G mobile network: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..1.= Can receive fax: Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= 2-D coding: Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 10..= Recording length: Unlimited
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .1.. ....= T.6 coding: Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Polled sub-address: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= T.43 coding: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...0 ....= Plane interleave: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Selective polling: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Sub-addressing: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Password: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Basic transfer mode (BTM): Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Ready to transfer a character or mixed mode document (polling): Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= Character mode: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Mixed mode (Annex E/T.4): Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Processable mode 26 (Rec. T.505): Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Digital network capability: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Duplex capability: Half only
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= JPEG coding: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...0 ....= Full colour mode: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= 12bits/pel component: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= No subsampling (1:1:1): Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Custom illuminant: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Custom gamut range: Not set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 1...= North American Letter (215.9mm x 279.4mm): Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...1 ....= North American Legal (215.9mm x 355.6mm): Set
2011-03-07 15:53:47.960824 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set
2011-03-07 15:53:47.961834 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set
2011-03-07 15:53:47.961834 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   0... ....= Extension indicator: Not set
2011-03-07 15:53:47.961834 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  DIS with final frame tag
2011-03-07 15:53:47.961834 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  ff 13 80 00 ee fa c4 80 95 80 80 80 18
2011-03-07 15:53:48.421113 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2011-03-07 15:53:48.500752 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2011-03-07 15:53:48.500752 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX
2011-03-07 15:53:48.500752 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set rx type 4
2011-03-07 15:53:48.500752 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set tx type 0
2011-03-07 15:53:48.500752 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Start T4
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:48.550632 [DEBUG] switch_core_session.c:885 Send signal OpenZAP/4:1/43851609 [BREAK]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:48.564746 [DEBUG] switch_ivr.c:557 OpenZAP/4:1/43851609 Command Execute answer()
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 EXECUTE OpenZAP/4:1/43851609 answer()
2011-03-07 15:53:48.624188 [DEBUG] ozmod_libpri.c:429 4:1 STATE [PROGRESS]
2011-03-07 15:53:48.634284 [DEBUG] ozmod_libpri.c:429 4:1 STATE [PROGRESS_MEDIA]
2011-03-07 15:53:48.645378 [DEBUG] ozmod_libpri.c:429 4:1 STATE [UP]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:48.655456 [DEBUG] switch_core_session.c:666 Send signal OpenZAP/4:1/43851609 [BREAK]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:48.655456 [DEBUG] switch_channel.c:2677 (OpenZAP/4:1/43851609) Callstate Change RINGING -> ACTIVE
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:48.655456 [NOTICE] mod_dptools.c:746 Channel [OpenZAP/4:1/43851609] has been answered
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:48.658061 [DEBUG] switch_core_session.c:885 Send signal OpenZAP/4:1/43851609 [BREAK]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:48.675180 [DEBUG] switch_ivr.c:557 OpenZAP/4:1/43851609 Command Execute rxfax(/srv/fax/in/fcf15a4c-48a4-11e0-919e-7bf53bcc3f81.tiff)
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 EXECUTE OpenZAP/4:1/43851609 rxfax(/srv/fax/in/fcf15a4c-48a4-11e0-919e-7bf53bcc3f81.tiff)
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:48.675180 [DEBUG] mod_spandsp_fax.c:1092 Raw read codec activation Success L16 20000
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:48.675180 [DEBUG] switch_core_codec.c:116 OpenZAP/4:1/43851609 Push codec L16:10
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:53:48.675180 [DEBUG] mod_spandsp_fax.c:1108 Raw write codec activation Success L16
2011-03-07 15:53:48.681288 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2011-03-07 15:53:48.880866 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Framing OK (-6) in state 17
2011-03-07 15:53:48.880866 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Start T4A
2011-03-07 15:53:49.174661 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1
2011-03-07 15:53:49.295376 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Abort (-8) in state 1
2011-03-07 15:53:49.751053 [DEBUG] ozmod_libpri.c:106 T203 counter expired, sending RR and scheduling T203 again
2011-03-07 15:53:49.751053 [DEBUG] ozmod_libpri.c:106 Sending Receiver Ready (55)
2011-03-07 15:53:49.751053 [DEBUG] ozmod_libpri.c:106 
> [ 00 01 01 6f ]
2011-03-07 15:53:49.751053 [DEBUG] ozmod_libpri.c:106 
> Supervisory frame:
2011-03-07 15:53:49.751053 [DEBUG] ozmod_libpri.c:106 > SAPI: 00  C/R: 0 EA: 0
>  TEI: 000        EA: 1
2011-03-07 15:53:49.751053 [DEBUG] ozmod_libpri.c:106 > Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
> N(R): 055 P/F: 1
> 0 bytes of data
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 
< [ 02 01 01 9f ]
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 
< Supervisory frame:
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 < SAPI: 00  C/R: 1 EA: 0
<  TEI: 000        EA: 1
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 < Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
< N(R): 079 P/F: 1
< 0 bytes of data
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 Handling message for SAPI/TEI=0/0
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 -- ACKing all packets from 78 to (but not including) 79
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 -- Since there was nothing left, stopping T200 counter
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 -- Stopping T203 counter since we got an ACK
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 -- Nothing left, starting T203 counter
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 -- Unsolicited RR with P/F bit, responding
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 Sending Receiver Ready (55)
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 
> [ 02 01 01 6f ]
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 
> Supervisory frame:
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 > SAPI: 00  C/R: 1 EA: 0
>  TEI: 000        EA: 1
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 > Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
> N(R): 055 P/F: 1
> 0 bytes of data
2011-03-07 15:53:49.757112 [DEBUG] ozmod_libpri.c:106 -- Restarting T203 timer
2011-03-07 15:53:49.762166 [DEBUG] ozmod_libpri.c:106 
< [ 00 01 01 9f ]
2011-03-07 15:53:49.762166 [DEBUG] ozmod_libpri.c:106 
< Supervisory frame:
2011-03-07 15:53:49.762166 [DEBUG] ozmod_libpri.c:106 < SAPI: 00  C/R: 0 EA: 0
<  TEI: 000        EA: 1
2011-03-07 15:53:49.762166 [DEBUG] ozmod_libpri.c:106 < Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
< N(R): 079 P/F: 1
< 0 bytes of data
2011-03-07 15:53:49.762166 [DEBUG] ozmod_libpri.c:106 Handling message for SAPI/TEI=0/0
2011-03-07 15:53:49.762166 [DEBUG] ozmod_libpri.c:106 -- ACKing all packets from 78 to (but not including) 79
2011-03-07 15:53:49.762166 [DEBUG] ozmod_libpri.c:106 -- Since there was nothing left, stopping T200 counter
2011-03-07 15:53:49.762166 [DEBUG] ozmod_libpri.c:106 -- Stopping T203 counter since we got an ACK
2011-03-07 15:53:49.762166 [DEBUG] ozmod_libpri.c:106 -- Nothing left, starting T203 counter
2011-03-07 15:53:49.762166 [DEBUG] ozmod_libpri.c:106 -- Got RR response to our frame
2011-03-07 15:53:49.762166 [DEBUG] ozmod_libpri.c:106 -- Restarting T203 timer
2011-03-07 15:53:50.220900 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop T4A (13280 remaining)
2011-03-07 15:53:50.220900 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  TSI without final frame tag
2011-03-07 15:53:50.220900 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 43 78 61 66 72 65 70 75 53 20 20 20 20 20 20 20 20 20 20 20 20
2011-03-07 15:53:50.220900 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Remote gave TSI as: "Superfax"
2011-03-07 15:53:50.540609 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  DCS with final frame tag
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 13 83 00 62 f8 44
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx final frame in state 17
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 DCS:
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= 3G mobile network: Not set
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..1.= Receive fax: Set
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..10 00..= Selected data signalling rate: V.17 14400bps
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   0... ....= 2-D coding: Not set
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..00= Recording width: 215mm +- 1%
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 10..= Recording length: Unlimited
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .111 ....= Minimum scan line time: 0ms
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= Frame size: 256 octets
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .1.. ....= T.6 coding: Set
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   0... ....= Extension indicator: Not set
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Selected compression T.6 (3)
2011-03-07 15:53:50.541622 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Get document at 14400bps, modem 7
2011-03-07 15:53:50.542643 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from state 17 to 7
2011-03-07 15:53:50.542643 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Start T2
2011-03-07 15:53:50.580883 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2011-03-07 15:53:50.580883 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_C_NON_ECM_RX
2011-03-07 15:53:50.580883 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set rx type 0
2011-03-07 15:53:50.580883 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set rx type 7
2011-03-07 15:53:50.580883 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set tx type 0
2011-03-07 15:53:50.641294 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2011-03-07 15:53:50.641294 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2011-03-07 15:53:50.761189 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Non-ECM signal status is Training in progress (-3) in state 7
2011-03-07 15:53:51.754816 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_A_CED, state 1
2011-03-07 15:53:51.754816 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Starting answer mode
2011-03-07 15:53:51.754816 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from phase T30_PHASE_A_CED to T30_PHASE_B_TX
2011-03-07 15:53:51.754816 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set rx type 0
2011-03-07 15:53:51.754816 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set tx type 4
2011-03-07 15:53:51.754816 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Start T2
2011-03-07 15:53:51.754816 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from state 1 to 17
2011-03-07 15:53:51.754816 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Sending ident 'SpanDSP Fax Ident'
2011-03-07 15:53:51.754816 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  CSI without final frame tag
2011-03-07 15:53:51.754816 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  ff 03 40 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
2011-03-07 15:53:52.040913 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Non-ECM signal status is Training succeeded (-4) in state 7
2011-03-07 15:53:52.040913 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop T2 (44000 remaining)
2011-03-07 15:53:52.040913 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Switching from V.17 + V.21 to V.17 (-13.79dBm0)
2011-03-07 15:53:53.334614 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 DIS:
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= 3G mobile network: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..1.= Can receive fax: Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= 2-D coding: Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 10..= Recording length: Unlimited
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .1.. ....= T.6 coding: Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Polled sub-address: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= T.43 coding: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...0 ....= Plane interleave: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Selective polling: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Sub-addressing: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Password: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Basic transfer mode (BTM): Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Ready to transfer a character or mixed mode document (polling): Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= Character mode: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Mixed mode (Annex E/T.4): Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Processable mode 26 (Rec. T.505): Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Digital network capability: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Duplex capability: Half only
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= JPEG coding: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...0 ....= Full colour mode: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= 12bits/pel component: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= No subsampling (1:1:1): Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Custom illuminant: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Custom gamut range: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 1...= North American Letter (215.9mm x 279.4mm): Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...1 ....= North American Legal (215.9mm x 355.6mm): Set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   0... ....= Extension indicator: Not set
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  DIS with final frame tag
2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  ff 13 80 00 ee fa c4 80 95 80 80 80 18
2011-03-07 15:53:53.580705 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7
2011-03-07 15:53:53.580705 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Trainability (TCF) test result - 21876 total bits. longest run of zeros was 21601
2011-03-07 15:53:53.580705 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from phase T30_PHASE_C_NON_ECM_RX to T30_PHASE_B_TX
2011-03-07 15:53:53.580705 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set rx type 0
2011-03-07 15:53:53.580705 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set tx type 4
2011-03-07 15:53:53.580705 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from state 7 to 8
2011-03-07 15:53:53.580705 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  CFR with final frame tag
2011-03-07 15:53:53.580705 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  ff 13 84
2011-03-07 15:53:53.794840 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2011-03-07 15:53:53.875238 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2011-03-07 15:53:53.875238 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX
2011-03-07 15:53:53.875238 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set rx type 4
2011-03-07 15:53:53.875238 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set tx type 0
2011-03-07 15:53:53.875238 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Start T4
2011-03-07 15:53:53.895315 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2011-03-07 15:53:53.915402 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2011-03-07 15:53:54.661321 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 8
2011-03-07 15:53:54.740807 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 8
2011-03-07 15:53:54.740807 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from state 8 to 12
2011-03-07 15:53:54.740807 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_C_ECM_RX
2011-03-07 15:53:54.740807 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set rx type 7
2011-03-07 15:53:54.740807 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set tx type 0
2011-03-07 15:53:54.740807 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Start T2
2011-03-07 15:53:54.760881 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier down (-1) in state 12
2011-03-07 15:53:54.760881 [WARNING] mod_spandsp_fax.c:293 WARNING T.30 ECM carrier not found
2011-03-07 15:53:54.920919 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier up (-2) in state 12
2011-03-07 15:53:54.920919 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier up (-2) in state 12
2011-03-07 15:53:55.041038 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Training in progress (-3) in state 12
2011-03-07 15:53:55.061203 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Training succeeded (-4) in state 12
2011-03-07 15:53:55.061203 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Switching from V.17 + V.21 to V.17 (-15.56dBm0)
2011-03-07 15:53:55.081335 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Framing OK (-6) in state 12
2011-03-07 15:53:55.081335 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Start T1A
2011-03-07 15:53:55.420854 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop T1A (277280 remaining)
2011-03-07 15:53:55.420854 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:55.420854 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 00 4f 25 70 ca cd 33 55 21 3c 08 8b 49 98 b4 e6 83 53 0e ce 86 70 37 38 1d ce 4c 3c f3 4c 54 08 0f c2 4c cc 10 66 62 86 b0 98 0f ce 86 f0 20 24 85 b0 cd 33 1d ce 06 a7 c3 d9 e0 cc 33 9b 67 82 51 b6 79 26 1b 02 c2 9d 89 6c c2 ff ff ff ff ff ff ff ff ff ff 7f fa 86 a1 10 7d 30 44 1f 0c 85 42 d3 0c 9a a0 0f 86 e9 0e d3 0d 43 b4 c3 61 28 44 1f 0c d1 07 43 34 a7 0f 86 68 4e 1f 0c 4d 5f 88 3e 18 a2 0f 26 02 02 d1 07 3f 4c 37 0c d3 1d a6 1b 86 e9 0e c1 0f 43 e1 21 fa 60 88 76 18 fa ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 3f 1e e3 0f 85 42 67 ed f1 e1 0f 41 30 1c f3 fd 0f 41 0f 41 0f c5 ff 21 fa e0 ff 1f fe c3 21 da 61 18 86 a0 c3 ff 70 e8 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 6c 94 67 3a 41 d0 50 be 49 98 b4 fe 9f 3e fe ff ff 43 7f be 1f fa 0f c3 f8 3f
2011-03-07 15:53:55.420854 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 0, length 256
2011-03-07 15:53:55.580757 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:55.580757 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:55.580757 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 01 7d fc e9 23 04 c1 f0 1f c3 30 d2 cd 36 db 74 e2 41 d8 a6 d3 c8 43 79 e6 99 76 37 9d a8 f6 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 43 50 da b9 27 37 8c dc 61 08 1a e2 fe 4f 38 82 f0 0c c5 11 fa b4 73 8f 30 42 30 42 09 ef 10 f7 09 3f 84 86 a0 21 ee 61 3c 1d 85 49 f7 08 c5 11 8a 1f 46 e8 b4 87 49 77 18 66 1b a2 0f 86 91 3b 0c c1 ac 0c c3 10 04 c3 6c ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 10 c4 87 8f b3 21 21 38 03 0e 41 ff cf 95 48 26 12 4e fb b4 ff 71 88 3e 18 a2 0f 4e 1f fc a7 0f 86 e8 83 87 a0 9f 84 9f 64 7f 9a a0 7d da a7 7d fc c3 c9 46 f6 7f 1c 82 c3 10 09 08 8c d0 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 3f 04 e1 43 74 1f 11 82 c2 ff 70 f8 e3 30 fc ff ff e1 ff 87 a0 f0 3f 0c ff e3 30 3c a6 2f 44 3b fc 70 18 fe
2011-03-07 15:53:55.580757 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 1, length 256
2011-03-07 15:53:55.741190 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:55.741190 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:55.741190 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 02 4f 2f 71 88 ee ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 43 50 e8 7f be ff ff e7 fb ff 43 ff 7f e8 0f 11 e1 ff 79 ff ff 3f d9 e4 fe ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 9f e7 ff f9 7e be a7 fb 1f 86 21 e8 ff ff ff ff 87 a0 f1 ff a1 ff 43 10 0c f3 fd ff ff 3c ff 43 b4 c3 cf f7 ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff ff 0f 93 1c 86 e0 18 46 88 03 67 c3 89 0a 87 11 42 18 21 18 21 08 86 0e 23 f4 08 23 04 23 04 23 04 23 04 23 04 23 c4 81 0f 23 c4 43 61 42 c0 08 c1 08 41 30 84 a1 43 f0 ff 10 fc 30 42 30 c9 e1 08 c1 08 41 30 04 c3 24 87 21 08 86 23 34 33 fc ff ff ff ff ff ff ff ff ff ff ff ff ff ff 7f 44 44 44 44 44 44 44 44 44 44 44 44 44 44 44 44 44 44 44 44 44 44 44 44 4c b3 31 e8 54 fe ff 10
2011-03-07 15:53:55.741190 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 2, length 256
2011-03-07 15:53:55.881338 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:55.881338 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:55.881338 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 03 0c c1 3f e2 ff ff ff ff ff ff a7 55 b9 d8 33 c4 99 3d 09 91 21 42 44 13 12 91 2b ad 16 61 23 f1 1d 27 2e e2 22 21 08 0a e6 12 12 23 e2 22 f0 cc 22 8a f8 42 0c 89 59 30 dc 48 5c 92 84 0c 21 5b 88 10 c5 10 71 51 2e 08 06 89 f8 a4 0b 12 f1 25 ba 88 8b 40 f7 19 df 6d 11 1f 82 90 79 64 44 5c 74 49 c4 e3 88 2f 48 12 71 91 90 24 aa 05 9b b8 d8 32 f5 10 4a 92 c4 5a 30 3b 1a 8b c2 39 4e 0d a5 a0 14 14 82 c2 a9 85 ab 50 2c 4c 50 9c 63 28 0d 8a 53 bb a2 7d 91 1e 87 0d 3f 2a e3 18 ff 19 8a d7 1e 04 6a 10 15 5f 32 f5 a3 e9 9b 0a aa a4 37 6b b1 85 8e 20 90 12 d7 c4 45 5c cc 62 88 31 12 17 09 81 40 4f 5c bc 25 e2 22 2e b6 48 88 59 7c 11 17 43 0c 41 d0 5c e7 89 3e b3 08 56 88 a1 d9 2a ba b0 b1 29 33 1a 32 15 10 c4 bc 2d 14 5d c4 43 84 21 e2 35 f1 20 19 64 0a a7 70 aa 8d 88
2011-03-07 15:53:55.881338 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 3, length 256
2011-03-07 15:53:56.041188 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:56.041188 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:56.041188 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 04 61 50 4b 25 12 9b fa c4 45 7c 83 60 4a 08 12 04 82 a8 a6 9a c2 29 9c 12 92 98 4d 5c 86 24 e2 a2 4f 31 08 24 49 12 71 49 12 08 a2 cb 14 17 4a 88 08 4b 82 64 1a 22 24 2e 41 74 49 10 84 cc 42 0b 4d 85 20 11 17 6b 11 23 02 c9 30 5b 6c a5 5d c8 94 53 66 54 42 ef 17 85 1b 2d 48 82 88 4b 10 84 10 a8 ab e2 20 10 71 51 88 2d 88 f2 c3 26 2e 08 6a c7 96 5c da d7 b0 64 7d 08 32 71 41 f5 b7 46 1c 82 a2 c7 06 81 26 7e 83 a8 58 c4 45 58 6c 0d 02 49 f1 cd 22 10 08 9b 65 d4 91 42 13 17 71 f1 03 81 cc 9c 0b 9e e5 dd ec a3 84 e6 02 06 a1 2d 5b 22 2e de ec 61 08 32 82 25 11 17 c2 20 83 40 92 4c 35 95 21 09 82 20 34 4a ec 43 82 e8 82 44 3c 89 b8 a4 2e 48 82 24 10 c8 14 41 b9 8d 16 04 a1 11 85 e4 17 0c 41 66 54 96 20 ba f8 81 20 35 25 40 10 a6 2f 41 92 25 81 88 8b b8 0c 22 2e e2
2011-03-07 15:53:56.041188 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 4, length 256
2011-03-07 15:53:56.180851 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:56.180851 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:56.180851 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 05 52 48 8c 21 de d4 43 1c 09 82 10 36 15 68 34 04 53 04 95 29 a7 da f6 9e 09 23 b6 b8 a1 44 d7 0c 11 17 91 45 4a 51 22 72 71 c4 16 47 a4 8b 23 9e b8 59 45 9d 21 b6 90 a7 dc 07 11 17 08 24 6b 6c ab 4c 5c 24 04 09 04 7a 08 5a 12 c2 5f 10 74 d4 2f 6f dc 2b f6 8a bf 04 43 c4 05 33 5c 97 04 51 b8 b6 21 55 3c 0a 5a 6d 99 b8 5a 12 62 ff 02 81 24 24 ae 9b b8 b1 58 44 5c a2 70 18 82 10 25 83 0c 1b 66 8c 71 44 48 20 08 4d 61 8b b8 10 11 11 11 25 e2 13 17 35 04 41 a6 12 09 63 50 4e b9 88 6f 6e 47 63 4f c5 ff 82 85 28 10 5d 92 6c 08 b2 c5 bc 9b c2 41 92 64 13 97 29 5c ec 4d f5 96 04 42 53 1d 82 88 0b 16 71 49 5a 06 11 2f 04 99 3d 45 96 20 ba 24 e2 82 24 08 04 bb 29 87 24 e2 2a 88 0a 88 b8 06 41 74 a1 0b 12 85 83 40 26 ae 10 64 51 0e 22 2e f5 22 b1 25 11 97 d4 46 53 2c 0a
2011-03-07 15:53:56.180851 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 5, length 256
2011-03-07 15:53:56.321514 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:56.321514 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:56.321514 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 06 87 2d 49 1c 87 20 c8 5c f0 f7 fe 91 2d 10 85 43 0e 41 88 48 5c a4 12 55 85 b0 88 cb 90 21 24 3e 5d 20 0a 07 a1 84 08 41 74 21 f6 c6 38 04 13 8f 67 48 10 e5 28 48 82 88 2f 3b 0c 92 24 e2 1b 64 ea 41 10 96 89 ef 92 54 20 81 28 07 11 67 fa 12 f1 25 e2 0a 49 10 da 54 4d 3d 42 c4 8f 46 94 24 48 16 8c 90 44 5c 92 25 59 c2 92 05 89 21 83 ac 86 1c 45 7c a3 51 02 81 4c d5 88 6e 41 92 20 97 8c b0 24 88 b8 d0 76 1b a6 76 e1 86 c5 16 47 ec d9 62 cf 16 47 94 08 82 26 2e e2 13 21 66 11 45 c4 45 5c 64 08 0d 92 9a 48 82 20 24 f2 12 25 22 17 b2 16 47 f4 89 8b f8 10 64 31 04 02 21 ba 44 fc 4d 61 89 b8 40 54 18 69 47 15 8b 8c 10 c4 3d a6 70 ca 15 b2 d0 94 43 11 15 22 2e 08 1a 63 2c 5c 4b be 6f 10 3e 85 17 44 5c d7 18 06 51 0e 11 97 24 95 20 50 41 c4 05 41 63 21 38 08 34 62 69
2011-03-07 15:53:56.321514 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 6, length 256
2011-03-07 15:53:56.481087 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:56.481087 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:56.481087 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 07 e7 f8 1d 25 08 54 25 88 b8 42 6e 2a 42 42 29 85 ab 5a 0c 07 11 57 a7 70 10 5d a6 82 72 08 02 15 71 c5 6c 31 86 a8 80 cd dc e6 fb 11 f7 dd 14 34 34 24 10 68 c4 e9 02 51 38 64 0a 87 18 42 24 7e 99 42 cb 2c 9e 46 41 92 20 0b 12 08 24 11 9f b8 88 0f 49 12 44 97 45 17 22 04 21 04 31 64 2a 77 08 f4 ec 8d 82 20 53 0f 11 17 08 42 89 f8 30 08 04 b2 2c f6 06 4b c4 45 1f a6 70 c1 10 a4 32 55 40 c4 65 94 6c b5 21 48 02 81 24 99 12 dc 45 5c 46 c3 98 31 08 b2 40 22 2e fe 74 81 6c 4a 70 97 44 5c 82 8c 32 25 40 f4 19 92 20 08 0d 12 2c 11 57 0c 81 41 10 71 37 c5 52 b5 20 81 20 44 18 06 49 12 64 49 08 4b 12 a2 a0 ff 23 22 82 40 a6 70 0a 89 72 ca 29 07 49 6c 8d 38 f1 71 c4 51 22 b0 60 8e 0a 4d b0 38 33 04 41 48 17 08 24 d1 85 08 02 a1 05 81 86 29 a7 a8 d4 7b 49 6a 2a 51 45 2c
2011-03-07 15:53:56.481087 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 7, length 256
2011-03-07 15:53:56.621268 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:56.621268 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:56.621268 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 08 10 08 87 20 84 b0 f1 e3 5f 4d 5c c2 dd df 43 d0 2d db 4b c2 a2 f0 f6 a5 72 a8 ff 55 a5 23 1e 47 6f 08 b7 7c 14 08 b4 85 96 a2 28 1f 44 fc 10 14 96 b4 fd 3a 04 8a 88 32 15 fe 30 0c 43 20 cc 5d 89 37 5b c4 c5 13 04 6d 11 97 04 49 42 04 81 24 ba 4c e1 14 4e e1 14 12 c8 10 5d 54 4b a2 27 81 40 12 f1 91 2e 10 08 21 08 4d 81 29 07 a1 24 81 20 43 16 24 e2 11 17 04 21 7d 09 24 41 74 41 92 04 41 10 48 92 25 08 82 4c 25 0a 88 3e 51 96 50 22 ae d6 41 a0 92 25 e2 b2 64 a1 45 e1 14 7e 41 c4 25 48 94 2f 10 e5 32 86 28 b7 20 e2 6a 53 4f e1 6e e2 b6 24 53 85 c2 65 11 17 04 a1 5b 5d b6 5b 45 5c a2 70 10 71 11 17 f1 24 a3 36 85 57 83 20 9b 2a 94 f3 40 5b 46 11 97 05 b5 57 41 92 54 a6 70 0a e7 78 b5 9a bc 6e 6f c6 41 20 89 71 49 12 43 12 04 8d 20 e8 89 8b ea b0 20 c8 54 40 68
2011-03-07 15:53:56.621268 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 8, length 256
2011-03-07 15:53:56.780793 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:56.780793 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:56.780793 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 09 68 e2 c2 be 1c 87 8f 17 19 c2 62 88 b8 24 e2 4b 10 84 12 f1 2c 08 e4 4b 92 20 e4 22 10 84 20 0a 97 25 49 30 64 4e 61 98 5a a0 4d 25 f4 5a a6 ec 34 6d 1f 6e 41 10 2e 57 45 84 88 cb ef e3 0e 0b 44 fc cb 7f 43 a0 a5 23 42 28 10 15 86 a0 df 72 10 e5 3a 85 1f e3 d1 4d 05 88 78 1b 44 fc 1d 71 c9 22 2e 4b 42 ad 37 14 dd 47 a9 57 8d b0 11 a2 70 db d1 20 58 c4 05 41 16 08 24 49 90 10 82 2c 89 f8 10 35 92 24 04 81 24 44 10 d5 10 a2 40 28 99 c2 21 8b b8 50 02 49 22 2e 08 32 64 8a 0e 19 24 49 92 64 48 10 43 c4 65 08 82 0c 81 28 1c 24 41 90 29 1c 42 41 74 09 82 64 41 1a 25 90 20 d9 86 40 b5 65 48 68 8b c2 29 5c c4 05 1b 6a 21 88 0a 10 71 11 57 85 8c cd f1 82 5c 8a b6 9a 42 ec 45 b9 25 c9 5c 94 88 4b 05 41 96 88 4b 12 42 c4 0f fb 20 6a 95 24 10 5d 2d 49 28 88 33 9a c2 45
2011-03-07 15:53:56.780793 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 9, length 256
2011-03-07 15:53:56.921120 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:56.921120 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:56.921120 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 0a 5c 90 21 83 58 c4 4d e1 25 51 ae 12 4a 55 92 88 0b 59 b6 9b 9e e1 20 aa 29 1c b2 24 4b 12 44 97 20 ba 10 91 b0 4c 05 64 d1 93 24 ba 88 8b f8 06 49 c4 27 96 4c e1 14 16 84 16 04 a1 39 99 c2 21 44 c4 46 08 42 08 04 43 08 02 49 96 04 41 8d 90 24 12 32 48 82 20 44 44 4b 02 81 24 08 42 09 82 50 90 29 9c c2 29 b6 fa 63 8e b7 2c 41 c4 c5 bc 84 d1 c4 05 41 20 2a e4 92 d0 06 99 7a 0a a7 f0 82 d4 5c c8 54 06 d1 25 10 f1 d0 94 43 1e 63 59 76 b4 d5 96 20 94 25 c6 a6 1a 46 99 78 f4 05 11 17 71 21 b3 18 b7 44 5c b0 40 54 c0 25 b1 77 08 6d 09 22 be 28 dc 29 47 8e 77 db a2 a8 fd fd 90 0c 32 d4 b2 4b cb 74 11 17 7d 84 a0 1b b1 21 08 d1 46 43 c4 47 09 61 fe 41 14 0e 11 97 84 0c 11 17 71 11 17 5d 88 c4 05 12 04 0d 02 21 22 22 22 a2 24 e3 68 19 54 c4 b5 84 48 54 11 d7 18 36 7a
2011-03-07 15:53:56.921120 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 10, length 256
2011-03-07 15:53:57.061470 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:57.061470 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:57.061470 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 0b c5 f7 36 bc 6a 88 b8 04 1d 1f d7 2a ef 1f 4e e1 10 71 35 43 a6 5c c4 25 0a 57 5d 20 0a 3f 71 f1 eb 5a 10 f1 41 d4 c8 e3 e9 22 ae bf 21 ca fd 83 28 5c 6a 8e 43 e8 71 07 81 86 88 4b fe 61 63 98 f8 d8 20 98 c2 21 10 48 12 24 10 04 49 20 09 04 41 68 48 92 40 14 0e 21 12 17 88 c2 21 ba c8 25 09 82 20 53 38 97 50 0e 09 42 0b e9 92 24 4b 22 59 20 10 42 54 43 88 48 f4 0d 42 10 84 74 11 97 64 99 4a 54 14 ce 91 41 88 12 71 71 d6 6a 5b 25 10 48 16 24 cb 0e 53 81 36 f7 b0 89 0f 97 88 cb 14 0c 41 26 2e a1 18 4b 28 09 92 83 88 4b 92 ac a8 02 81 ba 6d 09 61 88 f8 a8 02 81 50 90 40 20 e2 4b b6 45 5c 08 53 0e d3 65 0b aa 46 17 04 a1 f9 0b 04 42 da 05 43 90 85 2d 95 91 7f c3 e6 28 4a 88 88 88 88 38 04 82 29 a7 16 c8 52 44 b4 d0 89 0b 82 50 2a 09 02 95 29 9c 72 11 97 e4 12 04
2011-03-07 15:53:57.061470 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 11, length 256
2011-03-07 15:53:57.220989 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:57.220989 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:57.220989 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 0c 99 c2 29 48 08 42 ba 06 25 2e 41 10 84 05 02 21 82 40 8d 78 11 d7 f8 17 88 72 d5 88 25 0a f7 0f 41 58 78 9c c2 35 22 43 77 0a 87 88 ab 66 88 b8 30 ea b1 4f e1 c7 5a 10 64 16 11 17 08 a4 45 5c 74 bd 21 e2 7b 34 4a 8d e1 1c 3f 71 c9 4a 97 e2 1e ab ba ea 16 09 ed 6c 7d 6e 38 53 38 67 08 04 32 84 28 09 22 44 20 0a 87 e8 22 17 a2 04 02 19 42 cb 14 2c 89 2e 48 c4 25 08 82 2c 49 c2 41 20 09 11 0d 92 88 cb 10 d2 47 98 d2 59 25 49 a6 70 6e a7 96 7d a3 2c 3b 3f db 73 1c 44 7c 4b 4d 45 8a 0a 2a 13 df 5c f0 a6 1e 05 02 45 49 20 90 29 02 64 41 85 19 97 1c dd a6 84 8c 22 2e cb e8 2e a3 14 2c b6 1a 99 c5 58 10 5d 51 b0 14 d1 bc e2 78 09 a2 9c e3 58 90 29 bc 6d 11 97 4a c4 e5 86 11 a2 dc 38 04 a9 2c 10 48 a2 8b 65 49 11 ed e9 22 3e a2 04 41 28 41 10 1a 42 dc 52 09 e9 82 20
2011-03-07 15:53:57.220989 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 12, length 256
2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17
2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Retry number 1
2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX
2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set rx type 0
2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set tx type 4
2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Sending ident 'SpanDSP Fax Ident'
2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  CSI without final frame tag
2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  ff 03 40 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
2011-03-07 15:53:57.361252 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:57.361252 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:57.361252 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 0d 44 53 99 0a ca 29 a7 28 a8 41 4b 20 8a 32 13 8d 38 ba 89 0b 27 b0 48 17 13 92 48 16 fd 6d 43 10 c3 5f ef 15 af 05 02 49 41 c4 2f 53 cf f1 f0 f7 83 40 b1 46 94 e4 94 db 05 f5 af 1d ad 62 f8 bf a0 1a 22 ae cb 70 0a 2f 09 22 97 52 e8 fa b8 85 92 20 0a 77 11 57 6c e6 b0 6c b3 6f 10 0c 99 62 48 32 84 2e 09 a2 5a 20 0a a7 10 1c 19 24 11 1f 41 20 89 b8 10 04 32 05 0c 41 20 58 c4 05 02 99 42 88 f8 20 90 84 86 0c e1 a2 80 20 a4 8b b8 88 b5 20 6e 29 11 49 f8 b0 40 10 21 83 40 a6 c0 54 50 4d 49 51 2b 6f 4a 4d 05 c7 29 60 aa 43 90 21 ba 8a b8 de d8 5a b2 20 89 c2 61 18 db 92 64 de d8 56 63 10 16 44 5c 92 64 74 84 e9 37 5a a2 70 10 f1 4d 55 aa 4b 90 38 0e 21 6c 34 e5 14 0e 22 2e e2 32 15 14 1e 2d 99 b1 04 02 2d 86 97 34 5a 62 8f 1d 45 5c b1 da 60 db de a1 dd 97 22 24 6c
2011-03-07 15:53:57.361252 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 13, length 256
2011-03-07 15:53:57.521365 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:57.521365 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:57.521365 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 0e 10 68 ba d0 b6 90 b8 88 8b 7c 08 84 d2 53 2d 08 c2 11 82 50 62 0b 04 42 09 82 b6 a4 1e 21 41 10 12 17 9a c2 b9 3a 08 f4 12 12 17 69 c5 10 5d c4 45 45 c4 23 2e e2 96 89 8b b5 f8 9a 25 b6 20 08 87 a0 43 74 71 11 82 42 e9 1a 82 10 04 21 17 11 21 2e 92 e8 62 5e 22 2e c8 c4 45 b9 ba 4c 39 85 53 60 08 41 89 db 20 47 e2 35 04 09 44 dc 7e 10 f1 1b 05 41 10 a6 70 09 22 2e 41 c4 f5 be a0 0f 81 fe 8b c2 41 c4 cf 71 10 71 3d 63 c9 97 6a 10 85 1f a3 20 90 61 29 5a f5 e2 48 b1 c6 21 ca 4d 5c 1e 44 fc 05 b1 55 1f 82 be 25 d9 23 44 05 0c 87 21 0c 82 e8 22 3e 2c 49 e2 8b 21 81 4c e1 10 a2 20 89 2e 35 84 08 41 a6 60 aa 29 1c 42 53 38 84 92 04 09 82 4c e1 14 1d 42 e2 2a 09 82 10 04 92 04 21 4a 08 43 68 90 04 41 16 5d 20 0a 07 09 64 8a 41 39 48 22 2e ba 24 e2 42 48 02 31 96 20
2011-03-07 15:53:57.521365 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 14, length 256
2011-03-07 15:53:57.661125 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:57.661125 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:57.661125 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 0f c8 94 53 b0 d0 22 2e 89 f8 c4 45 5c 20 0a a7 8a 20 19 62 08 82 10 12 71 81 40 12 24 89 b8 04 41 20 58 88 20 c8 14 0e 59 90 24 48 20 0a a7 0a b5 82 24 89 b8 20 10 43 be 7a 88 b8 8e 76 7e 96 a4 02 81 b0 3b 24 88 3b 44 5c ed a2 70 10 f1 8d 71 53 4d bd d5 88 c6 12 64 10 f1 42 10 ca 14 01 32 cf 10 64 37 25 a8 86 88 9b bf 21 ba 18 32 e5 90 20 e2 2a 5b 32 b9 a9 4c b5 a2 45 e1 21 08 c2 06 19 64 8a 70 dc e6 78 41 c4 45 c7 08 c9 29 ea 36 d5 19 22 ae 68 08 51 da b2 88 8b b8 88 8f 28 11 17 42 10 d2 05 41 86 10 41 90 45 5c 20 90 05 02 21 a2 04 41 16 08 64 c8 14 21 c9 32 25 01 49 c4 27 2e 81 18 82 20 4b 82 24 11 17 48 08 41 48 5c 90 88 8b b8 88 8b 18 41 90 05 c9 b2 88 0b 82 e8 43 10 42 20 58 86 0c a1 44 3c 41 12 71 21 2c 49 92 18 81 44 17 04 81 28 1c 24 41 22 3e 2c e2 5b
2011-03-07 15:53:57.661125 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 15, length 256
2011-03-07 15:53:57.801460 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:57.801460 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:57.801460 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 10 10 04 09 25 08 32 84 74 11 17 f1 05 41 92 88 0b 04 82 45 5c 82 2c fa c4 05 02 49 c4 05 0b 92 24 08 92 44 5c bb c6 b2 28 5c 6a d9 6e 13 4f 10 71 ed 46 8c 6a a9 28 1c 22 2e a8 24 95 20 89 b1 41 a6 0a 96 04 11 0f 4b 12 aa b4 85 20 41 86 4d 09 19 6d 2a 28 1c 44 7c 53 82 e3 90 b4 b4 41 c4 87 21 e2 52 34 45 84 41 30 85 9b 38 b6 52 38 88 b8 98 87 4b d6 fe 88 b9 97 9a b8 08 2c 08 42 e2 a3 20 ba 2c 44 be 40 20 49 42 10 68 10 08 25 08 84 6e 10 22 47 10 64 49 0c 0d 81 10 f9 d2 41 90 41 52 df 10 64 90 64 68 84 21 44 ba 20 0a 17 71 0d 41 08 c7 f1 82 40 30 85 43 f4 11 e9 82 40 30 c8 87 20 84 20 83 10 0f 11 57 23 42 10 5a 12 88 0a 59 10 64 0a 96 4e 5c 7b 09 65 0a a7 26 f1 2d 1d 9d b8 a4 92 10 36 71 21 12 97 82 40 3f c6 c4 05 41 bb 05 a2 9a c2 29 5a c3 3c 10 d2 f2 be 5c 38
2011-03-07 15:53:57.801460 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 16, length 256
2011-03-07 15:53:57.961159 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:57.961159 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:57.961159 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 11 ae ce f1 24 54 1f b1 ff 1d 22 ae 55 1a a3 2e e2 ea 41 94 ab 82 88 4f 17 c2 3f 15 3f 3e 96 3e 44 5c c6 ef 0c 21 04 0d 81 d1 37 e8 d8 86 28 dc 3d c2 30 42 20 7a 05 82 40 88 88 10 08 2d e2 02 81 24 e2 d3 85 68 11 79 49 42 08 32 55 73 1c 42 e2 13 97 25 41 90 41 88 48 17 71 a1 20 08 32 45 09 82 21 44 86 38 82 41 08 53 0c 21 a9 43 24 2e 48 c4 45 3c 34 24 41 10 1a 82 20 53 15 85 4b 92 20 19 a2 8b 3c 49 82 04 09 2d 89 b8 88 8b f8 c4 65 49 48 5c c4 05 09 24 49 02 81 24 90 24 49 82 20 44 94 4c c9 a9 a6 b2 24 e2 43 54 43 06 59 92 25 11 17 71 4b c4 45 7c 48 10 64 08 11 21 90 c4 22 08 b2 20 08 11 41 20 49 82 a9 ff dd 25 2b 34 c8 65 41 82 20 2c 99 7a 14 4c e1 05 a1 a9 37 43 96 16 9a 1f c3 43 0c 17 5b 06 59 29 df a6 84 0d 33 0f 82 88 2f 99 72 81 44 5c 6a ca a5 86 88 67 2a
2011-03-07 15:53:57.961159 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 17, length 256
2011-03-07 15:53:58.100620 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:58.101649 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:58.101649 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 12 ef 10 64 08 44 7c 61 49 66 ec 12 73 92 0c 35 85 43 c4 17 64 95 44 5c cc 5e 3f 04 21 5d 20 10 12 17 04 49 88 ae 7c 11 97 12 17 a2 44 7c 10 b4 f2 b5 33 d4 86 36 34 5d c4 c5 55 10 84 08 02 8d d2 12 5d c4 85 08 89 2e 10 48 82 20 53 20 89 0b 11 6b 69 84 20 9c 2e 84 21 44 5c 43 a0 68 10 22 22 a2 4f 5c 92 12 17 a2 5b 88 88 60 35 71 45 11 97 41 d8 52 5d 88 08 4b 05 41 18 11 b1 8a b8 46 18 24 d1 a5 2b a2 45 5c 12 b6 5b 75 08 74 3b 88 6e 41 bf b7 86 88 0f 41 71 5f 20 a8 05 41 78 e2 22 5f 10 71 85 20 84 40 43 72 4b 5f 43 ef e3 bf 2e 94 88 cb 7f 41 c4 55 7a 7c fb a7 f0 d6 15 22 ae 88 5b 1e 11 6c db 30 f1 19 6a 41 42 48 c4 05 a2 5a c4 45 c6 12 4c 15 0a 87 90 c8 82 20 43 82 24 34 25 04 82 4c d1 41 0c 59 90 50 92 24 a4 8f 08 09 25 08 42 44 44 53 12 d1 45 0a 41 10 c2 1c e7
2011-03-07 15:53:58.101649 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 18, length 256
2011-03-07 15:53:58.260635 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:58.261632 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:58.261632 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 13 59 12 08 42 84 20 94 48 88 b8 20 21 5a 20 49 92 64 08 89 c4 92 64 0a 87 50 20 c9 14 ce 11 22 2e 4b 22 be 21 fa 30 c8 92 88 2f 88 1e 5a 88 28 49 c4 57 9e c4 71 87 88 ab 6d b7 5c 90 87 cb 54 77 37 04 61 87 29 c1 43 86 20 54 84 88 6b 2c cb d8 1c 87 88 4b 90 29 bf 20 ba ca 42 49 90 28 1c 44 5c a6 84 d0 d8 54 68 49 92 40 20 89 b8 2c 10 a8 6d e2 bb bb 24 c8 6d 09 a2 70 6c 29 c2 92 29 e2 ab e1 70 34 b5 33 42 a0 22 2e 44 53 21 a4 8b 23 66 a1 29 97 64 8e 53 e8 22 ae 15 21 08 51 25 90 0c 11 97 22 43 cf 2c e2 82 40 8e 93 8b 23 b6 38 62 91 54 78 d1 85 45 5c 96 74 f8 91 b0 e8 da 32 c2 10 d4 74 85 88 0b 4d d5 94 53 a5 84 05 41 96 1b 42 09 82 90 b8 60 d4 82 04 b1 88 f8 96 de 3f 4a 0a 41 6f ca ad 57 bb 74 fc 8f ff dd 29 8f 8f 1f f1 83 28 37 de 74 41 d0 8f 82 fe d1 1a d5 38
2011-03-07 15:53:58.261632 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 19, length 256
2011-03-07 15:53:58.401278 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:58.401278 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:58.401278 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 14 85 4f e1 86 20 6c 24 2e d5 ff 1c cf 1f a2 70 1f 22 ae 6e 53 6d 0c 82 d0 22 2e 62 89 b8 88 8b f8 90 20 11 17 a2 04 02 49 12 71 49 82 90 d8 82 20 cb 10 17 11 17 12 17 08 84 06 31 64 81 a8 03 49 88 68 49 c4 45 2a 49 88 88 10 64 c8 14 59 c4 27 2e 0b 12 48 92 24 88 2e 84 28 58 48 5c 28 49 20 90 84 08 0b 2c c2 4b 12 04 99 e2 02 81 24 e2 92 24 e2 82 24 41 10 5d 96 a8 a6 b8 60 4a 2d 41 74 19 54 c4 05 53 81 c3 6d b9 42 b2 88 2b 88 b8 ca dc 05 35 15 c5 f1 a2 70 49 d8 10 88 b8 6c 10 c8 94 90 24 4c c7 0c 49 f6 ad 92 39 2e 66 19 82 64 99 72 10 f9 b0 4c 09 14 71 b5 a9 e7 b8 84 65 ba cc 5f 68 0a 97 e5 0a 53 78 37 85 0b de 68 63 47 10 84 88 96 20 14 84 88 28 21 22 5d 06 49 10 84 10 84 c4 45 5c 88 92 20 ba 88 8b b8 50 22 2e 34 c7 41 92 44 78 e2 22 2e ba 20 c8 92 10 e9 a2 8b
2011-03-07 15:53:58.401278 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 20, length 256
2011-03-07 15:53:58.540949 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:58.540949 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:58.540949 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 15 59 74 71 44 5c c4 45 17 f1 0d a1 84 b0 24 04 0b 0d a1 44 7c 4b a2 6f 49 c4 16 71 81 20 10 48 22 3e 2c e2 42 4b b2 d0 92 50 22 2e 94 88 4b 20 04 81 24 e2 92 40 10 5a 68 0a a7 a8 dd 86 34 a4 6e 0b a2 0b 92 14 04 42 db 82 40 6d d7 61 0a 6f 0a b7 44 5c 14 97 4c 7c 81 40 96 d0 4d e1 20 e2 c3 e5 ae 20 e2 92 60 4a 58 10 84 32 e5 10 08 36 25 64 88 b8 9d 59 92 21 c8 82 24 e2 a2 6f 13 df c3 c4 e5 f4 91 c4 42 ab c2 94 63 d9 b5 88 4b 94 1b e2 6e 86 42 20 27 2e 62 4b 92 20 08 89 8b b8 e8 62 8b 2e 9e 18 22 2e de 90 88 8b 2e be 10 9f b8 88 0b 21 c8 59 44 97 05 41 8f ac 04 02 59 82 24 49 22 2e e2 17 e2 12 2c d2 0c 41 d8 c4 87 a0 25 e2 02 81 2c d5 82 a4 24 4b 4d 39 85 a1 32 48 32 44 5c b0 0c 2a 35 85 75 e2 ea 0d 11 17 04 72 04 51 61 84 20 53 38 85 31 71 41 90 05 02 c5 a6 70
2011-03-07 15:53:58.540949 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 21, length 256
2011-03-07 15:53:58.700944 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:58.700944 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:58.700944 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 16 0a 13 f1 5b ff 0f 37 04 2d 59 c6 11 22 ae bf 2e 17 42 c4 15 ff 1a 92 a2 91 b8 fa 78 ab 3a 86 a8 a8 52 14 f1 d5 89 0b 3f 88 f8 ca 16 74 08 8a 36 95 e3 42 7b 10 c8 10 f4 0c d5 3d 5e e2 b8 40 94 1f d5 d6 bb 84 48 5c 28 a1 04 02 a1 45 7c 43 28 49 12 88 7a 49 c4 05 41 48 3c e2 02 81 24 09 11 04 21 5a 92 24 49 c4 45 5c 88 92 25 59 68 2a 40 92 29 17 71 c1 02 51 b7 25 09 25 41 82 24 08 02 51 38 44 17 42 20 84 04 41 08 12 71 a1 29 9c 92 8b 42 90 88 cb a2 0b 2c b4 b8 c8 92 2c ba 4c e1 10 b9 e8 22 2e 43 68 aa 82 24 49 22 2e 14 48 32 55 8b f8 c4 85 82 20 88 2e 34 45 1d f7 12 85 73 97 bb 20 e2 12 85 3b 85 4b 92 2c 59 1c 87 88 cb 54 94 24 c9 42 87 a1 94 83 24 0a 09 92 89 8b b8 44 e1 90 44 5c a6 5c b6 20 e2 4b 30 15 92 50 14 0e 41 26 2e e2 99 72 44 a1 24 10 5d 6d 8e 4b 32
2011-03-07 15:53:58.700944 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 22, length 256
2011-03-07 15:53:58.840674 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:58.840674 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:58.841687 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 17 95 05 f1 b0 4d 97 a3 b9 30 58 96 4d 1e 0c a2 cb 05 41 68 34 17 76 fb 7a b3 45 17 71 71 44 d1 0d a2 70 08 25 e2 a2 bf 4f 17 08 64 08 11 21 0a 87 20 dc a6 70 be 50 93 b8 58 e4 c4 65 49 d2 c8 75 10 68 34 c8 d4 02 c2 96 20 10 2c 08 02 45 ba d0 37 e5 90 65 e8 82 a0 68 6c e2 e2 2a b6 e8 a2 6b 0c 41 11 04 52 ec 61 e1 20 ca 23 b9 20 c8 14 b2 54 10 c8 e3 c4 85 23 71 19 24 ad 6d 91 21 53 38 45 34 04 1d 04 32 85 53 6a 41 10 82 40 aa bc 05 22 ee 10 71 6d 08 84 5e 1f 83 88 2b e2 be 1a c2 52 45 77 ab 7c 78 ec e3 08 b7 64 8b c2 25 6c eb cd d7 8f be 21 e2 8a af d8 fc 6b 81 40 25 7c 09 02 51 78 b9 86 20 10 2c c5 6c 15 f3 61 84 11 d2 a6 82 ca 25 1c 4d 09 21 71 11 5f 92 20 c8 14 0e 81 24 19 24 d1 25 19 84 90 10 d1 14 01 21 5a 12 04 21 71 c1 32 b5 0a 04 92 e8 a2 0f 89 78 68 41
2011-03-07 15:53:58.841687 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 23, length 256
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 DIS:
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= 3G mobile network: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..1.= Can receive fax: Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= 2-D coding: Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 10..= Recording length: Unlimited
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .1.. ....= T.6 coding: Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Polled sub-address: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= T.43 coding: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...0 ....= Plane interleave: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Selective polling: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Sub-addressing: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Password: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Basic transfer mode (BTM): Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Ready to transfer a character or mixed mode document (polling): Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= Character mode: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Mixed mode (Annex E/T.4): Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Processable mode 26 (Rec. T.505): Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Digital network capability: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Duplex capability: Half only
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= JPEG coding: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...0 ....= Full colour mode: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= 12bits/pel component: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= No subsampling (1:1:1): Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Custom illuminant: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Custom gamut range: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 1...= North American Letter (215.9mm x 279.4mm): Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...1 ....= North American Legal (215.9mm x 355.6mm): Set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   0... ....= Extension indicator: Not set
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  DIS with final frame tag
2011-03-07 15:53:58.915605 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  ff 13 80 00 ee fa c4 80 95 80 80 80 18
2011-03-07 15:53:59.000633 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:59.000633 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:59.001642 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 18 22 2e 34 48 a2 8b 78 68 c8 92 40 20 c9 92 44 0d 45 86 24 53 b8 20 49 c4 13 c4 90 45 17 c2 bc 86 10 12 24 43 44 16 f1 d1 92 04 d1 65 08 21 49 c4 25 99 d7 90 22 4c 09 11 17 12 17 24 09 04 42 4b 32 48 42 8b 2e 49 50 fd f1 4d 3c b5 d1 14 8e b6 40 54 c8 54 34 36 d5 14 71 41 c4 45 31 4c 3d e5 0e 13 17 71 49 c4 45 3e 5a 2a 0a a7 bc b7 14 6d a8 04 09 65 90 31 4a 02 81 24 11 17 7f e3 16 e5 16 e4 0c 79 c4 06 d1 25 88 b8 18 4b c4 65 90 1d da 8a 36 15 20 e2 ab ee 63 84 84 88 1c b1 05 49 20 fa 10 74 79 58 0f 81 dc 52 ff f4 61 0a a7 10 14 0e 42 89 b8 88 8b 2e 4b 82 20 44 08 42 63 94 8a b8 0a 4e 5c 52 49 c4 15 82 22 22 22 22 22 22 de 58 05 41 e8 77 cf 17 04 aa d9 2a e2 2a 25 2e 92 45 5c 10 74 d5 4b 32 71 d9 12 82 40 68 90 53 2e 09 82 2c d2 45 5c c4 57 58 c4 e5 74 35 71 e1
2011-03-07 15:53:59.001642 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 24, length 256
2011-03-07 15:53:59.141308 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:59.141308 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:59.141308 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 19 68 0a a7 9c 72 ca 29 9c 0a 34 12 17 cc 03 21 5d d8 25 48 56 49 e3 41 c4 7f 66 ba 7d 0b 44 05 7a 2c c6 4d ff b5 2f e2 ca f1 32 62 4b 73 d4 46 ec 85 a6 70 10 71 bd 82 28 fc 2a e3 35 7e a7 f0 9d e1 6f e2 da 11 a2 02 06 81 e8 31 14 44 1d 85 53 38 85 53 38 85 53 38 c7 a9 01 49 c4 25 81 28 1c a2 4b 82 20 44 53 38 24 8a 80 0c 09 32 08 51 02 81 0c 11 59 90 10 12 12 17 08 84 96 24 49 10 84 c4 e7 22 48 f4 50 02 81 2c e2 32 e5 42 84 24 49 a4 ce 22 2e 83 10 04 19 b2 40 90 41 48 58 82 24 44 53 72 0a 87 10 92 c4 10 63 b6 40 90 39 36 d5 92 24 34 85 53 19 32 85 22 53 c8 90 44 5c 20 6a e8 45 90 24 08 32 95 05 89 b8 40 20 49 12 04 41 90 24 49 12 24 81 44 17 c9 02 81 24 49 92 20 c8 02 51 38 05 a5 17 c3 17 71 c1 16 44 5c 2e 08 44 e1 96 24 e2 92 79 49 42 a9 4a da 12 24 51 b8 45
2011-03-07 15:53:59.141308 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 25, length 256
2011-03-07 15:53:59.281007 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:59.281007 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:59.281007 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 1a e1 02 81 d0 54 14 6c 2c a8 ba 89 cb 89 8b 3e c5 96 24 0a 87 e8 92 50 20 49 14 0e d1 25 09 9b bf 10 2d e2 4a e1 8d 52 d9 94 90 da c2 92 b1 0a bb 44 5c ff dd 54 4b 52 44 84 05 c9 82 24 a1 84 48 17 71 21 22 22 22 22 1a 84 c4 45 5c 05 02 49 c4 85 20 90 4a 88 28 08 05 09 24 49 20 10 2e 99 c2 a9 02 21 12 97 25 49 c4 25 11 37 2c 48 10 88 21 41 74 d1 27 2e e2 62 6c 08 25 72 41 92 24 e2 42 83 24 41 20 81 20 08 22 b7 04 12 22 71 41 92 04 41 22 2e 09 12 61 11 17 71 a1 24 99 8c 80 10 25 84 25 59 88 82 24 94 0c 99 12 b2 2c 49 b2 2c e2 13 17 71 d5 1b b7 64 c9 16 ac e8 90 65 ee 20 e2 39 06 11 cf 20 e2 bb a9 a6 1e 41 c4 d5 20 33 86 a4 86 45 75 53 30 e5 22 ae e8 ae 12 88 2e 53 0d 81 60 89 f0 41 c4 45 3c 53 2d 48 ec 2d 41 a6 82 87 0d a2 4b 76 4b 82 29 a7 70 09 72 97 dc d5 ea
2011-03-07 15:53:59.281007 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 26, length 256
2011-03-07 15:53:59.395264 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2011-03-07 15:53:59.441209 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:59.441209 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:59.441209 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 1b 1f a3 24 39 71 11 1f 12 88 6a 08 37 85 43 48 21 08 c2 09 2a b6 10 04 42 44 69 d4 9a 42 20 90 29 0c 34 71 59 12 71 41 50 ef e6 01 e1 5b c4 15 41 c4 45 5c 88 30 08 0d a1 bb bb 44 17 ba cd 22 ec 21 ca 05 51 2e 08 64 07 11 17 08 d4 16 71 41 d0 a5 06 d9 17 71 8d 08 41 88 30 44 5c c4 45 5c 74 41 10 12 d7 50 10 48 7a b9 47 53 0c 10 76 db 8e 38 04 09 76 6f d9 1a a2 f0 41 c4 bb 2b 9e 2d 23 04 1a 82 30 4a 10 b4 4d 5c 5f 6e 87 40 e3 d6 af 10 71 99 12 8e 5b b2 ae 96 be fd 15 65 41 90 83 88 4b 92 1f 43 c4 75 ca bd 4b 3f 08 64 98 3e 18 e2 29 37 bc 24 81 40 12 c2 10 24 e2 0b 02 51 2d 09 0d 92 24 10 64 8e 0d 99 72 be 50 21 08 42 fe 10 24 41 90 f9 22 e2 e2 22 44 94 24 44 18 64 ca 21 08 02 41 a6 a0 24 81 25 49 0c 49 f4 25 41 f4 25 10 48 82 20 94 40 20 89 b8 04 62 c8 94 43 10
2011-03-07 15:53:59.441209 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 27, length 256
2011-03-07 15:53:59.475585 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2011-03-07 15:53:59.475585 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX
2011-03-07 15:53:59.475585 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set rx type 4
2011-03-07 15:53:59.475585 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set tx type 0
2011-03-07 15:53:59.475585 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Start T4
2011-03-07 15:53:59.494751 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2011-03-07 15:53:59.535134 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2011-03-07 15:53:59.581538 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:59.581538 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:59.581538 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 1c 64 41 92 20 49 82 24 89 2e 08 42 41 90 24 b4 04 19 a2 cb 92 cc 30 0c a7 60 49 28 19 24 11 17 eb 25 88 c2 08 04 99 52 40 74 49 c4 63 5e 92 e8 16 04 a2 70 08 44 e1 20 b6 26 ae c5 f0 21 e2 d2 1c 17 6c 19 dd 92 c2 82 d4 92 24 51 b8 24 a3 18 19 e4 02 81 70 14 08 64 4a 94 24 09 a6 70 88 b8 4a 92 5b 96 ca 10 f1 cb 30 9a 0a d3 97 84 92 44 39 6c 4e f0 a0 70 c8 b2 24 e2 92 40 54 c8 94 08 a1 4d b9 20 a9 dc ea 7a c3 54 07 69 c9 d8 20 0a 17 04 b2 2d 10 a5 1e 16 54 4b 28 49 74 a1 8d 4b e4 02 81 24 09 19 22 2e e2 b2 d8 43 10 42 82 39 87 24 2b 6e 53 42 6d 50 f5 88 88 b1 41 d2 85 88 88 88 8e 43 d0 88 31 71 41 a0 a5 26 2e 41 c4 25 08 8a 68 0a 87 4c 39 95 7b 99 82 29 02 d4 85 cc a2 ab 31 42 c4 75 43 50 07 21 22 22 8e 68 84 40 e3 8a 21 50 59 74 b5 30 76 17 71 75 6f b4 28 5c 20
2011-03-07 15:53:59.581538 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 28, length 256
2011-03-07 15:53:59.721210 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:59.721210 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:59.721210 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 1d ca 21 e2 7a 6b 5c aa 68 08 fa f7 22 ae 0a 11 17 08 d4 1a 82 56 87 88 eb a8 0d 22 ae 4d e1 20 e2 2a 10 85 4b 26 9e 28 2f b4 f7 fe 4d 1c e9 1f 77 18 46 88 c2 0d bb 41 8c 50 82 25 88 be 20 08 42 08 a2 0b 0c 99 42 15 4c ad 92 20 08 05 81 04 53 38 b5 72 82 ea 68 0a 87 10 05 d1 25 08 82 4c e1 14 25 e5 20 89 b8 20 10 43 28 81 20 83 10 11 11 a2 e8 92 88 4b 92 04 31 24 49 a6 56 19 32 85 04 92 60 c1 a2 0b 51 82 20 24 2e 14 84 90 c8 97 0c 21 4a c4 05 73 ba 04 09 32 24 35 ff 51 1b b6 ab 79 2b 85 9b 1e 4c 85 24 09 d1 23 04 d9 92 04 09 22 7e 43 32 a8 cb 88 25 e2 da 6e a3 25 94 24 b9 9a 2d 49 20 e2 b7 4c 7c 53 5c c2 32 04 a1 5b 92 b7 0b 92 ca 1c 97 da 74 6c 99 c2 e5 ce b8 88 0b 41 54 94 e9 61 14 88 b8 10 d1 2d e2 43 20 38 71 41 10 28 22 08 84 32 f2 65 2a 0c 41 38 b9 20 10
2011-03-07 15:53:59.721210 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 29, length 256
2011-03-07 15:53:59.763737 [DEBUG] ozmod_libpri.c:106 T203 counter expired, sending RR and scheduling T203 again
2011-03-07 15:53:59.763737 [DEBUG] ozmod_libpri.c:106 Sending Receiver Ready (55)
2011-03-07 15:53:59.764754 [DEBUG] ozmod_libpri.c:106 
> [ 00 01 01 6f ]
2011-03-07 15:53:59.764754 [DEBUG] ozmod_libpri.c:106 
> Supervisory frame:
2011-03-07 15:53:59.764754 [DEBUG] ozmod_libpri.c:106 > SAPI: 00  C/R: 0 EA: 0
>  TEI: 000        EA: 1
2011-03-07 15:53:59.764754 [DEBUG] ozmod_libpri.c:106 > Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
> N(R): 055 P/F: 1
> 0 bytes of data
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 
< [ 02 01 01 9f ]
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 
< Supervisory frame:
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 < SAPI: 00  C/R: 1 EA: 0
<  TEI: 000        EA: 1
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 < Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
< N(R): 079 P/F: 1
< 0 bytes of data
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 Handling message for SAPI/TEI=0/0
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 -- ACKing all packets from 78 to (but not including) 79
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 -- Since there was nothing left, stopping T200 counter
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 -- Stopping T203 counter since we got an ACK
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 -- Nothing left, starting T203 counter
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 -- Unsolicited RR with P/F bit, responding
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 Sending Receiver Ready (55)
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 
> [ 02 01 01 6f ]
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 
> Supervisory frame:
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 > SAPI: 00  C/R: 1 EA: 0
>  TEI: 000        EA: 1
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 > Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
> N(R): 055 P/F: 1
> 0 bytes of data
2011-03-07 15:53:59.769871 [DEBUG] ozmod_libpri.c:106 -- Restarting T203 timer
2011-03-07 15:53:59.774897 [DEBUG] ozmod_libpri.c:106 
< [ 00 01 01 9f ]
2011-03-07 15:53:59.774897 [DEBUG] ozmod_libpri.c:106 
< Supervisory frame:
2011-03-07 15:53:59.774897 [DEBUG] ozmod_libpri.c:106 < SAPI: 00  C/R: 0 EA: 0
<  TEI: 000        EA: 1
2011-03-07 15:53:59.774897 [DEBUG] ozmod_libpri.c:106 < Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
< N(R): 079 P/F: 1
< 0 bytes of data
2011-03-07 15:53:59.774897 [DEBUG] ozmod_libpri.c:106 Handling message for SAPI/TEI=0/0
2011-03-07 15:53:59.774897 [DEBUG] ozmod_libpri.c:106 -- ACKing all packets from 78 to (but not including) 79
2011-03-07 15:53:59.774897 [DEBUG] ozmod_libpri.c:106 -- Since there was nothing left, stopping T200 counter
2011-03-07 15:53:59.774897 [DEBUG] ozmod_libpri.c:106 -- Stopping T203 counter since we got an ACK
2011-03-07 15:53:59.774897 [DEBUG] ozmod_libpri.c:106 -- Nothing left, starting T203 counter
2011-03-07 15:53:59.774897 [DEBUG] ozmod_libpri.c:106 -- Got RR response to our frame
2011-03-07 15:53:59.774897 [DEBUG] ozmod_libpri.c:106 -- Restarting T203 timer
2011-03-07 15:53:59.881137 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:53:59.881137 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:53:59.881137 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 1e a2 20 44 ba d8 a3 1a 32 84 38 3a 71 95 44 5c 20 10 3a 22 5d 16 b6 29 e0 10 84 8f 10 84 0d 41 b8 ca 90 11 11 21 68 e2 72 6f 24 2e 08 72 09 82 de 14 03 42 b7 7d 1b e2 b8 04 11 97 ef 1b b2 20 e8 ab 49 48 14 ae 68 1c 8d 53 78 af 0a 22 5e 8e 43 a0 e3 45 5c 55 30 25 28 d7 6e 2a 20 e2 1a bd 0e 41 18 31 88 b8 ea 9b f8 a2 7c 41 10 b6 29 62 74 6d 1f 5f f5 6c b5 ef b9 61 41 92 88 0b 05 09 62 0b 21 10 0a 92 18 22 92 4c e1 94 9c c2 29 01 19 42 48 92 44 9f c4 12 71 11 17 15 4b 08 81 10 25 ba 88 0b 41 14 0e a1 25 49 8c 25 fa 20 08 25 e2 a1 64 11 1f 92 a9 55 74 09 42 83 60 8a 00 51 2d d2 45 7c e2 92 2c 10 48 a2 4b 42 8a 21 a1 20 08 04 53 72 49 06 49 c4 25 11 4f 92 d4 d1 68 57 41 20 10 f1 0c 11 4f 2d 19 86 25 f6 06 11 cf 62 84 8c c4 5d 96 44 5c 37 b8 20 08 b1 b9 d8 b6 0c 49
2011-03-07 15:53:59.881137 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 30, length 256
2011-03-07 15:54:00.021451 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:00.021451 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:00.021451 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 1f b2 20 49 02 81 30 92 58 10 84 70 35 f1 d9 43 68 98 ca 94 8b b8 64 e2 8a 0d 62 c8 3c 2c 77 8e cb 94 43 c4 65 49 e2 78 15 85 db 14 21 b4 6d c3 ce d0 61 c9 16 f1 25 98 e1 14 4d 83 70 09 41 a0 91 2f 09 21 50 a1 04 41 88 16 71 e1 9c 41 20 24 2e 08 32 64 31 84 48 5c c4 c5 55 21 a8 29 30 05 83 6c 37 04 41 a6 80 51 05 02 21 a2 29 a7 70 0a b9 d4 c4 45 b9 20 c8 14 58 a3 a9 ba 21 9c f8 5e c4 35 71 21 c6 c4 85 8f 16 4a 02 11 77 88 21 e2 49 82 88 cb 4d 05 08 64 e2 12 04 ad d5 92 d1 05 41 38 88 f8 21 e2 0a 11 57 41 22 2e 51 b8 ac 07 51 b8 4e f9 1a d5 b2 cd dd 74 19 14 41 c4 3f 42 c4 f5 5d 06 d1 47 b5 e2 14 3e 8c 82 88 ab 42 47 11 d7 7f bf c7 cf 7c c3 54 47 29 42 12 48 90 20 73 0c 08 89 8b 12 19 24 31 84 10 64 0a a7 16 49 74 a1 e9 96 59 86 10 a6 70 2e 28 17 4a c4 05 a2 1e
2011-03-07 15:54:00.021451 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 31, length 256
2011-03-07 15:54:00.181498 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:00.181498 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:00.181498 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 20 32 25 04 41 28 11 17 29 04 41 88 48 58 16 24 e2 92 88 8b f8 12 04 49 74 09 a2 2f 49 c4 45 24 59 f4 b9 4e 5c 86 2c 09 82 4c 2d b2 24 8b 45 12 4a c4 25 09 a2 4b b2 24 53 85 e3 9c 2e 48 a6 d4 20 89 b8 b8 59 df ef 43 db 10 b4 64 a9 d1 45 5c b6 2c ca 45 5c 14 4b e6 2e 99 f8 4e b1 d5 90 0d ab 04 a2 08 1b 4b 33 b6 44 e1 96 10 55 02 43 c4 a7 87 06 41 36 35 20 e2 42 41 20 98 3d 9a 8a d1 a2 70 d8 92 4d 11 92 77 28 1a 4b ae f2 36 76 07 51 47 21 04 81 60 aa 86 40 20 89 b8 24 e2 92 10 12 08 84 92 84 92 c4 17 88 0f e5 82 a8 a6 32 35 10 5d 86 18 42 8b 2e 10 d5 2a 04 09 25 89 b8 88 8b f8 c4 85 08 cb 12 84 28 49 c4 35 4c 15 49 04 5e 92 04 c1 a2 4b 82 64 41 02 41 a6 e4 92 04 31 64 11 17 24 b4 d0 92 20 c1 10 08 24 21 04 a1 65 6a b3 86 a9 40 08 b4 71 b7 6c 53 38 85 63 41 a0 96
2011-03-07 15:54:00.181498 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 32, length 256
2011-03-07 15:54:00.321151 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:00.321151 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:00.321151 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 21 20 cb 10 84 dd d4 a3 0d 41 26 2e a9 24 e2 2a 59 82 29 21 c6 92 28 9c c2 21 50 73 82 6a ea 6d 39 09 31 98 b8 06 19 29 96 20 88 2e 09 92 cc d8 10 f1 91 59 90 34 85 cb 1c 97 a9 5c 02 51 6e 15 04 ea 14 c9 a6 9c c2 21 e2 f2 90 20 e2 b2 21 0a 77 1f 04 52 97 20 d1 65 ee 7a 08 42 73 1e e6 99 6a 6a 28 22 04 59 c4 27 17 08 b2 10 89 0b 0d 0d 41 88 c4 25 99 22 20 24 2e 09 82 50 02 51 d1 14 4e b9 88 8b b8 18 a2 8b b8 38 22 97 2d 6d ca 29 4e 15 8e 42 e1 22 2e 27 2c e2 32 34 0e 09 82 36 05 1c 9f b8 20 68 6c e2 47 11 d7 6a 08 7b 38 71 15 5b cd 90 45 5c 06 15 71 41 10 12 d7 8d c4 ef 6f 37 f3 e6 9c c2 41 c4 b5 a5 98 c7 14 0e 41 7e e2 c2 26 fe 18 22 2e 49 10 64 97 40 20 6b 43 c4 15 44 7c e2 ba 3b e5 10 f1 a3 a5 92 d0 32 88 b8 36 e5 20 90 a5 14 0e 11 8f ad 2f 48 97 40 c4 b5 31
2011-03-07 15:54:00.321151 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 33, length 256
2011-03-07 15:54:00.460794 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:00.460794 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:00.460794 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 22 7b f4 14 0e a2 bf 21 e2 1a 67 e8 4f 39 44 5c 1e a7 f0 05 53 f8 a3 47 e2 82 a0 11 c4 dd 32 63 2b 22 5a 30 84 28 41 90 45 17 42 20 34 e5 d4 53 48 12 d4 82 40 20 89 b8 2c 73 9c 82 41 12 cb 82 40 20 b4 20 c8 10 4a c4 65 41 90 85 8a f8 60 41 90 a4 88 c4 65 48 22 2e 08 92 50 32 25 0f 53 59 92 44 3c 43 88 88 c4 85 28 90 84 28 d1 47 09 82 24 04 41 88 10 84 90 24 e2 42 cb 42 98 2a 14 0e c1 90 29 66 bd fd 92 4c 7c 49 61 aa 65 49 12 71 59 8c 05 02 49 90 c4 3e 85 43 92 25 11 97 44 5c 96 44 5c 61 a9 04 41 51 42 b4 24 b0 89 4b 91 b1 20 ba 20 09 5e 82 41 06 59 02 41 42 4b 10 f1 5b 92 24 e2 42 88 e3 22 2e 41 2c 4b 20 90 51 11 0b ac 12 71 49 96 21 c8 12 0a 85 90 04 22 2e c9 94 8b b8 24 48 88 fb 9b da 99 7b 11 5d c4 65 43 d0 c4 45 17 24 c9 90 a9 a0 e4 42 a4 8f a4 cb 7a 10 48
2011-03-07 15:54:00.460794 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 34, length 256
2011-03-07 15:54:00.620714 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:00.620714 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:00.620714 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 23 fa 25 64 16 04 a1 39 9c b8 0c a1 24 a1 25 11 17 88 c2 21 44 fa 92 20 e2 1a 19 82 20 73 24 c8 8b b8 4c e1 14 d2 c6 3b 42 c4 35 43 10 84 e4 62 68 89 b8 48 88 ae 1d 82 4c 65 9b 92 27 2e 10 84 c6 c8 45 1a 22 2e 1c e6 64 49 43 90 60 63 b7 24 53 6f 49 d6 22 ae 58 be e5 f6 86 36 e4 78 2b 04 61 21 06 81 26 2c 86 a2 21 50 17 61 1d 1b f5 6f 70 63 d4 46 cc 90 d1 8d 63 10 7d 71 fc da 6f b4 a9 a0 c2 14 02 51 81 e8 8e d8 94 08 02 19 46 30 08 21 08 4b 12 12 17 4a 88 08 8b b8 20 c8 10 42 62 88 23 08 32 e5 54 a1 70 ca a9 c8 11 b9 24 66 49 c4 87 21 94 2c 34 44 1f 4d e1 10 8b 40 42 8b b8 4c e1 d4 0a 59 92 29 9c 3a 6a a8 a6 30 92 24 e2 22 2e 89 42 74 41 90 21 49 10 5b 16 f1 25 a4 8b b0 40 20 09 89 8f 82 18 02 0b 21 08 21 08 89 4b 82 b4 24 19 42 0b 04 92 20 a1 44 17 89 25 84 59
2011-03-07 15:54:00.620714 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 35, length 256
2011-03-07 15:54:00.761430 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:00.761430 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:00.761430 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 24 c5 17 10 5a c4 45 f9 74 21 04 31 64 11 17 71 11 96 a9 2b 86 9f 78 18 ab 65 ca 29 6f 4b 20 ca 21 ab 6c e2 22 ae 92 8c 96 89 0f 37 e5 22 2e db 20 50 34 c5 a5 a2 70 09 6d ae 86 64 6d 88 58 70 5c 48 31 04 d1 55 e2 f8 db e6 6f 2a 50 30 85 37 85 87 db a6 44 88 e1 92 64 b7 a4 c5 f0 ea fa fd 86 a0 95 2e e2 a2 ff 8d 10 08 06 21 8e 12 22 9a 7a 4e 40 8d 10 64 88 b8 8a 2e 94 2c 08 e4 10 84 30 85 53 4f 35 85 53 58 42 10 08 11 65 10 68 53 0e d2 56 ba e8 a3 41 92 84 08 82 fc c2 41 20 dc 6e 53 40 14 b8 41 4a 5c 10 84 33 04 02 49 10 34 c2 14 30 68 47 ba 04 22 2e e2 62 59 47 08 6a 53 4e e1 94 53 38 a8 9a b8 e0 24 54 10 84 74 59 85 e1 e9 e2 4f 57 69 13 17 71 31 a4 db c6 83 29 e2 10 34 ae 6e 99 ca 6e ef cf 7d fc 88 a2 70 c5 d8 1a 87 88 ab 08 8b ad 22 2e ab 2a 88 b8 98 15 cd f8
2011-03-07 15:54:00.761430 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 36, length 256
2011-03-07 15:54:00.901092 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:00.901092 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:00.901092 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 25 18 6d e2 2f ee 2a dc 21 28 8e 21 e8 da 8d 36 e5 57 83 e8 3a f6 0e 81 c2 30 04 0a 77 83 10 11 11 61 6a 95 20 09 82 90 b8 58 04 02 49 10 64 8a 53 4e e1 94 0b 44 e1 10 5d 74 11 17 71 49 88 92 04 02 49 10 84 30 e7 92 20 d1 85 c4 25 a1 44 5c 90 d0 14 0e 21 71 d1 05 96 04 49 12 c4 90 20 08 04 0b 02 c1 14 2c 8b b8 e8 a2 cf 58 10 a2 24 21 f1 89 8b 59 90 04 21 71 49 c4 c5 17 43 20 90 44 5c 20 49 b0 e8 e2 8b b8 24 fa 12 04 21 71 59 90 90 5c 90 e8 32 64 88 1d 5f 20 90 64 49 10 c4 93 20 89 2e 43 92 64 59 10 84 cc 82 20 b4 88 0f 12 f1 4c 09 11 97 39 4e ad 92 88 0b 44 e1 90 64 08 96 04 02 49 10 08 a6 40 e4 86 54 22 2e 55 67 6c 88 b8 a0 92 44 5c 12 42 92 88 cb 54 53 8f 32 77 41 92 18 d9 32 63 cb 3b 56 09 52 2c 9b d7 52 53 c1 71 09 42 58 cc 1b 2d 11 17 0c 11 97 32 2f 89 e3
2011-03-07 15:54:00.901092 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 37, length 256
2011-03-07 15:54:01.061001 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:01.061001 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:01.061001 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 26 90 ab 25 a9 8a e1 92 2c 71 bc 0b 02 a1 d1 55 10 22 88 0a cb f4 b0 44 5c f4 89 8b 64 11 17 7d 44 84 04 41 06 21 43 20 08 2d 90 20 59 92 44 5f 42 94 24 09 82 90 b8 e8 e2 8c c8 97 24 99 ca 94 53 38 64 a1 24 41 a2 0b 0d 31 64 81 40 12 24 49 42 09 25 49 02 81 10 21 08 0d a1 25 08 04 42 a4 4f 5c 20 90 84 20 c8 54 50 b8 20 10 4a c4 25 41 10 a2 41 28 49 92 40 0c 19 b2 88 cb 22 be 21 94 40 20 89 2e 86 24 98 e3 14 87 24 e2 22 2e 10 64 88 2e 08 c4 90 25 49 88 f4 0d 19 42 09 12 24 48 20 0a 87 40 20 49 10 3d 49 22 2e 84 19 ce 0b 84 90 50 90 04 41 28 49 20 90 64 90 64 0a 87 2c 4b 02 41 68 2a fb dc fb bb 20 51 38 6e e2 5b 36 7d 18 82 ac 92 41 c4 45 5c 42 17 71 29 0a b2 0a d1 14 0e 22 2e 0b 4d 11 14 0e 81 a8 90 20 12 41 92 21 09 82 50 a6 08 ee 10 f1 51 b2 84 16 44 fc 32 35
2011-03-07 15:54:01.061001 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 38, length 256
2011-03-07 15:54:01.200661 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:01.200661 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:01.201694 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 27 b0 1b 22 be 04 02 99 bb 0a 44 05 6e 09 44 05 96 a2 98 17 44 5c 9b 7a ca fb 33 f4 ff 8b 21 58 10 84 74 19 a2 4b f2 82 25 fb 74 81 40 96 04 41 8d 17 71 d1 85 32 08 24 19 c2 6e 10 a8 0d 49 12 08 9a b8 1c 1d 4d e1 90 29 1c d4 63 d1 05 02 21 42 a0 82 3a 5d 20 4a 55 20 d0 10 84 05 02 99 c2 a1 76 f6 74 41 10 b2 65 e9 92 55 12 5d 20 10 fa c4 d5 78 9b c2 41 85 43 d0 d8 05 43 58 20 10 4a 12 5d a6 9c 12 02 51 51 42 2c 41 90 0d a1 05 51 2e b4 e8 e2 22 41 50 a7 8b f8 10 84 30 c7 d8 14 e6 df 58 88 86 88 6b 54 63 41 a2 f0 2b 82 40 11 22 ae 28 89 b8 aa 84 3f 62 1f 9d 21 e2 72 9d ab 84 4f e1 d9 9a ae c3 23 71 59 30 85 0b 63 e2 fa d1 ee c2 32 15 55 1c 87 88 ab c7 42 e2 8a 05 11 57 f9 83 28 7f e9 0f f3 68 98 b8 6e a1 29 2c 90 29 b9 24 90 40 14 0e 41 82 04 a2 70 48 82 20 14 48
2011-03-07 15:54:01.201694 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 39, length 256
2011-03-07 15:54:01.360593 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:01.361611 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:01.361611 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 28 42 44 49 42 04 81 d0 82 44 17 47 10 48 42 84 20 ba 60 41 90 04 02 49 f4 25 41 92 84 20 08 11 96 41 68 0a 17 24 b4 04 92 20 49 c4 45 7c e2 32 84 12 42 92 40 92 60 aa a2 1c 92 24 ba 40 20 49 12 44 17 4a c4 65 90 24 41 90 21 04 09 11 a6 70 41 10 24 c9 9c 10 42 90 39 2e 48 12 71 11 97 05 49 82 44 5c 9c 21 c8 82 20 34 85 91 24 08 12 48 92 44 5c 90 20 08 0d 82 25 49 82 90 f8 10 84 12 ab 25 44 49 90 39 ce 1d 24 49 12 ff 70 ed 96 4b 86 20 6c 10 f1 2c d9 54 b8 4c 3d c3 55 e6 04 c8 12 88 b8 61 9b e2 20 f3 07 81 d0 32 f1 14 92 f9 a3 15 92 04 49 f9 d9 12 82 13 9f 1f 54 4a e1 10 71 29 24 49 20 e2 1b b1 6c ee 21 85 8c 5b d5 92 dd b8 d4 54 18 61 4b 4d 39 e5 22 43 3a b6 05 41 88 82 a0 a5 88 38 08 6a e2 22 2e e2 a3 a9 06 4d 17 c2 d0 88 a6 0a c7 41 e3 13 17 71 21 22 22 22 22
2011-03-07 15:54:01.361611 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 40, length 256
2011-03-07 15:54:01.501451 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:01.501451 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:01.501451 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 29 22 22 22 12 57 94 34 22 22 88 72 49 25 23 0a b6 47 c4 d1 42 5f 11 43 a0 23 22 42 a0 22 2e e2 42 44 44 44 44 44 44 1d 11 4d f9 10 22 22 a2 dc 88 88 88 88 88 88 88 7f 43 10 a6 70 9b ca fb 63 53 a7 53 f8 87 b7 24 41 d0 a6 1e 41 c4 05 31 fc 1f 47 95 43 92 e0 b6 a9 93 b5 b9 a0 02 74 59 6e 47 b7 bd f1 29 1c 44 fc cb 3b 88 f8 d7 e8 6b 17 71 b5 7d 47 44 44 84 25 d1 b3 24 41 88 88 88 88 88 88 88 88 88 88 88 86 10 12 24 09 82 04 22 d0 88 88 88 16 04 42 4b 42 44 44 44 34 f5 0e 81 16 12 97 24 41 55 90 10 22 2e 31 5c 90 5c 6a e2 b5 04 99 b8 52 0e bb 05 c9 d8 12 88 b8 24 14 44 5c a6 70 10 71 c1 36 25 5a c4 65 11 17 04 22 9e 20 09 e4 16 85 83 88 2f 6d 49 92 38 0e 22 2e c9 22 ae 2a 73 97 40 c4 8f 26 2e 41 a0 a6 62 11 97 20 08 bb ed 14 97 7a 59 61 bb 31 22 22 22 22 22 22 22
2011-03-07 15:54:01.501451 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 41, length 256
2011-03-07 15:54:01.641419 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:01.641419 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:01.641419 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 2a 22 a2 ed 28 ee 36 17 14 be a4 a3 2d 41 c4 f7 cd 1d 7f dc cb da 41 c4 d7 b8 41 a0 38 62 41 c4 85 8d 28 10 08 37 71 21 71 9d e3 10 71 55 17 88 72 99 0a 34 46 7c 97 0a 6a 81 40 2f 7d bd 23 88 f8 bf 87 88 cb 6f 04 51 dd 06 31 77 5a 7d fd 17 24 41 a0 30 95 41 e6 c1 71 10 71 c9 92 24 11 97 64 10 43 68 2a bb 89 1f 72 94 89 0f 97 24 b8 40 c4 13 85 43 4a 1f 89 07 53 b8 8c 21 99 6a 74 1b 96 4c 11 12 c7 c5 32 6c 5b 20 e2 22 ae 46 f3 00 11 cf 10 84 43 10 ca 72 2f 10 15 58 92 54 6a e2 92 24 a4 78 10 e5 f6 2d b6 36 aa 31 c2 86 aa 24 09 1b 87 2d 17 98 c4 90 c4 71 10 71 19 6d 2e 32 82 51 90 dd dc 1b d5 90 90 62 8b b8 20 2c 8e 1b 25 81 e8 0b 92 89 4b 1a 16 f1 25 10 85 83 88 ab 22 da 99 17 24 84 22 88 f8 86 e8 92 55 b2 44 e1 20 e2 9b 0a 7b 97 d4 36 e3 22 2e dc f0 36 e2 79 0b
2011-03-07 15:54:01.641419 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 42, length 256
2011-03-07 15:54:01.800833 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:01.800833 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:01.800833 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 2b 44 2f c4 10 04 11 bf ee 26 5e 08 44 39 b4 14 44 7f af 0f 22 ae d5 02 51 38 44 5c 29 7f 55 0a 77 fc 98 e3 8d 5a 10 c8 6f 10 28 be 50 8e 46 6b 2a 24 95 44 5c 41 c4 27 ae 26 2e 41 7b a1 c6 eb 1f bf 8f e6 6e 10 d5 30 e3 b5 ef 13 17 18 c3 54 6e c1 08 11 97 40 90 c2 6a 2a 20 08 f2 8d 16 08 64 e2 92 50 22 2e 43 90 41 20 43 2a 99 7a d3 e5 8c b1 8d 05 11 17 63 83 2c 41 c4 37 15 28 e2 32 45 14 e5 56 88 e1 72 09 d4 5c 8c 92 0c 41 38 5a 32 77 2e 04 a2 da c4 85 92 28 57 e5 38 44 97 0b 44 5c 52 ef b8 7d 98 6a 84 20 10 3d 44 51 78 d4 16 08 84 c3 05 a2 5c 6b d7 28 08 34 42 d0 9d 42 a3 c6 86 88 77 43 d0 21 a8 3a 85 40 14 3e 88 f8 96 e2 32 4a fd 07 11 97 97 fe 75 08 42 c7 15 5c fe 3a f3 1e dd 92 60 6d e2 42 41 c4 87 79 c0 20 10 7d ac fd 33 12 a6 27 88 b8 24 21 64 74 11 17 ba
2011-03-07 15:54:01.800833 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 43, length 256
2011-03-07 15:54:01.940756 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:01.940756 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:01.940756 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 2c 4d 3c 89 11 c5 d1 12 71 99 6a 9e 41 c4 25 49 8e 56 98 b8 0c 22 2e e2 92 2c 4b 52 49 62 8b b8 24 58 c4 6d ee 14 0e d1 a5 5b 12 71 c9 c6 6d 10 f1 25 e2 e2 cb 23 48 20 e2 07 11 57 aa 6b 0b 04 92 ba c2 20 e2 a3 51 c4 55 c6 f4 0c 81 28 f7 37 43 1d b7 1d 61 08 f4 af 10 71 c1 6e be 30 42 c4 b5 f3 25 ff 5a b1 4b df a5 af a6 fe 51 e3 10 74 a8 5e 02 51 ee bd 3d 88 f8 75 c9 da 1d 87 24 e2 6a ff f8 17 6a 3c e5 10 28 7c e2 1a 36 dc c6 cb bc 05 22 ee 46 e2 82 a4 96 9b c2 a9 40 18 04 2a 53 c1 83 72 9b 72 99 e3 50 0a 07 11 17 08 64 a1 44 5c 02 31 24 ca 05 02 99 73 ee 22 ae 90 ca 92 29 98 72 c1 20 43 c4 37 45 80 d0 10 5d 92 2d 59 a6 5c 82 e8 2a 41 90 4d 71 08 32 88 82 2d 19 c4 50 ca 15 96 8a b8 a4 42 63 c9 16 85 9b 62 81 88 db 94 c3 92 34 9a 0a 23 6c a9 bf db 61 9b b8 56 09
2011-03-07 15:54:01.940756 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 44, length 256
2011-03-07 15:54:02.100895 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:02.100895 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:02.100895 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 2d 45 e1 20 e2 07 d1 07 81 be ba 88 6b 17 d4 f1 4f a1 2f 10 48 12 2c d5 f8 c3 5e 92 c6 98 b8 1e 04 ea f7 bf 04 77 14 5d 1f d3 45 5c d2 4b cb 60 bf a9 48 42 e2 11 dd 14 ae 32 4a 0a 41 18 27 ae b6 fb 82 b0 d1 86 43 52 89 b8 e0 a6 42 b2 21 c9 fc 04 02 99 1a 49 92 44 5c 86 88 cb 96 24 e2 32 d5 6a 09 44 b5 64 89 b8 1c 52 84 e8 82 20 c8 20 90 29 21 10 68 66 12 48 92 24 4b 32 77 48 92 30 82 18 42 18 26 2e c9 20 e2 b2 60 ea 05 89 b8 24 51 38 44 5c a0 96 9a 22 a8 c9 3c 24 49 20 e2 99 0a 10 71 11 17 c2 20 e2 22 2e c6 96 20 1b c4 c3 a6 76 31 cf 54 1b 44 85 4d 11 92 5a 65 83 e8 09 02 61 41 d6 10 24 11 97 41 96 40 12 71 09 a4 96 44 5c 5a 92 24 49 96 04 53 05 b6 25 c9 32 95 29 1c 02 51 2d 53 04 d5 92 88 4b 90 61 e2 9b 12 fc c4 3c 04 92 24 e2 12 24 11 97 a9 2c c6 92 6c 49 9a
2011-03-07 15:54:02.100895 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 45, length 256
2011-03-07 15:54:02.241564 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:02.241564 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:02.241564 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 2e 72 88 b8 14 55 22 2e 08 44 5c c4 85 30 c8 12 88 2e 49 20 90 24 89 79 08 8d 12 82 a8 70 8f 11 7d 1f eb 4b 78 bb bf f0 25 e2 1a 82 1e 21 50 dd 1d d1 da da 10 74 8d a6 60 8a c0 08 a2 70 10 71 d1 f5 10 74 3c 71 95 ef 10 85 43 c4 f5 32 fe 7d 6f 10 71 1d af cf d0 88 de d2 4b 8d 6d ea a7 70 10 f1 d8 ba 7d 8f 88 88 88 88 e6 55 c9 92 2c 4b 86 24 58 22 2e 44 84 29 1c 42 b7 20 58 c4 2e 48 8c 21 0a 87 e8 a3 05 49 92 2c 81 2c 09 24 09 25 49 22 2e 84 41 90 2c 41 12 22 2c e2 4b a2 62 6b 99 aa 78 40 10 8a b8 f8 4b c4 25 44 0b 82 20 90 d0 08 11 97 64 ea a9 b6 c3 46 b6 72 6b 88 09 5d 88 88 88 88 88 88 88 88 88 88 88 88 88 88 88 88 88 88 88 88 88 88 de 20 90 f6 ad 6c 1d fd 87 88 0b cb f9 2a 6c 08 da 3c 8c b6 14 91 9b 2e d2 c5 10 22 22 22 22 22 a2 65 10 48 12 89 2d e2 32 77 10
2011-03-07 15:54:02.241564 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 46, length 256
2011-03-07 15:54:02.381485 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:02.381485 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:02.381485 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 2f 71 11 af ac ff dc 18 31 44 b2 28 11 f1 b9 88 eb 44 9a 95 08 3c 31 22 85 88 0b 59 09 04 12 8c c4 85 88 88 88 88 88 88 88 88 21 c8 92 88 8b 23 08 42 24 2e e2 9b da 2e e2 42 89 b8 e8 62 08 91 a0 5b 3c 49 34 8b 51 09 a2 d0 24 11 e7 a6 4b e2 32 71 11 7b 44 02 89 b8 2c 48 20 10 12 5b c4 05 53 95 88 6f 61 44 ba 88 8b 12 21 a6 ce b8 a3 73 99 b8 10 21 68 58 88 a8 68 ca fc 28 9c 83 50 34 d4 56 53 b4 94 c9 70 57 02 81 10 91 3d 24 98 bf 50 82 65 49 c4 45 94 f9 22 97 20 94 04 41 50 23 71 91 42 0c 41 02 81 24 86 48 39 69 44 5c a4 55 45 b6 dc 9d b8 24 e2 ea a6 5c 6d 51 2d 27 2e 10 08 d1 9c ce 39 93 50 38 65 01 63 2d 10 68 0d 43 20 27 be 70 13 45 13 ff 89 0f 02 c9 c4 45 4a 11 9f 14 d3 45 4a 11 17 37 59 61 aa 25 49 c4 25 49 16 71 09 62 8b b8 20 88 b1 29 06 64 9e 39 ce 71 2a
2011-03-07 15:54:02.381485 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 47, length 256
2011-03-07 15:54:02.540743 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:02.540743 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:02.540743 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 30 38 0e 99 ca 92 24 e2 22 2c 16 d1 45 5c c4 67 1e 82 40 14 0e d1 c5 11 04 62 88 27 44 e2 a2 cb b0 20 89 b8 24 28 88 c2 21 10 88 21 e2 82 04 49 10 dd 10 15 02 89 b8 20 c8 90 20 ba 88 cc 92 20 ba 88 64 81 d0 22 2e 41 90 88 8b b8 24 89 78 a4 0e 89 ab 34 b3 88 cf 10 71 91 4b b6 7c 24 3e 71 b1 12 71 d1 67 8b 59 74 21 08 64 31 84 10 68 86 88 8b b7 60 16 99 6a fd 10 08 97 e8 17 0a 0d 02 cd d0 78 d1 55 c4 55 20 90 8e 10 74 81 40 12 71 cd 22 e2 62 88 2e 10 a8 bf 47 68 88 b8 ee 21 e8 05 41 e3 10 71 75 13 17 f4 43 20 8c c4 05 51 38 08 b4 0f 41 9b 2d 10 85 5b e3 06 41 ef 74 81 32 74 89 b8 d0 10 48 12 5d 12 4a 30 75 14 0e 49 c4 27 2e ce 92 04 41 74 99 ca 92 40 02 09 24 41 20 90 24 21 67 48 12 4c 71 c8 22 6e 43 66 47 c1 1a 82 d0 10 12 97 85 c4 85 b0 24 44 4b 42 09 51 42 58
2011-03-07 15:54:02.540743 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 48, length 256
2011-03-07 15:54:02.680636 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:02.681646 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:02.681646 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 31 a2 70 08 04 49 c4 17 08 b2 24 09 12 48 92 24 e2 92 24 51 03 49 82 20 c9 12 f1 6d ba 88 9b b8 b0 04 09 02 19 92 24 11 37 71 99 c2 21 ba 04 81 60 e2 12 44 17 2c a1 20 11 97 a8 80 40 02 89 b8 5a bc 25 88 6a 49 c4 25 f1 64 ce 45 5c 10 64 41 12 24 09 42 84 21 84 24 48 12 08 cc 61 21 04 42 98 5a 45 5c 12 71 81 20 44 44 e2 12 48 22 2e 98 0f 84 96 44 5c 90 40 10 9a 2f bc 50 23 ba 90 b8 9c b8 2c 48 82 0c 41 26 2e c3 20 90 24 49 92 25 59 c4 45 5c 61 b0 24 88 bb 20 89 c2 21 28 42 92 20 41 dc 21 ba 60 4b a2 70 08 32 24 09 92 24 48 82 04 09 92 a8 b1 c0 10 44 3f 04 39 fe 18 a3 4a 6e 08 11 4d b5 89 8b 21 f4 dd 39 22 2e a4 0b 89 2f 49 52 17 b9 20 08 7b c4 43 c4 35 16 f3 74 49 c5 ac 42 49 82 20 7c b6 88 0b bd 4b 1a 02 81 22 e2 bf 8d 45 5c 78 8c e8 74 15 22 ea 25 08 e3 8a 63
2011-03-07 15:54:02.681646 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 49, length 256
2011-03-07 15:54:02.840813 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:02.840813 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:02.840813 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 32 b4 89 8b b8 fc 20 e2 0a 41 06 d5 cf bd 0e 53 27 30 d5 1c 0d 32 3f c1 42 e2 a3 79 43 12 5d 96 24 48 22 2e 8b b8 d0 d4 43 68 b6 30 98 1a 59 c4 25 d1 65 5e 44 5c 30 95 45 5c 88 cc 22 2e cb dc 82 95 ec 86 20 8b d8 98 b8 20 14 14 0c 59 c4 25 29 cc 81 24 09 a2 4b 14 0e 12 04 82 25 e2 c1 02 19 0c 0b 66 48 94 0b 36 08 24 5b c4 83 24 08 92 40 0c 59 c4 55 a2 4f 5c 10 84 0d 41 58 9c 29 97 44 17 24 27 2e 85 40 c8 a2 27 2e 43 9b ca 54 a6 1a 6e a1 e9 a2 8b b8 f4 10 71 71 44 5c 21 e2 e2 89 78 78 23 7a 0c 41 68 31 8b b8 78 1a 8b b8 04 d9 43 d0 a7 f0 07 81 4c 17 04 e1 e3 8f ff d8 18 22 ae 0a 22 3e 04 11 97 21 e2 82 a0 fb 76 0c 11 7f c4 55 0f 43 92 44 97 20 53 47 a1 09 4c 9d 50 10 5d 48 7c 14 84 a6 4e 12 24 89 b8 20 08 41 20 98 c2 40 16 c2 d4 53 45 12 5a 92 40 0c 11 97 a9 87
2011-03-07 15:54:02.840813 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 50, length 256
2011-03-07 15:54:02.935468 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17
2011-03-07 15:54:02.935468 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Retry number 2
2011-03-07 15:54:02.935468 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX
2011-03-07 15:54:02.935468 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set rx type 0
2011-03-07 15:54:02.935468 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set tx type 4
2011-03-07 15:54:02.935468 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Sending ident 'SpanDSP Fax Ident'
2011-03-07 15:54:02.935468 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  CSI without final frame tag
2011-03-07 15:54:02.935468 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  ff 03 40 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
2011-03-07 15:54:02.980818 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:02.980818 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:02.980818 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 33 e8 62 8b b8 20 21 12 97 44 5c 88 20 08 91 b8 24 c9 dc 82 95 6a 13 5f 2d e2 32 77 49 25 10 24 89 f8 92 04 81 4a 82 79 b1 45 5c 02 49 92 60 58 60 41 96 24 44 51 38 24 8a 90 20 43 74 49 10 44 17 58 6c d5 7f e2 1a d9 53 ae 22 2e bd 11 02 e1 20 10 a2 a9 38 0e 51 43 e1 d4 88 ab 40 20 44 69 53 5d 25 09 7b 43 88 3b 09 41 10 5a cc 72 09 4d 7d 84 40 97 3a 62 8f be 8f 8f 76 08 4d e5 74 bd 28 1c 22 ae 41 20 c9 11 6f 7c 95 d0 9e 12 71 04 41 a6 dc 21 10 a2 64 21 67 98 92 f3 22 09 4d c1 2c 53 38 f5 fc a8 2c 24 2e 53 23 94 24 89 b8 10 f9 22 2e 89 b8 24 48 a6 70 6a 20 49 12 24 99 1a 21 22 71 49 c4 65 0e 0b 04 92 04 19 82 d0 c4 39 8c 20 5b c4 65 06 64 e2 62 09 04 49 c4 a7 c4 82 20 b9 0d c9 22 2e a4 67 11 97 45 5c 36 71 41 90 94 84 0b 12 98 6f 79 61 93 ab a0 38 f2 35 4f 02 11
2011-03-07 15:54:02.980818 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 51, length 256
2011-03-07 15:54:03.120717 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:03.120717 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:03.120717 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 34 d7 38 04 61 11 17 08 c2 d1 47 c2 23 5a dd 88 e6 81 4d 3d 95 3b 57 11 1f 82 3a 16 c4 a2 89 0b 51 22 ae 28 41 6e 88 b8 46 8f bd a3 82 88 8b b8 50 21 50 c9 21 e2 22 2e dd 41 94 47 08 b2 20 a1 77 44 d1 25 88 b8 4a c4 c5 22 e2 12 24 99 2b a8 22 08 42 49 b2 88 0b 05 21 71 49 12 4c bd 78 0b 42 e2 82 29 4e 8d 10 d9 09 42 09 82 10 82 50 22 2e 43 f4 f9 32 84 74 49 c4 05 41 28 cd 16 71 59 b0 89 2f 48 10 64 41 c4 87 84 52 10 43 86 5a 12 44 5c 16 f1 d1 92 60 43 22 2e 10 48 82 44 5c 61 e2 02 b3 0c 99 c2 a9 13 48 52 8b b8 2c f3 2c 64 8b b8 24 53 99 ca fc a8 9a ca 7c 44 5c 12 4a e6 42 74 41 92 24 24 2e 9e 88 8b 3f 71 21 71 49 7c 49 c4 d5 e6 2d 44 41 20 10 22 b9 24 49 92 e8 42 89 b8 2c e2 13 df 94 c5 51 38 24 08 e6 54 10 f1 0d d1 05 a6 18 e4 0c d1 65 49 c4 33 16 24 11 97 21
2011-03-07 15:54:03.120717 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 52, length 256
2011-03-07 15:54:03.281068 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:03.281068 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:03.281068 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 35 16 96 14 04 81 65 83 24 09 04 92 04 81 24 e2 5b c4 49 20 90 b0 85 88 c4 c5 13 9a c2 50 01 42 2c e2 42 b6 a6 8b f8 88 48 5c 2a 11 57 f1 24 9b 6a 9b 2a 21 08 7f 04 81 14 82 46 10 84 58 9b c2 29 39 c7 41 d3 c5 11 1a 1a 31 0c 02 29 5e 43 10 0a 1d 16 44 5c 52 35 82 40 1b 44 7c 3c 44 fc ab 50 96 88 8b b8 40 dc ef c4 d5 20 a8 43 d0 d6 58 10 71 d1 a5 63 ee 15 4b 14 31 82 04 41 08 12 08 42 41 74 b1 88 b8 24 89 2d e2 42 e2 32 84 48 5c c4 c5 d9 10 5d 16 71 49 c4 65 8e 83 20 21 24 53 47 65 a1 21 49 82 20 ba e8 32 45 29 08 42 14 24 81 25 48 32 24 c1 10 84 10 5d 92 64 11 97 20 e2 bb 21 90 64 43 c4 17 44 97 40 74 59 12 08 82 38 0e 11 17 4a 42 0a 87 e8 82 21 48 92 20 23 64 41 02 0b 86 04 d1 15 11 11 c1 d4 26 40 88 88 88 88 88 88 88 88 88 88 88 88 88 88 68 0a 87 90 16 23 10
2011-03-07 15:54:03.281068 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 53, length 256
2011-03-07 15:54:03.420968 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:03.420968 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:03.420968 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 36 48 28 89 ae b8 3f 44 5c ea 45 97 50 a3 fc 68 94 70 64 0e 41 e8 25 ba 8c 98 b8 e8 82 28 fc 4d 5c c4 05 02 fd 47 94 50 92 19 86 64 09 44 5c c4 65 11 97 24 09 a2 2f 88 2e 58 e0 c1 60 99 c6 41 a2 70 88 b8 88 c2 20 49 12 04 21 69 05 49 12 88 b8 0c 82 2c 41 28 e2 82 20 48 10 84 4d e1 94 45 19 22 7e 10 48 d3 f5 2d 8f 28 f5 10 68 10 48 0b 04 92 e8 0a 41 97 a4 41 94 ff 98 b8 a2 fc c7 14 7e 69 b3 0d b6 6c 08 0b 02 79 89 f8 16 71 81 1d 0c 41 74 c1 b2 0c 85 88 67 41 10 c2 12 2c d8 0c 87 20 cb 21 c9 22 3e 08 92 0c 81 24 48 2a 10 04 81 4a 10 88 b8 20 ca 41 20 49 82 54 92 24 09 04 21 2c 48 c4 25 11 97 44 7c 83 6c 0c 62 16 6c 41 82 2c 49 12 08 92 61 90 04 22 2e c8 92 40 c4 4d 9c 37 8e 5b f8 bf 45 42 76 fb e8 8f 10 84 88 a5 bb 04 a2 22 fc 5d b8 e9 22 2e ba 22 24 4b 32 44 7c
2011-03-07 15:54:03.420968 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 54, length 256
2011-03-07 15:54:03.581096 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:03.581096 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:03.581096 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 37 a3 25 e2 49 c4 07 41 60 0a c8 c4 05 92 44 3c 08 22 2e 48 c4 97 88 67 11 97 45 5c 16 48 88 08 43 41 a2 06 6a 48 11 02 4d 5c 28 13 57 10 f1 25 eb 5b df f7 27 96 4a fa 27 b2 64 0c 41 d0 6d ea df b6 24 49 90 4b 82 88 0b 16 f1 39 82 a0 10 64 41 c4 b7 04 92 88 4b 22 2e 08 8a 22 2e 62 a9 25 49 22 be 45 5c c4 b2 45 7c 41 20 90 41 86 e8 4a 73 6a b1 45 b0 c8 10 f7 88 1b 91 40 74 c5 c4 a5 1b 22 2e e9 1f 04 12 2a d6 43 c4 45 5c c4 25 81 40 e3 5b 0a 41 0d a2 dc f8 7f 10 f1 21 a8 89 cb 10 a6 70 21 36 45 a7 70 99 da cf 9d d2 a2 2a 10 43 b0 88 4b 20 90 24 58 c4 97 24 e2 81 40 12 24 49 08 8b c2 21 fa b0 88 b3 20 c8 2d 49 90 24 c8 32 71 11 b7 20 43 f4 25 08 c4 18 12 71 19 12 a4 92 44 5c a4 dc 90 84 86 e8 8a 74 49 60 0a 67 72 6a c4 1d 0c 89 40 23 22 71 41 58 92 25 41 c4 b7 04
2011-03-07 15:54:03.581096 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 55, length 256
2011-03-07 15:54:03.721162 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:03.721162 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:03.721162 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 38 92 24 41 12 71 c1 54 4b 82 04 49 3c 2c 88 be 20 59 16 e5 82 2d 0a 87 40 b0 04 41 92 20 10 24 48 92 88 1b 04 32 04 41 0d 99 21 ba 04 11 5f 82 44 5c 12 a4 34 a7 16 d2 65 ea 20 ba 46 44 e4 89 08 71 15 a6 70 59 74 c5 f5 10 f4 4b b8 74 88 b8 8c 02 51 be 0b e3 be 3e 15 c2 5f 11 11 3e 04 82 29 a0 3e 04 75 13 d7 13 3f c6 8f 12 5a c8 9d 85 28 71 57 1b 44 5c c4 25 10 24 81 24 4b 92 65 e2 b2 20 4b 60 41 90 24 89 c2 a9 96 24 59 12 24 90 04 41 92 44 35 85 4e 23 43 20 49 20 d9 92 20 58 c4 65 08 32 9d 19 ea 10 84 45 5c ac 45 58 c4 c5 45 e4 b3 88 fe 84 45 5c 84 c5 4a cc 62 c8 54 45 81 a9 42 c1 0c a7 9e bb 88 8b 3d f3 88 38 04 f2 1e f7 d1 2a dc 1d 71 19 8a 1e 25 10 71 a1 df e8 1e 82 6e 05 11 ff b0 a4 1f 22 3e 9e e6 d4 9b 57 54 16 b3 88 0b d1 14 4e 1d 85 43 08 53 38 95 21 c9
2011-03-07 15:54:03.721162 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 56, length 256
2011-03-07 15:54:03.860677 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:03.860677 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:03.861686 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 39 22 2e 53 4f 9d 88 8b b8 2c e2 e2 49 22 21 10 64 8a 53 47 b9 40 90 39 07 49 c4 25 08 92 84 68 41 82 6d 09 84 02 19 6c 10 48 92 41 12 42 92 04 d1 25 10 43 76 34 24 19 42 08 92 20 43 10 08 32 5d 30 db ea 1a 25 e2 62 48 32 75 d4 51 0a ea 41 38 08 84 c4 6f 6c ba 18 42 e2 da 42 b0 88 ab 25 e2 42 94 88 0b 89 0b 82 10 89 ab 20 10 12 17 22 1a 84 a6 86 32 c1 8a e3 df 28 08 64 1b 05 11 57 44 56 92 24 43 28 11 17 04 a1 44 42 3c 21 71 09 42 09 11 96 b9 5c 8e 16 b9 88 0b 91 23 94 d0 42 2e 92 90 b8 4c 1d 65 0a 05 42 34 04 41 b0 d2 5c 49 12 5d 92 04 41 68 99 7a 8a 5b 32 15 4d 0d 85 53 9c 42 08 79 62 6b e7 8c 23 4f c4 85 12 71 f1 44 5c c8 2c c9 94 10 22 22 a6 39 d5 74 41 42 09 04 b2 90 b8 4c 09 09 64 ea 64 aa 40 88 68 aa 02 99 6a 7e 42 53 2b 15 10 17 11 17 22 32 84 88 c8 18
2011-03-07 15:54:03.861686 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 57, length 256
2011-03-07 15:54:04.020694 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:04.020694 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:04.020694 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 3a 82 10 04 92 88 4b e2 6f ea 28 33 68 88 30 b5 4a 82 20 94 18 92 d8 22 2e e4 49 22 2e 09 25 53 38 c7 39 4e 49 04 02 49 88 c4 c5 16 b2 88 b8 10 2d 72 b1 08 f9 e2 8f e6 0d 59 88 68 aa 95 e6 94 22 2e e2 82 24 21 12 97 41 88 88 1c 29 e2 fc 25 e2 1a 11 21 10 e2 bc 11 79 82 20 44 e2 b3 45 5c 12 71 e1 ec 71 8e 24 41 48 17 47 28 b5 10 a6 56 7e d4 88 62 44 ce a6 08 49 a4 8f a6 1a e2 22 94 1c d1 22 17 22 4f 28 c1 54 53 23 e4 22 48 11 0b 82 70 98 1a 0a 93 af 47 72 11 97 d4 7c 21 4f 2d 71 44 5c a6 4a 66 71 1e 21 e1 3c a1 a7 39 e5 cc 9b 1a 6a 44 5c 12 b2 c5 97 24 21 71 99 a2 53 18 ea a8 11 4f 12 71 c1 d4 50 08 21 5d 10 84 88 fc d9 22 2e 64 0b 51 5a 28 21 71 21 12 f7 3c 21 08 74 9b ba c4 c5 90 29 9c c2 b9 9c c2 a9 a7 02 24 49 f4 25 b4 88 4f 5c 12 5d 12 71 71 84 f4 89 6f 68
2011-03-07 15:54:04.020694 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 58, length 256
2011-03-07 15:54:04.160940 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:04.160940 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:04.160940 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 3b be f8 3f 72 64 de d4 51 66 b4 25 e2 62 de d4 89 b8 44 b9 90 f8 08 09 11 d1 20 44 43 68 ea 21 34 d5 82 20 c9 90 64 41 90 65 5e 41 74 09 92 10 11 89 8b f8 88 88 ac 8e 88 88 88 88 88 88 88 88 88 88 88 88 88 28 21 36 45 4d 99 a1 82 24 a5 25 56 95 54 22 2e 53 94 8a 88 f8 ff a7 75 2d 91 9d 6a 12 8a a4 12 75 85 08 9e 34 22 17 91 44 b5 e8 e6 8b f8 c4 05 ed b4 86 25 09 04 32 15 14 41 e1 14 0e 51 88 b8 24 e2 32 c7 a6 20 82 40 c4 45 5c 90 04 41 86 05 41 69 ed 26 1e 71 11 17 5b a6 e8 14 4e 39 a5 cc 9e a2 a6 36 30 b5 55 05 c9 14 21 91 4e 52 0c 89 b8 24 49 20 59 90 24 88 2e 10 64 aa 40 0c a5 55 26 31 54 35 22 8e 5d 12 6b 21 99 c5 5a 4a b2 b8 e5 a3 3d 72 04 02 e9 ac 34 71 35 22 c2 6e c4 30 8e 47 44 18 42 b6 88 0b 11 a6 9e da 54 66 0b 12 04 99 da 40 5a 22 2e 43 12 12 97 2d
2011-03-07 15:54:04.160940 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 59, length 256
2011-03-07 15:54:04.300995 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:04.300995 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:04.300995 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 3c 0a 17 b3 c8 45 11 f1 a4 2d 89 b8 a4 b4 63 44 4d f1 c4 17 71 51 2e e2 22 2e c2 a2 b9 b6 18 42 34 85 53 0b 05 53 15 25 a8 a3 70 aa 50 3c 71 49 2c b3 4c 60 8d f6 48 54 51 88 59 cc e2 2a ba 18 9a b8 6e 9c b8 f8 86 d2 25 88 78 31 08 14 0c 02 4d 8f b8 40 94 23 2c 88 b8 d6 f1 dc ab b8 71 91 54 64 88 2e 16 b1 12 24 16 31 8b 84 48 08 82 10 04 42 86 10 29 04 41 88 08 12 a9 96 90 2f e2 02 41 88 7c 41 22 2e 89 b8 c8 43 53 27 e2 62 0b 82 10 4d 35 f5 10 df 1c 11 17 f1 49 08 92 0b 22 2e 41 04 15 71 31 c4 13 5b 10 84 fc 25 e2 02 81 90 2e 24 3e 04 83 20 b4 14 04 2a cb 10 5d 12 a4 92 04 99 72 41 50 58 20 c8 b2 69 1c b2 10 e6 08 6a c3 dc 5c 12 f1 e9 02 89 21 86 f8 a2 62 04 81 16 2c 11 17 c5 d3 05 81 8a b8 e8 af dd 8c 27 2e 08 9a f8 12 71 d1 67 8b 2e 10 e8 e9 62 0b 44 39 54 88
2011-03-07 15:54:04.300995 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 60, length 256
2011-03-07 15:54:04.460971 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:04.460971 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:04.460971 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 3d 58 1c 49 85 56 d1 c5 11 04 a1 15 53 08 15 04 42 10 d5 c2 9d b8 20 08 91 d4 a2 8b 78 ca 9e 21 10 0e a6 4f 9c 89 8b 31 71 31 cf 98 b8 20 e8 bb 6f e2 12 24 e9 c4 a5 20 10 12 57 ca 85 c4 05 f5 26 2e 53 03 1a 2d 41 4d bf 88 4f 5c 58 20 10 62 23 86 90 f8 90 38 22 2e 08 54 cc d2 12 08 32 04 41 48 c4 10 48 02 81 90 b8 24 e2 42 8a 0c 21 47 96 20 94 e8 92 24 14 64 6a 20 44 44 b4 d0 82 40 68 08 11 51 92 88 8f 88 34 07 0a 89 0b 4d 85 88 4b 22 2e 41 68 2a 53 38 d5 9c ce 71 10 9a e2 22 2e 10 48 42 34 d5 94 53 c8 ac 25 84 1d 06 49 e2 78 81 88 0b 82 c4 5d 92 30 4c b5 88 4b 92 d0 37 24 11 97 6c 88 f8 b2 43 82 64 b5 45 5c 6a 49 90 50 22 2e 44 49 10 71 49 b0 24 65 8c 25 d9 96 45 5c 10 84 18 61 08 11 05 21 4a 16 15 42 09 82 10 d1 42 44 49 22 2e 0a 11 16 04 21 a2 29 21 44 86 08
2011-03-07 15:54:04.460971 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 61, length 256
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 DIS:
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= 3G mobile network: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..1.= Can receive fax: Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= 2-D coding: Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 10..= Recording length: Unlimited
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .1.. ....= T.6 coding: Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Polled sub-address: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= T.43 coding: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...0 ....= Plane interleave: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Selective polling: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Sub-addressing: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Password: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Basic transfer mode (BTM): Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Ready to transfer a character or mixed mode document (polling): Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= Character mode: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Mixed mode (Annex E/T.4): Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= Processable mode 26 (Rec. T.505): Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Digital network capability: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Duplex capability: Half only
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 0...= JPEG coding: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...0 ....= Full colour mode: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= 12bits/pel component: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   1... ....= Extension indicator: Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ...0= No subsampling (1:1:1): Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... ..0.= Custom illuminant: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... .0..= Custom gamut range: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .... 1...= North American Letter (215.9mm x 279.4mm): Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ...1 ....= North American Legal (215.9mm x 355.6mm): Set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30   0... ....= Extension indicator: Not set
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  DIS with final frame tag
2011-03-07 15:54:04.515243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  ff 13 80 00 ee fa c4 80 95 80 80 80 18
2011-03-07 15:54:04.600726 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:04.600726 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:04.600726 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 3e 3a 42 e2 c8 20 cb 42 ce 12 67 8b b8 90 21 c2 c3 a2 2f f1 85 68 d1 45 5c a6 1a 42 cb e6 ad 20 08 d1 e1 70 2c 08 b2 0d 41 82 84 96 54 68 09 64 c8 42 cb 51 b2 dc 22 2e 97 79 1b 6e 57 48 2a 88 b7 12 97 43 50 b7 29 82 6f 78 4a 44 5c a6 82 2a 54 4d 11 47 e2 62 88 2e 72 21 4f a6 1c 1a 91 b8 46 e2 62 91 34 ff 89 8b 10 51 88 ad 69 ce 50 d1 85 92 29 58 c4 45 2e 18 64 aa 25 85 89 0b 89 ab 0c 11 97 a9 cb 11 4a 12 22 f1 1a 21 50 ef 90 96 88 0b d1 ab 04 11 7f ba 54 21 a8 91 ae 21 c8 54 0e 02 29 71 91 6b 1d 1f 06 c3 14 21 63 17 e4 d3 35 a2 42 f2 d7 c8 22 8b b8 88 0b 51 e2 88 b8 46 2a 64 0a 95 0a 08 25 44 44 84 20 34 c5 05 83 10 11 12 08 42 64 6c 6a a1 5a c4 65 ca 29 2e 09 92 04 c9 20 09 82 d0 54 43 92 84 10 84 20 08 91 b0 10 19 49 02 11 97 4b c4 25 09 22 2e 58 8c 5c 0d 72
2011-03-07 15:54:04.600726 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 62, length 256
2011-03-07 15:54:04.761500 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:04.761500 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:04.761500 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 3f e2 12 0c 53 63 49 82 64 c8 12 1a 8a 90 ec 22 2e 95 14 0e a9 88 ab 24 e2 92 88 cb 51 72 e6 6c 41 0a 81 50 c2 dc 8b 9a b8 34 26 8f b0 18 73 34 22 7b 10 08 f9 22 3e 17 11 17 23 69 2e 62 11 71 21 4e 17 71 71 15 71 91 8b 2e e2 a2 10 12 97 34 71 81 40 c8 16 71 b1 12 67 2c e2 2a be 94 45 38 e2 e6 d9 30 c8 94 77 94 88 0b 51 36 0f 0a a7 22 71 15 f2 05 a2 5c 50 fb f7 87 a0 fd 87 31 18 0c a9 20 08 0b 44 b9 55 10 71 a5 ea a5 38 be 2d 87 88 ab 25 1c 82 4c 0d cf 95 ad 62 8b b8 26 ae 9b b0 4a 82 4c ad 20 46 02 31 32 08 11 39 c3 22 6e 43 f4 11 c1 22 2e e2 92 f8 42 8b 5c 0c 19 42 44 44 c9 54 86 10 12 d2 45 5c 16 5d a6 0a 85 53 64 ea 44 5c 8c 4d 3d 65 ae 80 90 bd 29 04 44 17 42 42 41 12 71 59 84 07 53 01 81 40 16 9a 6a ca a9 85 a2 81 89 4b 20 81 88 4b b6 14 b7 88 0b 65 97 0c
2011-03-07 15:54:04.761500 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 63, length 256
2011-03-07 15:54:04.901250 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:04.901250 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:04.901250 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 40 5b 76 cb 28 21 58 b2 79 55 36 ec 24 c4 1e a6 56 59 88 68 1a b7 25 11 17 24 e2 42 c9 20 09 61 d8 9c ae 4b 48 5c 10 84 8e 70 27 2c 97 60 11 17 08 84 48 58 20 10 e2 d3 05 41 cd 13 be 44 5c 94 8b b8 20 a8 11 89 cb 96 d0 10 e2 48 58 3c b5 55 43 10 d2 7c d5 2e 11 17 4f c4 85 48 7c 8e 40 14 2e f3 42 9d 87 84 2a c9 3e 7b e3 b6 bd bd 15 d9 aa db b2 5d 8b b8 20 c8 14 4e e1 54 83 a4 66 c8 e9 32 e5 20 e2 f2 89 4b 05 21 47 c4 45 7c 93 4b 10 15 10 84 20 e2 da 1b 16 d6 c4 37 04 8a d8 d4 08 4d 3d 15 9c 41 85 88 4b 82 64 99 1a 2e 44 5c 88 16 a2 44 1e a2 45 3c 44 44 49 10 08 84 12 71 d1 47 e2 42 44 89 b8 24 44 34 35 b2 20 08 11 82 e8 b2 04 41 10 24 e2 42 44 44 44 44 43 83 a8 90 24 48 22 3e 24 98 6a 11 f7 c3 94 73 1c a2 8b 6e 09 c4 cf 12 24 34 55 04 92 60 e2 12 ac 82 63 a3 c9
2011-03-07 15:54:04.901250 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 64, length 256
2011-03-07 15:54:04.974776 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2011-03-07 15:54:05.041164 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:05.041164 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:05.041164 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 41 85 86 10 11 11 11 11 c9 85 88 88 4d 75 66 49 c4 75 d1 27 2e 2e 42 0b 8b b8 88 0b 82 46 7f 0a 15 5f a6 b8 b8 08 2d 56 22 2e 0a 6d 5e 28 11 61 8a 4c 1d 35 fc 64 3e 54 50 96 c8 3d da 65 5c 4d 8d c7 c7 c2 89 0b 47 44 0b 11 11 11 11 b1 84 68 19 64 48 22 2e e4 2f 11 17 7b 49 92 24 89 2d e2 92 2c f6 28 a1 a9 2c e2 62 16 12 17 61 99 62 c8 d4 43 10 84 12 24 44 34 25 40 88 a6 70 2a 83 60 6a 11 22 22 22 22 46 ce bc 11 82 50 10 22 17 a1 85 88 88 68 99 12 22 2e 98 1a 0e 53 99 b7 10 4d b5 88 cb 54 f3 85 22 43 68 aa 50 04 c7 50 04 75 3c a8 2c b6 4c 39 65 33 73 af aa da 94 84 c2 a9 a6 86 aa 63 4b 25 d5 14 02 09 11 0b c9 37 85 a0 6a 0a a7 88 48 5c 8e a6 84 20 08 69 39 ab 64 ea 29 82 1a 49 a4 0b 51 32 15 14 4e c9 29 98 0a 11 97 29 98 7a 99 c2 a9 42 2d 14 c1 8f 5a 28 9c c2 29
2011-03-07 15:54:05.041164 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 65, length 256
2011-03-07 15:54:05.055254 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2011-03-07 15:54:05.055254 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX
2011-03-07 15:54:05.055254 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set rx type 4
2011-03-07 15:54:05.055254 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set tx type 0
2011-03-07 15:54:05.055254 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Start T4
2011-03-07 15:54:05.075345 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2011-03-07 15:54:05.095429 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2011-03-07 15:54:05.201045 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:05.201045 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:05.201045 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 42 9c 2a 14 82 c2 a9 2c b6 e8 62 08 51 82 20 84 c4 96 c4 16 5d 96 54 42 c6 68 0a 07 29 12 17 9a 6a 08 11 e9 62 0e a5 c6 26 be ea 6c 21 71 11 17 5d 48 9f f8 8e c4 15 71 83 40 32 5d 7c 7d ab 22 71 91 18 24 24 5e e1 86 36 b7 59 4c 65 88 2e 49 12 64 8e 53 2b 75 d4 88 f8 66 0d 92 88 0b 89 5b 22 2e 98 1a 4e 40 82 20 10 c2 22 2e 48 92 04 41 08 43 88 ac 85 88 88 88 88 28 48 20 e2 42 44 44 41 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 d1 94 4d ce 44 48 2a 81 40 52 1c 33 09 61 fc b4 25 51 4c 9b 5c 44 17 13 ba 88 2e d2 88 14 8a 88 88 a7 35 15 71 63 26 3e d5 c2 a6 76 05 a1 a9 0d 53 e8 d4 66 a1 36 4c 25 30 08 34 8e 88 18 91 7b 15 a6 35 12 b1 12 17 11 5b 48 85 48 2b a2 8b 09 5d 44 17 69 44 0a a5 35 25 4d 21 e4 09 19 11 17 b3 90 12 11 9f b8 90 21 e2 13 17 47 c4 85 88
2011-03-07 15:54:05.201045 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 66, length 256
2011-03-07 15:54:05.340916 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:05.340916 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:05.340916 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 43 88 58 32 25 28 c1 09 11 17 d2 97 60 11 17 22 4a 48 5c 47 9e 90 b8 38 82 79 53 30 25 a7 b0 c4 91 84 c4 c5 4d a9 c8 11 22 22 f2 85 12 09 91 90 29 c1 4d c8 c8 99 27 12 42 d6 92 10 67 ab d8 22 2e e2 53 22 a2 8b 09 49 6c 11 5d a4 11 29 94 d6 54 45 5c 68 2a 28 9c 72 10 b2 c5 17 5b 88 84 27 17 a2 a9 42 91 23 8e 78 5a 83 62 89 b8 d0 82 44 17 a2 44 b9 d0 d4 53 41 bd 38 42 44 9c c2 f0 16 22 47 c8 45 88 e3 c4 85 3c 71 bb 88 2e ea b0 22 7f 10 08 11 f9 62 cb 14 4e ad 14 a7 e8 62 0b 71 4c 6b a0 42 64 d6 a6 96 a6 70 2a a8 e1 4d 15 90 84 98 76 8c c8 96 85 e4 62 16 12 17 5d 68 2a a8 0c 59 a6 8e c2 21 84 20 53 cf 25 14 3a 95 90 a8 a6 d0 29 1a 8a 82 da b4 48 17 93 5c 44 13 5b 44 9b db 23 2e 86 90 b8 78 42 44 44 1c 11 11 11 11 11 11 31 b7 5c c4 85 dc 23 ba 58 a6 b6 20 11 97 41
2011-03-07 15:54:05.340916 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 67, length 256
2011-03-07 15:54:05.481392 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:05.481392 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:05.481392 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 44 12 43 c4 05 53 9b 15 64 c8 14 84 3a 0a a7 4c c9 1c 94 32 25 b8 c5 c4 97 4c ed a1 10 d4 c6 a2 82 c2 19 4e 21 53 58 42 08 32 84 b0 20 49 10 84 48 5c 10 04 41 a6 10 92 24 44 26 56 11 5d d4 31 22 b6 38 7a 09 46 c4 2f 53 1b 4f 42 a4 0b 25 08 42 58 4d 0a 21 62 17 71 21 22 92 4a dc 52 42 c4 b0 78 22 2e 6e f5 86 10 06 49 82 e8 82 20 34 85 0d 21 9a 63 41 10 84 54 8b f8 82 2c 9a a2 89 54 a2 cb d4 ae 28 4a 35 35 77 08 84 88 8e c4 a5 13 17 a9 65 87 b9 8a 95 66 11 56 13 97 29 0c 65 2a 28 87 10 91 5b 21 72 91 8b 23 72 91 8b 23 b6 38 22 17 b9 38 62 8b 23 72 21 43 1c b1 c5 11 b9 c8 c5 11 5b 1c 91 8b 23 53 43 71 8a 53 43 d1 29 4e 0d d5 d4 50 9c e2 d4 50 5b 03 04 92 10 25 24 2e 32 49 20 b6 40 20 44 12 42 24 21 16 49 82 e8 e2 4f 42 c4 85 a6 b0 a8 5a c4 05 8b 37 f1 21 f1 05 41
2011-03-07 15:54:05.481392 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 68, length 256
2011-03-07 15:54:05.641380 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:05.641380 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:05.641380 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 45 e6 41 e1 94 b5 de d4 50 38 95 a9 4c 39 85 53 82 e3 14 4e 15 6a a1 70 0a a7 04 95 93 21 8e c8 c5 17 5d 74 11 17 71 d1 a7 0b 04 42 ae 22 2e 16 31 cf 9e b8 c8 25 85 25 a4 8b 1f 5b 0c 31 44 7c e2 62 6c 8e 53 30 e5 54 50 43 e1 d0 c4 9f 23 53 59 6c 41 a0 22 2e b6 40 d0 13 17 7b 8e 88 0b 04 52 e2 22 2e 9e 88 8b b8 e8 22 2e 9e 40 20 24 2e 4b 22 2e a9 4a 06 81 da 47 e6 a3 84 a6 82 c2 45 5c c4 25 41 10 22 88 c2 21 49 42 e2 22 2e 49 92 98 25 11 97 29 08 85 43 48 7c a4 cb d4 48 82 84 06 c1 90 29 08 b5 4a 22 2e e2 82 24 c8 a6 4f 5c 86 4c 11 10 12 17 04 21 44 0d 84 08 49 90 04 53 38 24 59 10 64 ca 25 49 82 24 e2 82 24 09 a2 cb 92 24 84 25 49 82 24 09 d1 b2 24 08 b2 24 09 92 44 5c 90 24 18 32 85 53 38 b5 52 07 42 09 89 4b 50 53 03 1a 0b 11 91 b8 d8 92 f4 89 8b 27 08 6a 08
2011-03-07 15:54:05.641380 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 69, length 256
2011-03-07 15:54:05.781297 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:05.781297 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:05.781297 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 46 b2 70 49 22 3e 04 a1 05 81 50 22 2e 08 c2 b1 88 0b 04 92 20 08 4d 21 20 54 44 dc 54 83 10 71 44 24 2e 89 b8 26 2e 2c 09 21 08 d1 54 05 21 96 84 48 fc 18 51 82 20 44 34 85 53 03 42 c4 89 0b b1 11 51 96 24 98 aa 82 a8 30 24 44 84 04 02 21 22 71 99 5b d5 06 81 24 09 12 04 99 1f 08 21 21 0c 59 12 04 21 c2 fc 85 96 20 53 60 10 22 f1 25 09 89 cb 14 0e 21 a2 21 04 81 4c 3d 35 d4 71 18 e2 8f 92 44 17 5b c4 25 88 f2 21 08 2d 8b 23 e2 b2 24 e2 92 88 0b 92 05 81 10 29 9f b8 2c 49 42 8b 2d e2 42 e2 62 88 b8 10 4d b5 04 49 20 0a 87 20 10 72 44 5c 0c d1 25 88 2e 53 38 25 24 a8 45 7c 43 86 50 10 7f 44 24 2e 53 38 75 bc 21 34 f5 14 98 23 44 5c 92 c4 11 24 84 20 44 44 49 10 7f c9 b2 50 10 08 64 8e 83 90 b8 20 59 c4 65 8a 53 60 ea 20 34 e5 54 50 38 55 43 12 08 42 44 b6 25 10
2011-03-07 15:54:05.781297 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 70, length 256
2011-03-07 15:54:05.941243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:05.941243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:05.941243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 47 48 92 24 8b 27 90 90 b8 24 08 b2 24 24 2e cb 42 94 24 94 a2 2f 88 45 c4 85 83 40 88 28 c8 1c 43 8d 98 45 5c c4 45 17 92 8b b8 bc d1 91 b8 78 22 ac 89 cb fc 22 04 72 fa f3 64 15 62 11 57 ac 20 90 44 8f 2e 94 18 42 fc 11 59 44 5c 0c 05 11 d7 38 b2 88 de a6 66 11 1f 04 42 43 88 88 5a c4 c5 fb b9 4a 22 ae dc 22 83 d0 ec c5 1f 91 2e f6 e6 0d d1 45 5c 14 02 81 24 53 5c 90 88 8b 13 a2 20 24 2e 48 20 08 25 e2 22 3e 71 99 12 42 94 38 1b 32 08 86 10 05 99 c2 29 a7 70 2a 24 21 a2 a9 86 90 eb 86 90 23 e2 42 44 49 02 41 68 99 c2 21 24 2e 14 48 92 88 0b 82 cc 0f 82 44 5c 12 f1 49 88 b8 60 de 12 24 99 82 85 38 04 8a 21 e2 32 b5 52 4b 89 b8 10 91 b8 38 c2 11 82 10 f1 10 09 11 97 a9 83 b0 88 ab 10 11 25 e2 32 85 53 4f 1d 35 25 09 99 35 5d 68 08 11 4d 65 ea f8 81 10 fd 91 27
2011-03-07 15:54:05.941243 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 71, length 256
2011-03-07 15:54:06.081066 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:06.081066 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:06.081066 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 48 d6 92 10 91 b8 2c e4 09 25 e2 b2 20 c8 20 34 08 d1 1f 02 a1 0e 02 19 84 4d 5c 2d 0b 12 5f 10 84 3c 11 5f 82 c5 1b 82 10 89 0b 82 90 b8 4c 41 b8 90 24 d1 65 99 83 25 21 04 a1 39 2e 43 a6 b0 24 11 17 71 41 92 10 89 4b 42 e2 42 e2 82 20 53 e8 14 4e 35 84 10 84 6c 21 71 59 92 84 c4 c5 96 24 10 22 12 17 4c 1d 84 48 5c 88 c4 75 53 a7 1e 51 89 0b 12 71 e1 c4 05 02 21 be 44 5c 12 71 61 a3 29 20 11 97 29 9c 5a 20 5c e2 22 6c 2c 89 b8 78 92 88 cb d4 41 e3 12 e5 22 21 44 a4 6b e2 1a 82 50 d1 54 68 53 74 0a a7 56 ea 2b 12 97 12 17 61 41 10 96 88 0b 92 f9 40 a6 9e c2 a9 11 2c 44 73 42 b0 2c ba 38 22 2e 24 2e 48 fc 60 d1 85 c8 13 71 99 09 4a 88 f2 61 76 d4 49 92 e8 b2 24 44 8e 88 0b 4d ad 10 22 6f e2 02 41 88 88 74 21 71 09 a2 0b 21 08 11 12 7d 09 21 c8 14 ce 71 08 11 04
2011-03-07 15:54:06.081066 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 72, length 256
2011-03-07 15:54:06.220859 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:06.220859 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:06.220859 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 49 42 58 1c 59 c4 15 4d ad 54 c5 5b c4 97 24 08 42 89 42 c4 25 88 2f e2 42 e2 43 10 a2 44 5c 92 44 5c 12 12 17 5a 6c 71 26 2e 0b cc 1f 42 14 44 17 44 35 75 6c 11 04 21 71 49 c4 c5 b7 24 81 40 88 c4 45 78 41 12 4c e1 14 4e 39 d5 14 4e b5 88 4b 22 2e 08 42 49 22 2e 14 44 1f 89 6f c1 22 2e be cc 0b 85 43 88 58 92 58 26 2e 41 c3 14 0e 39 12 17 43 10 34 47 88 13 17 04 21 42 20 98 52 50 41 bd 5a 92 88 0b 82 90 b7 83 20 24 2e 37 85 53 b8 04 99 72 8e f3 a1 e4 11 41 b2 e8 92 88 8f 10 84 c4 45 5c 6f 49 38 5d 6d 78 c5 47 44 58 88 6d 8a db 20 44 98 5a 39 83 b2 c6 0a 41 a6 60 11 17 1a 24 21 24 09 82 0c 59 92 44 17 71 11 1f 11 82 cc 71 ea 44 5c 20 41 90 a9 11 71 41 20 58 92 24 11 97 20 43 20 a1 24 09 92 88 0b 05 09 42 e2 32 05 16 71 19 42 48 68 c1 42 44 44 44 44 44 30 b5 81
2011-03-07 15:54:06.220859 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 73, length 256
2011-03-07 15:54:06.380979 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:06.380979 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:06.380979 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 4a 45 5c 11 11 11 4b 42 44 44 44 1c 11 16 22 22 22 22 22 22 22 22 22 22 22 22 22 9a da 9a 1a 43 42 44 c9 14 4e 6d 32 64 ca 1a 2b 4c d9 41 a8 ad 29 94 90 a6 af 90 24 53 1b 88 32 cd 0a 53 76 1a 71 93 04 41 08 12 48 c4 15 12 22 22 88 e3 31 4c 59 7d 48 a0 48 ab b8 88 0b 82 c6 92 64 a5 35 55 e9 18 04 92 04 49 69 a6 16 08 a4 23 d2 5e 89 12 45 b2 61 89 2e ba d8 22 88 20 4b b4 af 22 2e e2 6a ba 88 8b b8 18 c2 34 97 12 5f 74 d1 c5 16 d9 18 83 40 12 24 09 04 99 a2 82 4c 35 84 54 15 19 82 24 49 88 88 12 e9 22 2e e2 22 90 72 2b 66 8b 7a 11 47 74 31 31 f4 88 48 17 77 11 91 45 26 71 73 11 17 57 11 2c 44 44 44 24 2e ae a2 8b 8b a0 23 7b fa c4 45 21 ba 40 20 89 b8 18 23 43 dc 27 62 44 14 15 22 b2 12 08 84 16 43 ca 16 71 e5 2e 62 8b b8 e8 2a 65 62 11 7d e2 4b f4 2c e2 23 99 44
2011-03-07 15:54:06.380979 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 74, length 256
2011-03-07 15:54:06.521058 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:06.521058 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:06.521058 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 4b 50 91 10 a2 29 3a a5 42 61 89 78 90 08 22 72 11 df 22 2e 89 2e 10 08 a9 29 34 84 12 88 b5 88 8b b8 20 10 c2 42 53 4e d1 25 41 10 11 a2 0b 04 92 e8 53 43 c4 05 09 04 92 b8 88 b8 20 08 25 08 42 04 81 24 49 10 62 44 44 74 6e 0d 21 22 22 22 e5 a2 cf 34 b5 04 41 c8 7a c2 22 42 c4 25 b1 16 04 4d 5c 08 a2 22 98 d2 8c a3 d5 38 71 4d ba 90 f8 10 f4 82 a0 88 13 1f 82 90 59 c4 45 5c 86 46 44 e2 a2 0b 1f 11 51 10 5d 9b 42 8a 08 69 44 fe 91 95 88 8b b8 88 cb 94 53 3b 28 2a 88 3d 37 d3 02 31 44 5c 88 c4 85 82 e8 42 ba 10 a9 2d 53 b5 10 89 8b 20 22 2e 09 12 24 e2 23 41 84 c4 85 74 a1 84 88 10 44 9f 2d 48 92 40 88 c4 85 88 96 84 48 84 e8 42 53 74 4a 8a 2a 42 c9 94 26 45 a7 a4 21 0a 91 eb e3 c4 67 2d ee 17 24 25 2e 5c 10 04 72 84 20 c4 99 45 d7 12 b2 16 71 21 82 28 17 e2 86
2011-03-07 15:54:06.521058 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 75, length 256
2011-03-07 15:54:06.661446 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:06.661446 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:06.661446 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 4c 10 11 04 21 08 84 d3 c5 5d 49 10 04 42 d0 6d 08 42 9e 88 cb 1a 25 24 95 24 e2 22 2e 97 24 44 22 89 59 24 14 46 e2 42 2e 8a 08 41 86 18 32 b5 1f 82 40 b0 88 8b 1b 22 48 a6 36 9d c2 29 74 21 71 11 17 37 64 44 b4 88 0b d1 82 20 14 c4 cd 10 47 84 45 5c 30 a5 01 21 1a 42 10 1f 08 02 21 71 59 12 9a c2 50 47 15 8a 53 38 c5 41 92 25 81 40 12 4b 12 f1 e9 a2 10 b3 90 b8 98 45 17 09 71 f3 41 20 49 a2 07 41 a6 8e 42 12 5d b9 81 c5 4a 12 42 10 4a c4 45 64 09 42 44 6e 56 21 9a 6d 8c 20 59 30 25 28 9c 7a 2a 53 5c 48 17 b1 a6 44 82 08 16 2b 91 90 44 5c 12 5d a4 96 84 16 71 21 e5 62 88 2e d2 97 88 8b 59 74 11 17 b3 88 2c 09 44 e1 10 9d 21 88 2e e2 02 81 24 08 42 48 c4 12 71 21 24 34 35 54 8b b8 24 89 2e 44 8b 27 88 5a 04 41 a6 4e f4 11 25 16 41 90 a9 5d 47 e2 42 81 0c 21 f2
2011-03-07 15:54:06.661446 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 76, length 256
2011-03-07 15:54:06.820703 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:06.820703 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:06.820703 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 4d 99 2f ba 88 0b 24 29 f7 0b 47 24 2e e2 42 44 e2 92 88 0b 49 71 08 59 44 58 85 2c 22 44 5c 45 58 74 d1 27 17 5d 6f 3e 25 86 82 91 b8 f8 a2 8b 27 e2 a2 6b 66 31 c4 22 94 88 8f c4 67 11 12 17 04 6d 91 3f 12 16 63 72 f1 04 41 c8 10 71 b1 88 23 8e 88 8b b8 88 27 b1 6d f1 44 17 67 ba 88 cb a0 c3 5c 83 a9 3e 04 a1 24 49 f4 89 10 12 a2 df a0 82 84 68 2a 53 38 25 44 5c 16 8b 88 0f 53 c3 71 ea a0 25 e2 52 c4 36 85 53 2b f5 d4 27 2e ba 20 90 b3 b5 24 41 a0 89 0b 82 1a 11 1d 82 10 4d 11 94 53 11 11 21 4a 87 11 89 0b 82 10 21 08 25 09 4d 81 29 4e 09 ca a9 4c 39 c7 29 41 0d e5 d4 41 c8 2c 94 88 8b 8b 90 b8 98 45 9f 2d e4 c9 d4 50 2f 44 53 2e 14 24 49 12 4a c4 45 5c 96 64 11 97 20 09 12 71 21 5d 86 40 20 44 e2 4b c4 25 a1 44 7c b6 20 88 2e 89 b8 10 02 15 08 42 e2 42 08 42
2011-03-07 15:54:06.820703 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 77, length 256
2011-03-07 15:54:06.961621 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:06.961621 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:06.961621 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 4e e2 b2 20 41 10 4a 92 24 10 63 09 e2 38 64 aa 50 0e 82 25 41 90 21 83 58 44 97 40 20 90 44 5c 96 25 11 97 84 92 04 41 96 42 42 a4 4b 82 20 8b b8 2c 89 b8 46 09 96 24 49 06 c1 1c a7 86 c2 21 49 92 88 4b 02 81 24 94 20 08 25 08 32 f5 54 4b 42 53 cd 5f 30 85 f3 a6 50 28 2a 2a 0b 49 08 82 4c 05 95 21 44 16 99 c2 29 9d 73 03 0b 91 b8 88 8b 59 e4 a2 8b 59 1c d1 85 c4 e7 4f 17 71 8d c4 88 45 1c 31 26 3e b1 44 b0 b8 0a 39 a2 8b b8 a0 c4 65 0a a7 0e 32 d4 4b 4a 3c 73 5c 10 84 20 ca 13 57 13 17 71 c5 a6 1c 32 95 41 45 5c 12 71 11 5f 82 20 94 88 0b 32 7d ba 38 22 2e 83 50 32 95 45 5c c4 e7 88 b8 14 51 25 10 15 4d 35 7f c3 79 02 81 90 d8 22 5d 12 71 b1 12 04 42 e2 a2 8b 2d 10 75 91 b8 b0 46 74 34 e5 54 0b 82 10 82 10 25 ba 70 47 14 44 5c 90 60 21 8e a6 5e c8 d6 a8 88 88
2011-03-07 15:54:06.961621 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 78, length 256
2011-03-07 15:54:07.120981 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:07.120981 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:07.120981 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 4f 21 44 e2 9a b8 20 68 12 32 95 41 2b ce d0 a6 72 53 38 45 25 24 be a9 4c fd 96 10 47 34 b5 dc 14 41 e1 54 84 20 44 49 6a 74 09 4b 92 08 11 22 71 8d d2 c4 85 20 90 8e c4 15 29 17 22 08 64 2a a8 96 04 09 82 2c e2 02 81 60 0a a7 5e 92 29 9c 5a 28 17 71 21 08 24 09 12 44 17 a2 a9 cc 47 12 71 41 20 89 b7 24 a1 24 88 be 24 48 10 42 90 20 c9 90 44 97 20 49 12 48 42 89 b8 24 49 b2 04 49 74 49 74 49 12 a2 a9 91 44 5c e6 23 10 45 67 99 1a a1 24 48 10 22 4a 68 59 28 59 c4 05 53 0d a1 44 5c 48 5c 82 10 0d 21 12 97 c5 0b e9 12 44 97 85 48 8c d8 4b 48 5c 10 64 ea 24 11 17 4c 71 ea 05 41 a6 8e 42 08 25 08 82 24 08 11 82 50 82 21 44 e2 62 8b b8 24 e2 42 48 12 04 21 22 47 c4 85 10 84 16 08 24 71 44 5c 88 88 08 41 28 88 2e 84 44 5c 92 29 a7 48 45 c4 10 84 20 90 a9 83 10 91 2d
2011-03-07 15:54:07.120981 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 79, length 256
2011-03-07 15:54:07.260842 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:07.260842 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:07.260842 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 50 e2 e2 cb 90 41 48 57 6e 60 41 10 d2 c5 10 71 8d 6d 68 41 74 21 2c e2 92 20 08 11 82 d0 14 4e a1 e6 3f a2 84 4d 5c c4 05 41 88 58 20 10 3a 22 a2 20 e2 b2 51 10 4a c4 55 88 88 d8 12 22 08 d4 88 e3 a3 45 5c 1c f1 24 49 88 65 6a 12 17 67 b6 e8 82 43 10 42 90 29 41 9d 88 8b 23 9e 88 0b 91 2f e2 41 22 2e 53 01 59 20 90 54 2e 85 59 89 2e e2 1a 82 2e e2 0a 41 10 1a 42 89 b8 38 e2 4b 41 94 0b 89 4f 5c c4 85 28 09 a2 cb a2 4b 9a 2e 41 74 49 12 08 24 d1 35 09 35 bd d1 14 4e 35 85 53 4e f5 11 04 32 15 91 d5 74 d1 65 51 08 a6 32 f5 54 73 9c 82 59 58 22 2e 0b 82 46 c6 ac 64 0a a7 6a 0a a7 5a a6 86 9a 48 5c 88 48 5c 3c b5 24 d1 c5 19 91 b8 88 37 08 34 82 20 8c 08 02 21 04 21 82 40 88 68 aa 05 c9 82 25 49 c8 45 48 57 21 04 21 91 45 b0 04 49 92 20 ba 28 44 5c 92 64 aa f0 93
2011-03-07 15:54:07.260842 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 80, length 256
2011-03-07 15:54:07.401204 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:07.401204 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:07.401204 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 51 40 cc 22 2e 89 b8 2c 48 3c d1 c5 10 5d 45 5c c8 13 71 59 12 08 c2 e9 92 24 44 94 88 8b b7 79 40 a6 42 88 88 92 44 97 25 d1 25 49 10 84 88 dc 79 49 20 58 12 4a 88 92 20 53 38 25 3f a2 84 68 11 97 44 5c 92 44 5c 36 9a a2 8b b8 20 10 2c 89 2e e2 82 20 53 27 43 28 11 17 a2 45 5c 90 24 b4 88 4b 1a c1 10 a2 85 a6 b2 50 42 e2 12 04 09 92 44 5c 48 5c 92 04 12 22 47 c4 05 86 e8 b2 88 0b 11 89 0b 12 71 71 44 5c 28 21 43 88 06 49 c4 87 84 20 90 c4 cd 05 41 68 0a 87 4c e1 54 43 28 08 82 0c 42 34 a5 d3 21 11 97 f9 f3 4c 1d 84 88 e6 ab 4c ad 22 2e 10 48 32 cf 14 4e ed 12 0d a1 24 a1 79 86 10 a6 9e 6a 76 22 2e 44 14 64 59 88 96 25 c1 d4 0b 91 23 e2 e2 89 b8 24 44 18 24 41 10 9a c2 b9 53 38 85 53 cd 33 75 12 64 59 10 84 f4 d9 13 17 4c 9d 18 82 20 44 ba 24 66 11 97 a9 4c 9d
2011-03-07 15:54:07.401204 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 81, length 256
2011-03-07 15:54:07.561240 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:07.561240 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:07.561240 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 52 24 ba 20 08 4d e1 14 ce 9f 7a 0a a7 08 10 22 12 17 9a ca d9 23 2b 81 40 68 ea 84 92 64 08 51 22 2e 09 c9 85 86 24 41 68 8e cc 70 91 0b e9 32 55 a8 45 10 64 8a 1a 24 49 48 5c 16 b3 2c 10 c5 a0 8a 04 42 8b 2f e2 42 e2 32 27 a8 17 a2 45 5c 12 71 99 2a b2 24 e2 42 48 10 84 10 64 4a 50 7a 2e 89 f8 88 13 17 12 17 5b c4 05 a2 70 88 45 30 d5 62 c8 7a 54 73 2c 8a 02 22 2e 10 08 99 05 41 2e 88 3e 12 17 71 49 c4 45 5f b2 64 b6 58 64 8e 53 01 1a 59 cd 2a d5 55 10 4f c4 d5 23 d2 c5 13 08 24 09 ab 3f 16 04 2d 11 97 29 9c e2 92 40 a0 41 14 4c 61 53 74 5e 51 9c 3a ea 40 12 12 d7 28 d1 25 a1 a1 27 2e c2 c2 61 aa 21 44 9c b8 e8 42 10 e8 61 ea 28 b0 10 51 22 2e fe 88 a6 70 8a 27 2e 8e 20 08 11 89 4f d7 64 58 10 14 11 04 92 24 08 42 98 fa 04 16 22 12 1f 87 20 34 85 0d a1 84 32
2011-03-07 15:54:07.561240 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 82, length 256
2011-03-07 15:54:07.701471 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:07.701471 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:07.701471 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 53 22 22 4a 16 81 45 ba 2c 09 04 92 20 08 11 89 0b d6 12 12 17 71 e5 06 96 25 88 84 88 ab 25 e2 42 b4 88 0f 09 12 71 21 04 a1 44 5c ac b6 78 42 41 74 a1 24 a1 05 41 74 19 a2 4b 32 75 92 04 49 10 84 60 6a a8 93 20 09 24 41 fc 05 09 84 30 55 a8 9a 5a a9 4c b5 88 0b 2d 51 84 10 4d bd 24 49 92 60 6a 04 41 88 10 64 0a a7 18 2a 49 02 41 88 43 10 4a 74 19 42 53 04 5f 29 98 3a 99 ca 14 4e 09 21 72 44 5c 28 49 90 e8 82 44 8c 20 10 72 44 5c 16 72 42 44 53 64 09 22 b2 24 ae 42 53 cf 33 05 16 92 8b b8 10 04 42 24 2e 89 27 e2 42 e2 92 88 0b 99 c5 13 5f 14 52 53 40 47 53 85 1a ea 40 c8 16 71 99 5a 28 42 18 ad b1 88 cb 62 4c 5c 1c 81 40 5e c4 e5 83 40 5b 25 11 57 5b c4 65 41 10 22 22 a2 c4 90 23 22 08 24 41 90 29 9c 0a a3 84 58 33 24 f1 42 24 91 22 c2 22 2e e2 22 ae 4d e1 fc
2011-03-07 15:54:07.701471 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 83, length 256
2011-03-07 15:54:07.841501 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:07.841501 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:07.841501 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 54 20 54 53 d3 92 2c 94 10 d1 54 e9 c4 05 02 a1 a9 87 10 89 0b 21 90 ab 24 11 d7 a3 39 4e 7d 94 88 0b 82 4c 1d 34 22 12 17 a2 44 5c fc 8b 23 e2 82 c4 16 5d 10 64 aa a2 08 4a f0 00 21 22 04 a1 20 08 42 2e 22 2e a7 8b 0c d1 45 5c 20 10 22 04 35 04 21 71 99 9a 28 11 97 45 17 c5 08 09 82 90 23 b4 a5 91 74 49 12 5d 48 5c 6f aa f3 04 09 04 9d fb 45 b0 20 c8 d4 42 15 49 20 8a 10 5a 12 22 4a c4 07 81 24 0b a6 32 d5 22 2e 94 f8 a2 90 25 88 b0 88 0b 4d b5 24 10 08 d1 54 05 a1 a9 4c 05 85 53 85 82 85 1c f1 e3 4d 5c 92 84 68 09 92 88 8b b8 04 d1 05 41 28 48 22 3e 04 21 04 99 92 73 60 0a 0a 42 34 15 22 be 44 17 24 e2 83 40 12 47 c4 c5 93 85 c4 45 97 a9 97 64 2e 20 44 e2 92 24 cb 22 2e e2 22 42 90 58 c4 90 85 82 e8 42 14 44 97 21 24 2c 09 92 44 7c 89 59 10 04 12 c1 42 53 03
2011-03-07 15:54:07.841501 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 84, length 256
2011-03-07 15:54:08.000643 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:08.001651 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:08.001651 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 55 21 71 11 17 5f 88 28 11 17 71 d1 85 96 29 9c c2 50 38 c5 21 ba 10 82 4c 39 48 32 e5 1c e7 72 49 86 4c e1 54 45 e1 7c a8 a3 70 9e 29 a7 8e 1a 6a a8 36 a2 a9 22 fa 12 12 97 21 53 c7 5d 30 35 90 44 5c 11 97 24 53 97 b8 88 0b 04 21 5b a6 0a 05 96 84 88 28 21 19 92 20 08 25 ba 2c e2 c2 07 81 86 25 f3 45 5c 08 53 38 15 d4 8b b8 10 47 89 b8 b0 46 34 85 53 74 99 12 54 81 7b 0b 89 87 4d 17 7f 10 c8 91 60 71 9d 21 24 2e 08 42 08 32 64 a8 4b 42 53 38 08 11 21 d0 10 84 12 71 49 0c 81 40 12 71 75 99 c2 a9 22 8b b8 04 11 17 e2 86 88 cb 20 89 b8 88 4b 89 8b b7 c5 17 71 d1 45 5c 28 81 40 c8 9e f8 86 0c 61 d1 85 20 90 8e 4b 43 10 22 3d 89 f8 74 b5 6c 8a 4e b9 25 44 7c d4 11 25 b4 50 68 9e 46 09 9d b8 50 36 48 07 81 10 11 11 11 11 11 11 73 03 0b d1 54 16 22 c2 14 2c 08 42 34
2011-03-07 15:54:08.001651 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 85, length 256
2011-03-07 15:54:08.141600 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:08.141600 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:08.141600 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 56 6b ca a9 a6 70 0a a6 70 4a 85 aa 38 41 e1 d4 50 02 42 48 12 5b 6c 21 5a a6 08 d1 45 17 7f 49 02 41 a6 ce 0c 11 17 71 41 10 b2 c8 22 2e 66 21 22 1a 24 31 8f 88 e6 4f 39 e5 54 83 88 4b 90 44 17 c2 10 4a 30 48 a2 2f 19 b2 0c 41 12 24 99 e1 bc 29 5c 82 10 82 2c ba 0c 92 2c 09 4d ad d4 8b b8 24 c9 22 2e 10 84 16 71 59 92 04 0b 61 aa 08 2d 41 82 20 c8 5a 62 16 24 41 82 d0 54 a1 70 6a 24 08 82 10 06 41 42 44 44 be 20 10 22 22 22 22 22 22 22 12 5b a7 5c 52 99 0a 5e 28 e7 47 39 97 53 38 25 20 44 44 10 08 91 0c a1 20 44 e2 32 d5 dc b3 97 24 84 20 b3 a1 9c 1a 88 2e 53 70 45 c4 1a 06 09 df c4 d5 d6 bc 59 44 ae 11 89 8b 3e 5f cc 42 6c c4 11 82 1a 79 42 44 44 44 44 44 44 44 44 44 44 44 44 09 11 11 11 a6 70 2a c5 30 5d 2a 8d 3c 21 22 22 22 72 e3 22 2e c4 e9 62 8b b8 88 8b
2011-03-07 15:54:08.141600 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 86, length 256
2011-03-07 15:54:08.301396 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:08.301396 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:08.301396 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 57 b8 e8 22 2e e2 22 ae 08 41 28 11 17 b2 88 b8 24 10 85 43 12 4c b5 04 d1 85 e6 38 85 b3 e6 38 77 0a a7 9a 17 11 17 71 a1 b9 e0 b8 88 5b 90 40 30 24 2d 90 65 21 5d 82 04 b1 2c 81 24 89 42 15 71 21 22 d2 85 88 88 88 88 08 41 88 68 10 24 41 10 04 89 2d e2 42 ba e8 42 24 17 4a 20 90 24 49 92 24 49 52 6e 60 99 57 d4 b3 2c e2 92 2c e2 22 be 39 2e 94 24 e2 12 24 09 04 43 90 24 98 7a 48 92 10 86 10 25 44 0b 51 42 44 44 44 09 21 31 ad a2 8b 5c 7c 49 c4 45 4a 11 97 21 48 90 20 08 04 92 90 b8 40 10 4a 4c 8b 88 8b b8 24 e2 23 22 22 22 22 e6 ce 62 16 71 d1 45 d7 88 88 88 88 88 88 88 88 88 88 88 88 c4 9a f8 3c 31 44 17 5d 88 68 be b1 38 15 65 67 86 2c 44 44 44 44 44 44 44 43 cd 70 08 0d 59 20 21 12 17 4a 86 10 a6 36 03 84 12 71 41 82 44 6a 11 17 ed 2a 42 e2 8a 88 88 88 a8
2011-03-07 15:54:08.301396 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 87, length 256
2011-03-07 15:54:08.441218 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:08.441218 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:08.441218 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 58 10 84 44 93 55 48 74 41 11 31 d9 81 24 08 82 cc 29 28 97 39 15 28 74 44 44 c4 88 3f cd 40 8b 3a b0 48 88 8b 38 62 8b 23 9e f8 e2 89 5c 04 11 f7 8b 2d 62 8b 5c 1c b1 67 8b 23 b6 f8 a2 8b 2d 0a 91 8b b8 e8 a2 8b 2f 16 91 5a 84 88 b8 c8 45 2e 8e 18 22 2e b6 98 45 42 8c 89 8b 59 8c 39 33 44 7c e2 a2 8b 2e e2 22 2e ba d8 73 44 17 f1 e9 d3 c5 11 b3 e8 22 2e 72 d1 c5 10 f3 c4 c5 10 43 1c 11 17 71 f1 45 5c 1c 11 17 71 71 94 66 e4 e2 88 8b d8 42 d2 c5 13 c9 a2 44 3c b1 c5 16 85 e8 22 59 7c 11 17 43 6c 11 17 5b 94 88 27 b6 88 8b 37 b2 07 41 a6 82 07 25 28 17 71 81 20 73 87 90 b8 20 21 71 11 1f e9 f3 45 7c ba d8 22 2e b6 88 8b 27 48 74 11 17 8b 18 22 2c be 88 c7 9e 2d 9e 40 20 89 b8 24 e2 a2 8b 21 e2 62 88 2e fa 74 d1 65 aa a9 a1 04 e5 22 2e 10 64 0a 87 4c e1 dc 05 41
2011-03-07 15:54:08.441218 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 88, length 256
2011-03-07 15:54:08.514643 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17
2011-03-07 15:54:08.514643 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Too many retries. Giving up.
2011-03-07 15:54:08.514643 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_D_TX
2011-03-07 15:54:08.514643 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set rx type 0
2011-03-07 15:54:08.514643 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set tx type 4
2011-03-07 15:54:08.514643 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from state 17 to 3
2011-03-07 15:54:08.514643 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  DCN with final frame tag
2011-03-07 15:54:08.514643 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Tx:  ff 13 fa
2011-03-07 15:54:08.581059 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:08.581059 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:08.581059 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 59 a6 70 c8 14 4e 15 8e 53 45 48 7c 10 c8 54 50 21 c6 74 d1 45 7c 86 88 8b 2d ba e8 d3 c5 10 04 21 71 f1 c4 13 43 12 f2 c6 b7 88 0b 82 d0 94 73 9c 0f 08 91 b8 88 0b 04 92 40 d4 50 0e 81 40 92 20 09 12 36 04 21 08 e4 20 c8 90 25 08 12 7d 89 f8 74 49 90 94 3e 5d 92 21 73 9c b7 41 20 09 04 21 08 64 49 c4 45 7c 09 89 0b 82 10 11 04 92 88 2b 4d b4 c4 2c 9e 78 e2 8b 2f b6 a8 10 b7 cf 99 27 2e 33 44 5c 7c d1 c5 10 43 bc e9 e2 0b 89 0b 19 22 2e e2 92 88 4b 22 2e 86 e8 a2 8b 31 71 81 40 a6 02 42 fa c4 45 17 7d e2 d3 27 2e 10 84 74 d1 27 2e ba e8 d3 45 1f 82 90 be 44 5c 06 21 f1 89 8b b8 24 a4 8b 2e e2 d3 45 5c 74 11 17 5d 20 08 e9 82 24 81 20 04 41 88 20 08 19 22 2e c6 c4 a7 4f 7c fa 12 63 53 2e 53 0b 04 81 60 a1 29 9c c2 29 a7 70 08 d1 10 04 42 c9 10 43 c4 45 7c 49 10
2011-03-07 15:54:08.581059 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 89, length 256
2011-03-07 15:54:08.741022 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:08.741022 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:08.741022 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 5a 7b 08 82 24 08 12 71 41 92 88 0b 92 44 5c 20 08 05 49 a6 70 0a e7 2e 48 74 31 44 5c c4 05 89 21 83 04 99 c2 45 5c 20 49 86 d0 10 4a 86 50 82 20 48 90 e8 4b 20 c8 94 8b b8 88 4f 5f 82 64 08 92 04 49 10 24 09 92 04 49 22 2e e2 22 9e 20 93 1d a2 0b 82 24 48 90 40 20 09 04 82 25 c1 90 05 82 0c 19 82 20 43 16 08 32 c7 21 73 1c 32 64 08 89 8b 2e 08 a2 8b 6e e2 02 41 16 71 41 92 24 e2 a6 33 04 09 a2 0f 41 f4 41 92 24 49 22 9e 04 49 92 24 48 82 18 96 40 10 24 08 82 04 41 90 20 08 82 24 41 90 20 09 12 04 41 90 20 81 20 08 12 44 b7 24 09 a2 1b 24 4b 10 24 90 24 e2 22 4e 92 a9 ad c4 bd 8a 23 86 98 c5 11 43 1c 31 8b b8 88 8b b8 58 44 5c 7c 11 2c be f8 e2 89 21 e2 62 88 0a 11 17 17 b1 27 3e 5f 24 c4 10 71 11 9f 2d e2 62 4c 9f 37 5d 74 31 44 7c e2 62 4c 17 43 74 d1 c5 98
2011-03-07 15:54:08.741022 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 90, length 256
2011-03-07 15:54:08.880779 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:08.880779 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:08.880779 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 5b 2e e2 a2 8b 2e e2 a2 8b b8 e8 d3 45 17 71 31 8b 23 86 e8 d3 a7 8b 3e 71 11 9f 33 67 fa c4 a7 8b 2e c6 74 d1 e7 88 2d c6 c4 45 9f b8 d8 22 2e b6 f8 42 86 40 10 22 71 41 82 20 24 2e e2 82 a0 21 11 17 71 31 44 5c c4 45 17 71 d1 45 7c 73 e7 38 77 8e 73 e7 4e 39 77 ca b9 53 ce 9d e3 dc 39 ce 71 ca 29 e7 2e fa 45 5c a0 c4 45 7c e2 22 2e e2 17 08 84 13 0f 82 0c ad 10 34 71 81 40 05 41 48 f7 74 41 30 71 41 d0 36 04 c5 b1 de b7 42 1e 44 f9 55 d7 07 81 7e 61 7d 10 28 46 41 c4 05 a2 70 69 84 41 50 bc a2 db 10 b4 24 0c 81 26 de c4 05 41 f7 ff a6 3c 04 ad 48 5c c4 45 17 04 0d 41 58 c4 7f 44 34 84 10 64 6a 84 58 0e a2 5c c5 50 e9 58 8a a5 58 68 10 5e 52 41 d0 10 34 62 81 40 9b c2 29 1c 9a b8 40 10 06 81 24 85 24 10 08 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11
2011-03-07 15:54:08.880779 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 91, length 256
2011-03-07 15:54:09.040915 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:09.040915 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:09.040915 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 5c 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 0d 92 72 73 41 10 82 40 12 08 84 c4 45 7c 08 42 08 42 89 b8 20 31 86 64 49 c4 45 1f 82 2c e2 a2 4b 62 8b 3f 71 49 74 99 b7 e8 82 04 02 59 20 90 29 e7 4e e1 94 73 9c c2 b9 83 24 89 b8 18 a2 8b 2e 84 21 08 1a 2d 41 20 0d 81 10 a6 84 40 20 84 05 82 20 08 21 c8 10 1a 82 20 4b 82 20 4b 0d b2 0d a1 39 4e e1 90 39 2e 48 10 04 41 10 64 48 82 39 0e d1 17 04 41 82 20 49 10 04 09 82 e8 43 22 2e 09 04 19 04 8b b8 20 49 c4 4d 7c e2 4b c4 51 41 f4 88 13 04 d1 09 92 24 49 82 24 09 24 49 82 24 41 12 04 82 20 48 10 04 09 82 20 48 10 04 09 82 20 41 10 04 09 82 20 41 12 24 08 82 24 08 92 04 f1 96 20 49 12 04 49 22 2e 48 02 49 90 24 08 b2 24 41
2011-03-07 15:54:09.040915 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 92, length 256
2011-03-07 15:54:09.180761 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:09.180761 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:09.180761 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 5d 92 04 e2 23 09 a2 5a 88 28 10 22 22 22 22 22 22 22 22 22 22 22 22 22 22 22 08 64 19 12 24 0a 4c 3d 25 28 33 4f 6d 25 09 82 06 81 5a 41 a0 ab 15 92 0e 6d 21 96 c3 6d 4c 5c c4 55 12 f1 20 5b c4 ef 28 59 06 15 62 7d 8c 58 b5 89 4f 5f 89 eb e9 e2 89 2e 08 c2 21 0a 87 24 53 38 25 28 87 10 21 08 2d 1c 2d 1c 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 11 91 16 08 49 25 09 11 11 11 11 11 11 11 11 0c 82 39 17 08 24 49 02 89 1a ea 21 08 84 1c 81 29 05 65 8b 52 11 11 11 11 11 11 11 11 3b 77 0e a6 b6 92 25 95 a4 48 5c 34 39 d1 02 55 dc 18 51 d7 16 f7 8b 42 24 0b 43 10 22 ed 21 ea b4 e2 c6 05 41 48 6d b1 12 72 2b 8b 10 71 87 48 21 64 4c 5c a8
2011-03-07 15:54:09.180761 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 93, length 256
2011-03-07 15:54:09.321648 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:09.321648 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:09.321648 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 5e f4 91 6a 35 b7 70 aa 50 09 9a da 1a 14 13 85 4b e2 48 22 2e 24 97 a9 4c 21 a8 4d 16 24 e2 42 c6 a6 b2 88 22 08 84 08 41 18 59 89 ab 08 2d 26 14 71 63 c4 fd e2 88 59 e4 22 64 e2 22 3e e9 22 e5 c4 45 5c 84 c5 16 04 35 71 21 f9 0c 71 83 8a 54 2b f7 88 21 66 71 d5 cc 22 90 76 c2 1a b5 89 22 22 44 5c 88 8d dc 18 99 82 88 ab 50 02 81 24 e2 92 3e 77 91 c4 97 29 9c 3a 8e 53 38 15 14 98 1a 91 8b 21 ae 73 15 71 51 5a 20 08 49 29 89 28 a3 84 12 71 c1 12 44 97 24 c8 42 53 58 10 f2 46 89 b8 e8 73 a3 ca 14 05 84 12 22 a2 44 85 10 25 34 45 41 3d 84 5c 64 51 08 09 8b 4c 32 35 62 48 22 2e 64 16 29 27 21 53 59 7c d1 c5 99 af 26 62 8e 90 b8 28 84 6f 04 81 70 08 04 4b 92 e9 02 81 d0 3c a8 08 41 61 ba 88 ab 46 e2 4f 0a 21 71 cd 2c be 38 42 02 ab 88 3a 22 32 0b 11 27 5d cc 32 d5
2011-03-07 15:54:09.321648 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 94, length 256
2011-03-07 15:54:09.481538 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:09.481538 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:09.481538 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 5f fc 28 32 85 a0 70 6a f2 45 5c 5e c4 c5 13 b3 88 22 e2 62 16 5d 84 45 5c 1c 11 d7 c8 d6 c4 c5 9e 27 7e 3c 11 55 0c 29 09 b1 88 27 be e8 62 d6 6c b1 c5 11 f2 26 2e e6 4d 35 85 53 38 f5 d4 50 85 2a 94 53 0a 9e 79 50 82 ca 86 20 a4 8b 79 e2 62 8b 27 ba 78 22 2e 44 86 e8 32 d5 54 53 ce 9f 3a 11 9f 2f e2 33 4f 17 cb c4 45 17 43 6c d1 05 41 c8 99 31 e7 89 0b 95 2f e2 22 2e 8b 33 4f 0c 29 71 71 e6 88 b8 e8 a2 18 12 4f 10 84 c4 c5 10 71 21 08 64 aa 21 94 20 10 d2 d5 c4 45 5c 10 c8 49 0c 81 60 1e e9 83 40 b3 57 4e 74 11 9f 33 11 2a 21 4f 92 84 90 10 11 82 cc e5 12 c8 42 84 f9 c6 ca 17 9a 1a 08 51 a2 8b a8 42 24 17 92 8b 2e 56 82 20 24 3c 7b 86 88 8b 45 a6 e4 10 d2 85 12 4c b5 78 62 0b 2d 09 11 2d 10 08 41 20 e4 cd 13 5d e4 a2 cb 22 2e 43 c8 10 7b 8b b8 24 09 25 ba d8
2011-03-07 15:54:09.481538 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 95, length 256
2011-03-07 15:54:09.595217 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2011-03-07 15:54:09.621368 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:09.621368 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:09.621368 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 60 e2 88 21 e2 b1 85 a6 60 76 d4 53 0b 7b 3e 20 94 90 bd 44 7c 48 74 49 74 a1 04 41 a6 38 84 28 f1 45 5c c4 25 48 22 2e 08 92 24 49 62 0b 92 44 5c c4 05 12 6f 8b b8 f8 13 17 63 e2 22 2e 64 0b 82 24 ba e8 4b 88 08 41 12 24 ba 2c 49 22 2e c2 d3 45 5c 28 11 97 64 ca 85 e6 21 41 6c 19 62 2c 08 92 20 09 12 04 21 4a c4 65 76 14 0e d1 25 48 92 40 14 0e 31 64 49 a6 1e 64 08 05 21 04 41 90 05 89 2e e2 82 40 30 e5 54 f3 88 10 84 68 2a 78 f0 a7 06 42 94 10 04 32 15 12 64 a1 04 81 50 42 44 e2 42 c9 05 41 d4 84 b9 42 91 b8 58 85 12 12 17 f2 64 88 5c 96 44 86 04 51 49 74 71 44 21 ae 62 88 b8 50 22 2e 56 a2 90 44 7c 53 70 e4 88 5c 6c 11 17 24 e2 e2 22 e4 89 b5 38 82 04 41 a6 32 6f 21 b3 10 25 84 24 71 c4 98 b8 10 04 92 d8 13 97 a9 86 10 24 09 d9 22 2e e2 42 64 99 2d c9 ec 38
2011-03-07 15:54:09.621368 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 96, length 256
2011-03-07 15:54:09.674655 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2011-03-07 15:54:09.674655 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Disconnecting
2011-03-07 15:54:09.674655 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E
2011-03-07 15:54:09.674655 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set rx type 0
2011-03-07 15:54:09.674655 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set tx type 1
2011-03-07 15:54:09.674655 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from state 3 to 2
2011-03-07 15:54:09.761403 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:09.761403 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:09.761403 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 61 4e 35 44 17 04 59 c4 05 41 96 24 11 17 24 e2 32 95 25 a1 d9 71 2e e2 4b c4 85 08 a6 8e 1a 11 17 71 d1 05 81 38 9b 0a 91 4b 20 94 20 11 17 5b 28 09 42 09 25 90 88 0b 04 92 20 08 41 22 3e 43 c4 85 96 24 21 22 22 22 08 64 c8 92 04 02 41 10 24 41 20 58 68 b1 85 68 c1 14 41 0d d5 d4 40 12 4a 82 18 02 81 10 11 2d 44 44 e4 08 11 4d 71 ca 05 81 60 ea 65 c8 54 83 24 04 53 21 43 28 21 5a 92 84 c8 10 98 6a aa c5 10 a2 21 8b b8 24 c9 54 4d 29 64 ca 29 1c 84 c4 25 49 20 08 51 42 69 8e 18 4b 92 84 a6 02 12 c4 59 2a 85 e8 62 48 52 be 26 17 4c 71 2a 8b b8 4c 39 1f aa 29 a7 e0 c4 c5 9e 21 e2 22 2e 9e c8 45 17 43 74 2d 11 97 c4 2c 08 9a 21 e2 52 c3 23 5f c4 45 17 72 04 25 2e 8e 88 0b 61 2a 53 4e 75 25 ae 2d 08 e4 a6 8e c2 a9 a1 9c 1a ea cd 9e fd 58 38 42 d0 10 84 14 23 5d 20
2011-03-07 15:54:09.761403 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 97, length 256
2011-03-07 15:54:09.776509 [DEBUG] ozmod_libpri.c:106 T203 counter expired, sending RR and scheduling T203 again
2011-03-07 15:54:09.776509 [DEBUG] ozmod_libpri.c:106 Sending Receiver Ready (55)
2011-03-07 15:54:09.776509 [DEBUG] ozmod_libpri.c:106 
> [ 00 01 01 6f ]
2011-03-07 15:54:09.776509 [DEBUG] ozmod_libpri.c:106 
> Supervisory frame:
2011-03-07 15:54:09.776509 [DEBUG] ozmod_libpri.c:106 > SAPI: 00  C/R: 0 EA: 0
>  TEI: 000        EA: 1
2011-03-07 15:54:09.776509 [DEBUG] ozmod_libpri.c:106 > Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
> N(R): 055 P/F: 1
> 0 bytes of data
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 
< [ 02 01 01 9f ]
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 
< Supervisory frame:
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 < SAPI: 00  C/R: 1 EA: 0
<  TEI: 000        EA: 1
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 < Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
< N(R): 079 P/F: 1
< 0 bytes of data
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 Handling message for SAPI/TEI=0/0
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 -- ACKing all packets from 78 to (but not including) 79
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 -- Since there was nothing left, stopping T200 counter
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 -- Stopping T203 counter since we got an ACK
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 -- Nothing left, starting T203 counter
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 -- Unsolicited RR with P/F bit, responding
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 Sending Receiver Ready (55)
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 
> [ 02 01 01 6f ]
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 
> Supervisory frame:
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 > SAPI: 00  C/R: 1 EA: 0
>  TEI: 000        EA: 1
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 > Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
> N(R): 055 P/F: 1
> 0 bytes of data
2011-03-07 15:54:09.781544 [DEBUG] ozmod_libpri.c:106 -- Restarting T203 timer
2011-03-07 15:54:09.787603 [DEBUG] ozmod_libpri.c:106 
< [ 00 01 01 9f ]
2011-03-07 15:54:09.787603 [DEBUG] ozmod_libpri.c:106 
< Supervisory frame:
2011-03-07 15:54:09.787603 [DEBUG] ozmod_libpri.c:106 < SAPI: 00  C/R: 0 EA: 0
<  TEI: 000        EA: 1
2011-03-07 15:54:09.787603 [DEBUG] ozmod_libpri.c:106 < Zero: 0     S: 0 01: 1  [ RR (receive ready) ]
< N(R): 079 P/F: 1
< 0 bytes of data
2011-03-07 15:54:09.787603 [DEBUG] ozmod_libpri.c:106 Handling message for SAPI/TEI=0/0
2011-03-07 15:54:09.787603 [DEBUG] ozmod_libpri.c:106 -- ACKing all packets from 78 to (but not including) 79
2011-03-07 15:54:09.787603 [DEBUG] ozmod_libpri.c:106 -- Since there was nothing left, stopping T200 counter
2011-03-07 15:54:09.787603 [DEBUG] ozmod_libpri.c:106 -- Stopping T203 counter since we got an ACK
2011-03-07 15:54:09.787603 [DEBUG] ozmod_libpri.c:106 -- Nothing left, starting T203 counter
2011-03-07 15:54:09.787603 [DEBUG] ozmod_libpri.c:106 -- Got RR response to our frame
2011-03-07 15:54:09.787603 [DEBUG] ozmod_libpri.c:106 -- Restarting T203 timer
2011-03-07 15:54:09.921374 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:09.921374 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:09.921374 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 62 10 0e 41 88 a6 82 8a 88 28 49 21 f6 c8 11 08 64 21 47 0c 51 2e be 04 11 37 36 44 71 e1 88 82 1e a6 5a e8 88 6c 59 6c 41 94 6b 41 6e 11 17 5d 8c 21 08 41 20 0c 6d 15 08 84 18 62 0b 9b 2e be d0 7b 8e 10 89 af 20 d0 10 d5 1a 91 ae 10 0c 42 b4 4c 2d 54 e6 4b 09 27 2e ab d0 90 a9 4c 11 d4 48 62 08 07 81 26 ae 1a 89 6b 9c 21 d6 92 f8 22 2e e6 d9 42 8e 88 8b a7 d9 92 46 53 19 64 4a 24 be 54 08 02 4d 7c 66 49 88 f4 97 20 f5 1c a1 a5 90 88 6b 86 38 82 a0 39 e2 89 21 94 20 15 a2 21 86 d8 42 be 4c 35 b4 29 17 8e 38 8e c4 55 22 2e 0b cd 15 d4 08 25 4b 12 24 11 97 21 64 16 22 04 59 12 f1 21 c8 74 40 86 10 11 2d 53 59 68 b1 85 10 6f 9e 29 42 c4 45 97 84 28 21 08 24 31 96 20 c8 bc 40 88 68 08 25 10 c8 92 cc 61 4a 50 85 1a 2a c4 22 44 b6 24 96 04 99 82 41 92 41 06 21 71 d1
2011-03-07 15:54:09.921374 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 98, length 256
2011-03-07 15:54:10.061371 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:10.061371 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:10.061371 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 63 45 5c 74 59 c4 25 21 43 88 28 21 04 21 04 a1 a9 a6 0e 64 aa a9 a7 86 1a 08 25 94 24 8e 24 be 24 09 d1 d4 0a a1 64 21 22 71 a1 41 06 c1 54 40 88 f4 25 89 3f 0c 21 32 36 84 c8 90 20 41 96 41 a6 5c 92 a9 a1 82 72 aa 08 39 02 45 86 88 8b 2d 84 20 24 1e a2 64 2a a8 81 20 b1 25 b1 87 20 94 0c 32 f5 d4 0b 82 10 25 a4 8b 2e 48 4b 88 28 21 43 28 b1 05 41 16 04 a1 04 c9 20 53 01 21 4a 1c 41 10 22 08 32 d5 54 81 d0 54 50 4d e1 14 0e 21 c2 54 40 90 24 8e 38 92 40 54 43 88 28 49 92 64 90 c4 df 90 44 5c cc 92 10 d9 23 04 21 a2 64 aa a9 a1 86 6a 6a 78 53 cd 87 aa 21 dc 82 e8 22 2e e2 42 8e 88 8b 27 8e 14 09 cb 54 89 48 5c 6d aa 24 2e e2 2a 08 42 c4 b1 cc 83 c2 a9 87 4c 85 a4 67 08 cb d4 53 1d 25 16 2d 19 84 96 cd 50 11 17 5d cc 42 53 d1 bc a9 90 04 89 b8 e8 42 e2 92 4c e5
2011-03-07 15:54:10.061371 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 99, length 256
2011-03-07 15:54:10.221451 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:10.221451 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:10.221451 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 64 88 16 b2 05 02 21 5b 12 f2 97 f4 51 83 10 89 8b 2d 64 8b 21 e2 32 85 53 15 84 8f 92 60 84 84 83 40 c8 17 08 84 23 12 97 a9 68 aa 29 07 21 6f 34 d5 bc 29 9c 3f 05 d7 41 94 4b c6 9a 45 82 d8 42 e2 62 88 2d a9 21 08 89 0b a6 3a 4e 5c 8a 12 8e 3c 0d 51 ea 6c 8d 6c 99 6a 99 22 b4 83 40 12 5d b7 84 88 c4 05 02 99 e3 14 0e 42 c4 4d 05 85 53 2b 75 20 9c 23 53 71 53 04 e5 22 2e c4 89 8b 27 f3 01 99 3a 10 4a c8 11 12 17 b3 10 11 91 5c 6c b1 45 5c a3 a0 fe 46 be e8 0f 41 88 c4 d5 16 a2 05 82 d0 14 2c 24 2e c4 47 44 44 3c 4c ad b2 90 91 25 19 84 90 24 53 c3 61 90 84 16 9a 0a aa 70 0e b2 88 4b 82 84 a6 32 95 79 50 38 c8 54 16 5d 58 0c 99 6a 11 17 22 cc 42 82 24 0b 12 71 21 47 c4 c5 2c 09 11 4d 39 c5 29 a7 82 c2 21 09 21 08 4d 85 24 b6 78 62 48 b2 18 92 cc 70 aa 50 30 08
2011-03-07 15:54:10.221451 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 100, length 256
2011-03-07 15:54:10.361281 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:10.361281 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:10.361281 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 65 11 91 21 83 10 25 53 64 ea 65 0a 86 e8 62 8b 2e cb 32 d5 d4 51 38 e5 32 15 32 84 c8 5f 12 44 17 a2 44 2e 14 84 92 29 0c 75 3c 43 a6 5c 74 41 10 22 22 4c b9 10 82 50 10 22 4c 1d 1f 2a 44 2e e2 42 49 22 2e 12 e2 48 92 4c 2d 22 2e 53 99 92 cb 42 53 59 c4 c5 13 b3 10 d1 62 8b b8 50 42 34 57 08 25 e4 09 2d 96 11 51 10 04 a1 64 2a 73 09 35 dc 29 58 88 28 41 10 22 22 4c 39 15 54 4b 10 5d 06 21 22 24 e6 99 65 8a 10 71 f1 65 6a 28 9c 3a d1 e7 cf 11 4a 3c b1 c5 99 2d f6 74 d1 c3 92 a9 48 5c 70 35 25 91 a9 3a 43 c4 c5 10 4e 5c 7c 21 4f 12 12 17 5d 48 97 84 a6 5c 28 49 85 88 23 22 22 b2 25 10 71 99 6a 2a 4a 82 90 59 38 ce 16 cb 28 21 8e 33 b4 64 ea 23 71 21 2e e1 02 21 43 12 5f 10 84 44 4c 5c c4 85 92 41 48 78 ba d0 14 17 43 88 88 6a ea 85 88 8f 12 61 81 a8 a6 22 ce 13
2011-03-07 15:54:10.361281 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 101, length 256
2011-03-07 15:54:10.501196 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:10.501196 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:10.501196 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 66 6e 2a 43 23 d6 12 e2 c4 c5 2c ce d2 6c 21 71 41 d0 c4 45 5c c4 c5 11 a4 22 2e 10 08 ed 91 b8 e8 6a 9c f8 61 6c f3 4e 5c c4 c5 d8 92 38 42 e2 f3 33 15 89 6b 6b 08 42 8e 46 24 ae e6 88 b8 18 9a 3f 5f 0c 21 f1 91 b8 88 3f 47 48 7c e2 2a 9e 4c 71 ca a9 a1 9c 87 78 62 88 3e 8b ba 90 b8 65 b6 0c 42 e2 a2 8b 42 12 4a 88 88 6c 31 04 73 20 2a 53 5a 14 32 54 22 2e 09 04 42 53 2d e2 42 89 2f 53 59 e4 92 10 25 e2 b2 4c 55 54 a1 06 42 89 b8 24 64 cc d8 54 a6 5e c4 c5 16 43 c4 25 11 17 b3 18 73 c4 13 04 21 b3 2c e2 92 24 b4 10 99 65 8a 53 19 32 c5 b9 00 21 9a 72 99 6a 11 17 43 0c 99 ca 54 50 cd 9f 72 0a 07 a1 a9 86 10 d9 62 11 2c 89 f8 8c 4d 05 c7 a9 a7 70 2a 38 17 5d 3c 99 e2 20 33 9c 0a 0a e7 ce 87 1a ca a9 2c ba 4c 35 d5 14 ce 9b e3 62 6f ca 79 a1 5e 20 41 10 4a c4 63
2011-03-07 15:54:10.501196 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 102, length 256
2011-03-07 15:54:10.660812 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Stop none (0 remaining)
2011-03-07 15:54:10.660812 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  FCD without final frame tag
2011-03-07 15:54:10.660812 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Rx:  ff 03 06 67 4b 92 40 42 b6 88 8b 27 96 e9 4b 16 47 90 24 8e 4c 05 24 19 24 11 97 24 a1 25 49 10 64 49 88 90 10 0d 09 c4 13 0a a2 cb 92 50 42 14 44 97 40 0c 49 16 71 11 97 64 aa 21 44 94 20 a1 85 88 88 88 88 88 88 88 88 06 21 a2 04 81 10 11 11 11 11 11 11 51 10 b3 10 25 64 ad 88 88 12 71 a1 a9 a0 4e 12 71 d1 25 11 17 a2 a9 a6 60 aa e6 47 05 15 02 81 24 c6 16 32 84 10 08 25 64 8c 28 59 88 12 88 c2 21 53 2d e2 32 c5 a9 07 21 08 64 aa 39 0e 32 e5 54 a6 32 75 22 6e 86 88 0b 12 43 c4 65 a1 c4 9e 2d 49 10 5d 92 05 02 49 a6 b2 24 48 48 97 64 b1 87 64 ea 20 94 2c 41 28 c1 22 2e 49 42 08 82 44 5c 90 24 09 2c 84 21 90 2c 94 10 25 41 30 84 86 90 f8 88 88 88 88 c4 85 c4 65 11 17 24 e2 42 58 c4 85 86 50 22 2e 44 08 42 58 88 88 88 88 c8 3c 22 22 22 22 22 22 22 22 22 04 a1 44 5c e6 76
2011-03-07 15:54:10.660812 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Storing ECM frame 103, length 256
2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Send complete in phase T30_PHASE_E, state 2
2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:323 ==============================================================================
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:335 Fax processing not successful - result (48) Disconnected after permitted retries.
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:340 Remote station id: 
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:341 Local station id:  SpanDSP Fax Ident
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:342 Pages transferred: 0
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:344 Total fax pages:   0
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:345 Image resolution:  0x0
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:346 Transfer Rate:     14400
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:348 ECM status         off
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:349 remote country:   
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:350 remote vendor:    
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:351 remote model:     
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:353 ==============================================================================
2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from state 2 to 32
2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30 Changing from phase T30_PHASE_E to T30_PHASE_CALL_FINISHED
2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set rx type 9
2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX FAX exchange complete
2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX Set tx type 9
2011-03-07 15:54:10.674604 [DEBUG] mod_spandsp_fax.c:293 FLOW FAX FAX exchange complete
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.694686 [DEBUG] switch_core_codec.c:140 OpenZAP/4:1/43851609 Restore previous codec PCMA:8.
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.697748 [DEBUG] switch_core_session.c:885 Send signal OpenZAP/4:1/43851609 [BREAK]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_ivr.c:557 OpenZAP/4:1/43851609 Command Execute hangup()
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 EXECUTE OpenZAP/4:1/43851609 hangup()
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_channel.c:2440 (OpenZAP/4:1/43851609) Callstate Change ACTIVE -> HANGUP
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [NOTICE] mod_dptools.c:732 Hangup OpenZAP/4:1/43851609 [CS_EXECUTE] [NORMAL_CLEARING]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_channel.c:2456 Send signal OpenZAP/4:1/43851609 [KILL]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_session.c:1047 Send signal OpenZAP/4:1/43851609 [BREAK]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_session.c:1917 OpenZAP/4:1/43851609 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_session.c:1917 OpenZAP/4:1/43851609 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_state_machine.c:348 (OpenZAP/4:1/43851609) State EXECUTE going to sleep
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/4:1/43851609) Running State Change CS_HANGUP
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_state_machine.c:535 (OpenZAP/4:1/43851609) State HANGUP
2011-03-07 15:54:10.714665 [DEBUG] mod_openzap.c:540 Changing state on 4:1 from UP to HANGUP
2011-03-07 15:54:10.714665 [DEBUG] mod_openzap.c:556 OpenZAP/4:1/43851609 CHANNEL HANGUP
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_state_machine.c:46 OpenZAP/4:1/43851609 Standard HANGUP, cause: NORMAL_CLEARING
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_state_machine.c:535 (OpenZAP/4:1/43851609) State HANGUP going to sleep
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_state_machine.c:333 (OpenZAP/4:1/43851609) State Change CS_HANGUP -> CS_REPORTING
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_session.c:1047 Send signal OpenZAP/4:1/43851609 [BREAK]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_state_machine.c:314 (OpenZAP/4:1/43851609) Running State Change CS_REPORTING
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_state_machine.c:595 (OpenZAP/4:1/43851609) State REPORTING
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_state_machine.c:53 OpenZAP/4:1/43851609 Standard REPORTING, cause: NORMAL_CLEARING
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_state_machine.c:595 (OpenZAP/4:1/43851609) State REPORTING going to sleep
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_state_machine.c:327 (OpenZAP/4:1/43851609) State Change CS_REPORTING -> CS_DESTROY
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_session.c:1047 Send signal OpenZAP/4:1/43851609 [BREAK]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.714665 [DEBUG] switch_core_session.c:1210 Session 32286 (OpenZAP/4:1/43851609) Locked, Waiting on external entities
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.716645 [NOTICE] switch_core_session.c:1228 Session 32286 (OpenZAP/4:1/43851609) Ended
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.716645 [NOTICE] switch_core_session.c:1230 Close Channel OpenZAP/4:1/43851609 [CS_DESTROY]
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.716645 [DEBUG] switch_core_state_machine.c:427 (OpenZAP/4:1/43851609) Callstate Change HANGUP -> DOWN
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.716645 [DEBUG] switch_core_state_machine.c:430 (OpenZAP/4:1/43851609) Running State Change CS_DESTROY
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.716645 [DEBUG] switch_core_state_machine.c:440 (OpenZAP/4:1/43851609) State DESTROY
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.716645 [DEBUG] switch_core_state_machine.c:60 OpenZAP/4:1/43851609 Standard DESTROY
fcf15a4c-48a4-11e0-919e-7bf53bcc3f81 2011-03-07 15:54:10.716645 [DEBUG] switch_core_state_machine.c:440 (OpenZAP/4:1/43851609) State DESTROY going to sleep

From sadhika at gmail.com  Tue Apr 19 18:13:50 2011
From: sadhika at gmail.com (Sadhika Sharma)
Date: Tue, 19 Apr 2011 19:43:50 +0530
Subject: [Freeswitch-users] Help with rxfax debug "T4 expires in phase
	T30_PHASE_B_RX"
In-Reply-To: <BANLkTikjeWYgw5jJf0Gr0dyFj4_BGd01Zw@mail.gmail.com>
References: <BANLkTimW6WXQ+_Ge+ASG5u1bVe_j36pTHw@mail.gmail.com>
	<BANLkTikAz7Rcci6g_95mmJD7xAm=iHbZTA@mail.gmail.com>
	<4DAD87B1.7020102@coppice.org>
	<BANLkTikjeWYgw5jJf0Gr0dyFj4_BGd01Zw@mail.gmail.com>
Message-ID: <BANLkTinUgd2xApkHjbaWjdJHZpyXvshwJg@mail.gmail.com>

Is there a better way to attach logs? I am not sure if attaching them with
email is good enough.
Also how do i prefix uuid to mod_spandsp logs?

On Tue, Apr 19, 2011 at 7:15 PM, Sadhika Sharma <sadhika at gmail.com> wrote:

> Thanks Steve for replying,
>
> I will explain the problem in clearer terms. Whenever a call lands on
> freeswitch machine, I exceute rxfax(). I get about 5000 fax calls every
> month, out of which about 800 transmissions fail due to hangup cause 548
> 'Disconnected after permitted retries' Complete log is attached.
>
>
> On Tue, Apr 19, 2011 at 6:31 PM, Steve Underwood <steveu at coppice.org>wrote:
>
>> Sadhika,
>>
>> If you post only a fragment of a log, and describe only half of what
>> happens, people will generally ignore what you post. I guess they don't
>> like being treated like they are mind readers.
>>
>> The error seems pretty clear. You aren't getting anything from the far
>> end. Since you give no clue as to what you are doing, beyond trying to
>> receive FAX, that's about as detailed an analysis as you are going to get.
>>
>> Steve
>>
>>
>> On 04/19/2011 05:43 PM, Sadhika Sharma wrote:
>> > Anybody having any idea about this?
>> >
>> > On Fri, Apr 15, 2011 at 7:36 PM, Sadhika Sharma <sadhika at gmail.com
>> > <mailto:sadhika at gmail.com>> wrote:
>> >
>> >     Hi,
>> >
>> >     I need help in debugging hangup cause 548, 'Disconnected after
>> >     permitted retries' while attempting rxfax().
>> >
>> >     Logs show that T4 expires in phase T30_PHASE_B_RX.
>> >     T30 PHASE B concerns exchange of DIS/DCS messages. After sending
>> >     DIS, mod spandsp keeps waiting for DCS and exhausts its retries.
>> >
>> >     Please guide me how to debug the root cause of loss of DCS
>> >     message. Logs are mentioned below:
>> >
>> >     2011-03-07 15:53:48.675180 [DEBUG] switch_ivr.c:557
>> >     OpenZAP/4:1/43851609 Command Execute
>> >     rxfax(/srv/fax/in/fcf15a4c-48a4-11e0-919e-7bf53bcc3f81.tiff)
>> >     2011-03-07 15:53:53.334614 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>> >     Send complete in phase T30_PHASE_B_TX, state 17
>> >     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>> >     DIS:
>> >     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW
>> >     T.30   .... ...0= Store and forward Internet fax (T.37): Not set
>> >     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW
>> >     T.30   .... .0..= Real-time Internet fax (T.38): Not set
>> >     :
>> >     :
>> >     2011-03-07 15:53:53.875238 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>> >     Start T4
>> >     2011-03-07 15:53:53.895315 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>> >     HDLC signal status is Carrier up (-2) in state 17
>> >     2011-03-07 15:53:53.915402 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>> >     HDLC signal status is Carrier down (-1) in state 17
>> >     2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>> >     T4 expired in phase T30_PHASE_B_RX, state 17
>> >     2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>> >     Retry number 1
>> >
>> >
>> >     Please guide me how to debug the root cause of loss of DCS message.
>> >     Thanks,
>> >     Sadhika
>> >
>> >
>> >
>> >
>> > --
>> > Sadhika
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sadhika
>



-- 
Sadhika
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From infos at madovsky.org  Tue Apr 19 18:19:40 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 19 Apr 2011 10:19:40 -0400
Subject: [Freeswitch-users] Help with rxfax debug "T4 expires in
	phaseT30_PHASE_B_RX"
References: <BANLkTimW6WXQ+_Ge+ASG5u1bVe_j36pTHw@mail.gmail.com><BANLkTikAz7Rcci6g_95mmJD7xAm=iHbZTA@mail.gmail.com><4DAD87B1.7020102@coppice.org><BANLkTikjeWYgw5jJf0Gr0dyFj4_BGd01Zw@mail.gmail.com>
	<BANLkTinUgd2xApkHjbaWjdJHZpyXvshwJg@mail.gmail.com>
Message-ID: <5A5BB7C35D0E47CABC729B3F01D4F264@e1705>

JIRA
  ----- Original Message ----- 
  From: Sadhika Sharma 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, April 19, 2011 10:13 AM
  Subject: Re: [Freeswitch-users] Help with rxfax debug "T4 expires in phaseT30_PHASE_B_RX"


  Is there a better way to attach logs? I am not sure if attaching them with email is good enough.
  Also how do i prefix uuid to mod_spandsp logs?


  On Tue, Apr 19, 2011 at 7:15 PM, Sadhika Sharma <sadhika at gmail.com> wrote:

    Thanks Steve for replying,

    I will explain the problem in clearer terms. Whenever a call lands on freeswitch machine, I exceute rxfax(). I get about 5000 fax calls every month, out of which about 800 transmissions fail due to hangup cause 548 'Disconnected after permitted retries' Complete log is attached.



    On Tue, Apr 19, 2011 at 6:31 PM, Steve Underwood <steveu at coppice.org> wrote:

      Sadhika,

      If you post only a fragment of a log, and describe only half of what
      happens, people will generally ignore what you post. I guess they don't
      like being treated like they are mind readers.

      The error seems pretty clear. You aren't getting anything from the far
      end. Since you give no clue as to what you are doing, beyond trying to
      receive FAX, that's about as detailed an analysis as you are going to get.

      Steve



      On 04/19/2011 05:43 PM, Sadhika Sharma wrote:
      > Anybody having any idea about this?
      >
      > On Fri, Apr 15, 2011 at 7:36 PM, Sadhika Sharma <sadhika at gmail.com

      > <mailto:sadhika at gmail.com>> wrote:
      >
      >     Hi,
      >
      >     I need help in debugging hangup cause 548, 'Disconnected after
      >     permitted retries' while attempting rxfax().
      >
      >     Logs show that T4 expires in phase T30_PHASE_B_RX.
      >     T30 PHASE B concerns exchange of DIS/DCS messages. After sending
      >     DIS, mod spandsp keeps waiting for DCS and exhausts its retries.
      >
      >     Please guide me how to debug the root cause of loss of DCS
      >     message. Logs are mentioned below:
      >
      >     2011-03-07 15:53:48.675180 [DEBUG] switch_ivr.c:557
      >     OpenZAP/4:1/43851609 Command Execute
      >     rxfax(/srv/fax/in/fcf15a4c-48a4-11e0-919e-7bf53bcc3f81.tiff)
      >     2011-03-07 15:53:53.334614 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
      >     Send complete in phase T30_PHASE_B_TX, state 17
      >     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
      >     DIS:
      >     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW
      >     T.30   .... ...0= Store and forward Internet fax (T.37): Not set
      >     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW
      >     T.30   .... .0..= Real-time Internet fax (T.38): Not set
      >     :
      >     :
      >     2011-03-07 15:53:53.875238 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
      >     Start T4
      >     2011-03-07 15:53:53.895315 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
      >     HDLC signal status is Carrier up (-2) in state 17
      >     2011-03-07 15:53:53.915402 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
      >     HDLC signal status is Carrier down (-1) in state 17
      >     2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
      >     T4 expired in phase T30_PHASE_B_RX, state 17
      >     2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
      >     Retry number 1
      >
      >
      >     Please guide me how to debug the root cause of loss of DCS message.
      >     Thanks,
      >     Sadhika
      >
      >
      >
      >
      > --
      > Sadhika
      >
      >

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


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





    -- 
    Sadhika




  -- 
  Sadhika



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


  _______________________________________________
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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 wstephen80 at gmail.com  Tue Apr 19 19:46:52 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 19 Apr 2011 17:46:52 +0200
Subject: [Freeswitch-users] Freeswitch codec usage
Message-ID: <BANLkTim0xU+-jaG=XUfePy11AK2ra96a2g@mail.gmail.com>

In order to evaluate the number of necessary G729 codec, I have a question:
if there is a sip-to-sip call in G729 and I do a tone_detect, surely it's
necessary to do a decode operation. When the detect operation is completed
(and I see in the log a message "removed media bug"), for this call the
codec resource is released? The codec resource is allocated to the call for
entire call duration or only during tone_detect phase?

Stephen
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From wstephen80 at gmail.com  Tue Apr 19 20:04:26 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 19 Apr 2011 18:04:26 +0200
Subject: [Freeswitch-users] Port mapping disabled warning in Freeswitch log
Message-ID: <BANLkTi=SXobYbHp5ueY8qK+o1CrdMQnd7A@mail.gmail.com>

If I set log level to warning, I see in the log many row as:

 [WARNING] switch_nat.c:589 NAT port mapping disabled

My freeswitch is running as: "./freeswitch -nonat"

because it has a public IP and all connected gateways have also a public IP.

To avoid this warning I have removed the parameter "-nonat" but if I do
"show nat_map" in fs_cli I see "0 total.".

I have always started Freeswitch with "-nonat", any advise to solve that?
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From e.kornev at dcn.ru  Tue Apr 19 12:56:35 2011
From: e.kornev at dcn.ru (=?UTF-8?B?ItCa0L7RgNC90LXQsiDQlS7QoS4i?=)
Date: Tue, 19 Apr 2011 14:56:35 +0600
Subject: [Freeswitch-users] Async packetization?
Message-ID: <4DAD4E43.1040209@dcn.ru>

Hello
Recently I had issue with IVR file - it sounded faster than original 
sound in audacity.
After changing packetization on Cisco gateway from 80 to standard 160 
isuue disappeared.
Is that was a bug or misconfiguring situation?


WBR
Kornev E.S.


Email secured by Check Point


From william at xofap.com  Tue Apr 19 17:56:26 2011
From: william at xofap.com (William Alianto)
Date: Tue, 19 Apr 2011 20:56:26 +0700
Subject: [Freeswitch-users] No Audio on Gateway Incoming
Message-ID: <BANLkTikbW_g3nAJ_0ODxYohW-PCUvUFyZw@mail.gmail.com>

I have recently setup a freeswitch server and configure with gateway for
outbound/inbound call. I have succesfully done outbound call without any
problem, but whenever I do inbound call, the phone ring but no audio at all.
Has anybody encounter such problem before?
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From i.ivanova at mastervoice.it  Tue Apr 19 19:25:05 2011
From: i.ivanova at mastervoice.it (Irina Ivanova)
Date: Tue, 19 Apr 2011 17:25:05 +0200
Subject: [Freeswitch-users] transfer_ringback triggered by "183 Session
	progress"
Message-ID: <4DADA951.30708@mastervoice.it>

Hi,

I've noticed that if to set transfer_ringback (to any ringback tone) for 
already answered call and then to execute the bridge to some external 
number through the gateway, the ringing is triggered not only by "180 
Ringing" SIP response, but also when "183 Session progress" is received.
Does anybody know if there is a way to make transfer_ringback not to be 
triggered by 183? I need it because in the case when the destination 
number is busy and provider sends me 183 and then 486 (Busy here) the 
caller hears one ringback tone and then the busy tone which makes an 
impression that the called party rejected the call.

Thanks,
Irina

-- 
================================================================

Distinti saluti
--

Irina Ivanova
Settore Sviluppo MasterVoice

tel: +39 0522 1590100
fax: +39 0522 331673
mob: +39 334 6449290
e-mail: i.ivanova at mastervoice.it
web: www.mastertraining.it - www.registroelettronico.com

Master Training S.r.l.
Sede Legale: via Timolini, 18 - Correggio (RE) - Italy
Sede Operativa: via Sani, 15 - Reggio Emilia - Italy
Sede Commerciale: via Sani, 9 - Reggio Emilia - Italy

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From msc at freeswitch.org  Tue Apr 19 20:36:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Apr 2011 09:36:44 -0700
Subject: [Freeswitch-users] Playing tones
In-Reply-To: <E1QC0Gp-0002ul-3e@mx5.orcon.net.nz>
References: <E1QC0Gp-0002ul-3e@mx5.orcon.net.nz>
Message-ID: <BANLkTinNX8LsrYeKvu0dWY6E9o8emGTsTQ@mail.gmail.com>

Clive,

Welcome to FreeSWITCH! If you're doing event socket stuff after only two
weeks then I must tip my hat to you. Well done!

I've never seen this symptom before so I'm at a bit of a loss as to what is
happening. Out of curiosity, what language are you using for your event
socket communications? I'm wondering if you can send the tone_stream without
the "loops=-1" over and over again and receive the same symptom where the
playback speeds up. The other test I would try is doing something like
"loops=25" or some relatively high number. Again see if the symptom presents
itself. (I'm wondering if this happens only on loops=-1 or if it happens any
time there is a loop.)

Let us know what you discover.

-MC

On Mon, Apr 18, 2011 at 7:01 PM, Clive Lansink <clive at lansink.co.nz> wrote:

> Hello
>
> I am relatively new to Freeswitch; I've played with it for about two weeks.
> It is now fully installed as our home PABX system. I've also done some
> elementary IVR stuff using an external script communicating to event_socket.
> It can monitor Freeswitch events, answer calls, play and record files and
> get digits so I suppose it does all the basics.
>
> But I'm having difficulty playing tones. A simple tone is no problem like
> the UK busy tone, %(400,200,400,450);%(400,2000,400,450), but things go
> wrong when I try loops.
>
> In my external script when I want to play a tone, I sende a sendmsg request
> with the appropriate UUID. In the headers, I have:
> call-command: execute
> execute-app-name: gentones
> execute-app-arg: %(400,200,400,450);%(400,2000,400,450)
> ]
> That works, but if I want the ring tone to repeat for an indefinite period,
> the header then becomes:
> execute-app-arg: %(400,200,400,450);%(400,2000,400,450);loops=-1
> When I do this, I hear one ring tone, then a single DTMF digit is played,
> presumably the 1 of loops=-1. I've also tried like loops=5 and got the same
> result.
>
> After seeing some examples, I then tried the following, again using the
> sendmsg command:
> call-command: execute
> execute-app-name: playback
> execute-app-arg:
> tone_stream://%(400,200,400,450);%(400,2000,400,450);loops=-1
> ]
> This time it does at least seem to loop indefinitely, but after the first
> two or so loops, the speed increases (not the pitch) so that the tone
> repeats much more quickly than it should.
>
> Any help would be much appreciated. It seems to me that using playback in
> this way must be accessing the same raw functionality as gentones, so I
> don't understand why the result would be different. But in the end I guess I
> don't care which method I use but it would be good to understand why both
> methods currently don't work for me.
>
> By the way, I am using a recent version of Freeswitch, a pre-compiled
> version running on Windows XP.
>
> Cheers.
>
>
> Clive Lansink
> Email: Clive at Lansink.Co.NZ
> Phone: +64 9 520-4242
> Mobile: +64 21 663-999
> Fax: +64 21 789-150
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 19 20:40:46 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Apr 2011 09:40:46 -0700
Subject: [Freeswitch-users] Originated session callback.
In-Reply-To: <BANLkTi=FOj_FrgTjRCGyZ-hPLM9yrdsbkA@mail.gmail.com>
References: <BANLkTinje48f3MSjB6bRWqpfgTi-xsxC8A@mail.gmail.com>
	<BANLkTik81bAyQiwxJym=KH3VdAD-oTPiqA@mail.gmail.com>
	<BANLkTi=JeAnNfvwrb-Y2OnGQjJRNjD-vHw@mail.gmail.com>
	<BANLkTik+HScKABdarAXUUJZUkHAcV1b0SQ@mail.gmail.com>
	<BANLkTi=FOj_FrgTjRCGyZ-hPLM9yrdsbkA@mail.gmail.com>
Message-ID: <BANLkTi=AKkoBFoLqwGB-GUqHdde19KDmrQ@mail.gmail.com>

So you want one script to control two different calls? I strongly recommend
that you use event socket and not a dialplan script.
-MC

On Tue, Apr 19, 2011 at 3:07 AM, Mohammed Habib <habib at alexcoder.com> wrote:

> I was hoping to make a scenario like this:
> User 1 calls and uses ASR to set some information (like the pizza demo),
> then User 2 is called and confirms User 1 using ASR too.
> So, I need to have both both sessions at the same time and both working
> with ASR and TTS.
>
> Thank you,
> Mohammed Habib
>
> On Mon, Apr 18, 2011 at 11:03 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> What are you trying to accomplish with this script? Why are you creating a
>> new session right in the middle of handling an existing session? I am
>> curious to know what problem you are attempting to solve.
>>
>> -MC
>>
>> On Mon, Apr 18, 2011 at 5:35 AM, Mohammed Habib <habib at alexcoder.com>wrote:
>>
>>> I need help getting events from originated session.
>>>
>>> This is my lua script:
>>>
>>> function onInput_MainSession(s, type, obj)
>>>    -- This one is working fine.
>>>    freeswitch.consoleLog("info", "Callback with type " .. type .. "\n");
>>> end
>>>
>>> function onInput_NewSession(s, type, obj)
>>>    -- This one is never called.
>>>    freeswitch.consoleLog("info", "Callback with type " .. type .. "\n");
>>> end
>>>
>>> session:answer();
>>> session:setInputCallback("onInput_MainSession");
>>> session:sleep(200);
>>> session:execute("detect_speech", "unimrcp testgrammer trestgrammer");
>>>
>>> newsession = freeswitch.Session("user/1002");
>>> newsession:setInputCallback("onInput_NewSession");
>>> newsession:sleep(200);
>>> newsession:execute("detect_speech", "unimrcp testgrammer trestgrammer");
>>>
>>> while ((session:ready() == true) ) and (newsession:ready() == true) do
>>>     -- Loop
>>>     sleep(200);
>>> end
>>>
>>> I am unable to capture any of the new session events or dtmf.
>>>
>>> Please help.
>>>
>>> Thank you,
>>> Mohammed Habib
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 19 20:47:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Apr 2011 09:47:40 -0700
Subject: [Freeswitch-users] No Audio on Gateway Incoming
In-Reply-To: <BANLkTikbW_g3nAJ_0ODxYohW-PCUvUFyZw@mail.gmail.com>
References: <BANLkTikbW_g3nAJ_0ODxYohW-PCUvUFyZw@mail.gmail.com>
Message-ID: <BANLkTimY_kNfFfSxzDs+t5aLbMzz5Vvp1Q@mail.gmail.com>

Yes, no audio isn't all that uncommon, especially if you have a NAT device
in the mix. The best thing to do is to get a console debug log with siptrace
and put it on pastebin, then put the link here. Be sure to use "FreeSWITCH
log" as the syntax highlighting style.

More information can be found here:
http://wiki.freeswitch.org/wiki/Reporting_Bugs

-MC

On Tue, Apr 19, 2011 at 6:56 AM, William Alianto <william at xofap.com> wrote:

> I have recently setup a freeswitch server and configure with gateway for
> outbound/inbound call. I have succesfully done outbound call without any
> problem, but whenever I do inbound call, the phone ring but no audio at all.
> Has anybody encounter such problem before?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jcasale at activenetwerx.com  Tue Apr 19 21:59:14 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Tue, 19 Apr 2011 17:59:14 +0000
Subject: [Freeswitch-users] Voicemail params
Message-ID: <assp.0090d8dda5.92097A6A775D5147B1078E3F15430B923CBA6C@prato.activenetwerx.local>

I am trying to change the domain of the vm email message. I added
<param name="vm-mailfrom" value="1000 at foo.com"/> the user
but it is not modifying it, anyone know what else I am missing?

It remains as the ip of the internal sofia domain.

Thanks,
jlc



From elijah at crankenstein.com  Tue Apr 19 22:49:30 2011
From: elijah at crankenstein.com (elijah)
Date: Tue, 19 Apr 2011 11:49:30 -0700
Subject: [Freeswitch-users] attended transfer to gateway
Message-ID: <BANLkTinB6uFAWDnq-Ot1tiSPi99pu0Oruw@mail.gmail.com>

Attended transfers to an external c-leg is breaking 3-way conference
functionality for me. Like this:

<action application="att_xfer"
data="sofia/gateway/onesource/${attxfer_callthis}"/>

The c-leg is established and hanging up the originator's leg will
successfully bridge the other 2. However, pressing 0 on the originator,
which should establish a 3-way conference, actually hangs up the c-leg. Is
there another method by which I should specify the outbound leg for
att_xfer?
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From eric at loopfx.com  Tue Apr 19 23:13:07 2011
From: eric at loopfx.com (Eric Beard)
Date: Tue, 19 Apr 2011 15:13:07 -0400
Subject: [Freeswitch-users] Success
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DF3F5B34@doctorbob.loop.local>

Hello,

I just wanted to pass on my thanks and compliments to the FreeSwitch team.  I have been working on a migration from a home grown sip server over to FreeSwitch during the last few weeks, and things are going very well so far.  I am up to originating 140 concurrent calls, and the server (a relatively modest 1U Dell box) is humming along at around 7% CPU.  Call quality is perfect.  I plan to transition all of my production machines over to FS in the near future.

Thanks everyone!

-----------------------
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 anthony.minessale at gmail.com  Tue Apr 19 23:14:58 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 19 Apr 2011 14:14:58 -0500
Subject: [Freeswitch-users] BUG FIX: "Buffer size sanity check failed!"
 drops FAX receiving unneeded
In-Reply-To: <BANLkTinAe5vrZDDHEueFVpaHJruzjUL20w@mail.gmail.com>
References: <4D998A5A.6080901@priv.de>
	<BANLkTimSg1hmwtCUwfJkA_DRAgULW32QBQ@mail.gmail.com>
	<BANLkTi=sZwvZw+TJV0W1tMJOc38ytv-ddw@mail.gmail.com>
	<BANLkTinAe5vrZDDHEueFVpaHJruzjUL20w@mail.gmail.com>
Message-ID: <BANLkTinLrtd5zxb7u-p9zyh+wZ3gQzRkcw@mail.gmail.com>

To all involved:

I had to manually remember this issue in my head (13 days old)
you provided no traces, I just added a workaround and committed it.

I very politely asked you to open a JIRA and offered free help.
If you ever do this again I will remove you from the list......



On Wed, Apr 6, 2011 at 8:30 PM, Anthony Minessale
<anthony.minessale at gmail.com> wrote:
> 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
>



-- 
Anthony Minessale II

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 rupa at rupa.com  Tue Apr 19 23:15:04 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Apr 2011 14:15:04 -0500
Subject: [Freeswitch-users] nibblebill cdr
In-Reply-To: <AANLkTikqy8O50YJZxDUBiE=qSjHZ7JvHOm5F6cZMOVvZ@mail.gmail.com>
References: <AANLkTikqy8O50YJZxDUBiE=qSjHZ7JvHOm5F6cZMOVvZ@mail.gmail.com>
Message-ID: <BANLkTikns0nwdia6QP9rTexrMpKOZ4z_7w@mail.gmail.com>

Read the full wiki page.  At the end is some comment about a channel
var that will carry over to your xml CDRs or can be added to the CSV
CDR template.

http://wiki.freeswitch.org/wiki/Mod_nibblebill#Other_Notes

On Tue, Mar 8, 2011 at 6:17 PM, budi wibowo <bwibowo at gmail.com> wrote:
> hi
> is there any possibility that mod_nibblebill create cdr with call pricing?
> right now i just see the balance deducted, in fs_cli console i can see the
> call info
>
> regards
> budi
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From rupa at rupa.com  Tue Apr 19 23:22:10 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Apr 2011 14:22:10 -0500
Subject: [Freeswitch-users] Port mapping disabled warning in Freeswitch
	log
In-Reply-To: <BANLkTi=SXobYbHp5ueY8qK+o1CrdMQnd7A@mail.gmail.com>
References: <BANLkTi=SXobYbHp5ueY8qK+o1CrdMQnd7A@mail.gmail.com>
Message-ID: <BANLkTinBmhahLkcbvctXtd2O5REX-BUkuQ@mail.gmail.com>

I lowered it to INFO

On Tue, Apr 19, 2011 at 11:04 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> If I set log level to warning, I see in the log many row as:
> ?[WARNING] switch_nat.c:589 NAT port mapping disabled
> My freeswitch is running as: "./freeswitch -nonat"
> because it has a public IP and all connected gateways have also a public IP.
> To avoid this warning I have removed the parameter "-nonat" but if I do
> "show nat_map" in fs_cli I see "0 total.".
> I have always started Freeswitch with "-nonat", any advise to solve that?
>
-- 
-Rupa


From rupa at rupa.com  Tue Apr 19 23:27:57 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Apr 2011 14:27:57 -0500
Subject: [Freeswitch-users] How does mod_lcr handle invalid ANIs?
In-Reply-To: <AANLkTin3TaT=fV_mJhq0Hq4_X+GFQ4aDT46hSJdiFNfH@mail.gmail.com>
References: <AANLkTin3TaT=fV_mJhq0Hq4_X+GFQ4aDT46hSJdiFNfH@mail.gmail.com>
Message-ID: <BANLkTi=tm6Qjjm1NCn2bhBAtuJo5F7XwQw@mail.gmail.com>

Did you try it out?  What behavior did you observe?

On Wed, Mar 9, 2011 at 10:19 AM, Max Clark <max.clark at gmail.com> wrote:
> Hello,
>
> I am calling the lcr module from the dialplan like this:
>
> <action application="lcr" data="$2 default ${caller_id_number} intrastate"/>
>
> So I can get interstate vs. intrastate routing (I dropped the
> intralata column from the database). How does mod_lcr handle an
> invalid ANI? Which rate is used to route for this condition?
>
> 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
>



-- 
-Rupa


From msc at freeswitch.org  Tue Apr 19 23:40:52 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Apr 2011 12:40:52 -0700
Subject: [Freeswitch-users] Success
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DF3F5B34@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DF3F5B34@doctorbob.loop.local>
Message-ID: <BANLkTi=BfoHRxF8=amdjAsKSbgJ-k3-F1Q@mail.gmail.com>

Thanks for letting us know how it's going. Nice job on learning FS in just a
few weeks. Keep up the good work and don't forget to register for ClueCon
right away!

-MC

On Tue, Apr 19, 2011 at 12:13 PM, Eric Beard <eric at loopfx.com> wrote:

> Hello,
>
>
>
> I just wanted to pass on my thanks and compliments to the FreeSwitch team.
> I have been working on a migration from a home grown sip server over to
> FreeSwitch during the last few weeks, and things are going very well so
> far.  I am up to originating 140 concurrent calls, and the server (a
> relatively modest 1U Dell box) is humming along at around 7% CPU.  Call
> quality is perfect.  I plan to transition all of my production machines over
> to FS in the near future.
>
>
>
> Thanks everyone!
>
>
>
> -----------------------
>
> *Eric Z. Beard, CTO*
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> 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
>
>
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From michel at arneill-py.sacramento.ca.us  Tue Apr 19 23:39:26 2011
From: michel at arneill-py.sacramento.ca.us (Michel Py)
Date: Tue, 19 Apr 2011 12:39:26 -0700
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
	with productivity license.
Message-ID: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local>

Hi guys,
 
I'm reading the following page:
http://wiki.freeswitch.org/wiki/Polycom_Presence_Setup <http://wiki.freeswitch.org/wiki/Polycom_Presence_Setup> 
and I'm trying to make it work the "attendant" way.
 
The phone is a 601 with 3.1.7.0134. I think the productivity license is installed correctly, both because I see the phone loading the file in the logs and that I enabled corporate directory in sip_317.cfg and I see it in menu - settings - basic - preferences.
 
I made the mac-address-settings.cfg file as outlined, and none of the attendant things even displays on the phone.
 
What am I missing there? Where do I start the troubleshooting?
 
Thanks
Michel.
 
 
 
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From msc at freeswitch.org  Tue Apr 19 23:39:16 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Apr 2011 12:39:16 -0700
Subject: [Freeswitch-users] attended transfer to gateway
In-Reply-To: <BANLkTinB6uFAWDnq-Ot1tiSPi99pu0Oruw@mail.gmail.com>
References: <BANLkTinB6uFAWDnq-Ot1tiSPi99pu0Oruw@mail.gmail.com>
Message-ID: <BANLkTikkF95DCGgpQJ-kKCR-U_=X7fF5+g@mail.gmail.com>

Get a console log with full siptrace (on both internal and external
profiles) and drop it into pastebin. Be sure to specific "FreeSWITCH log"
for syntax highlighting to make it easier to read. Send the pb link here.
Hopefully you or another user will be able to parse the logs to see what is
happening.

-MC

On Tue, Apr 19, 2011 at 11:49 AM, elijah <elijah at crankenstein.com> wrote:

> Attended transfers to an external c-leg is breaking 3-way conference
> functionality for me. Like this:
>
> <action application="att_xfer"
> data="sofia/gateway/onesource/${attxfer_callthis}"/>
>
> The c-leg is established and hanging up the originator's leg will
> successfully bridge the other 2. However, pressing 0 on the originator,
> which should establish a 3-way conference, actually hangs up the c-leg. Is
> there another method by which I should specify the outbound leg for
> att_xfer?
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 rupa at rupa.com  Wed Apr 20 00:09:21 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Apr 2011 15:09:21 -0500
Subject: [Freeswitch-users] [freeswitch] <defunct> when using cmd="exec"
 in freeswitch.xml
In-Reply-To: <4D7F251F.2070101@ewetel.de>
References: <4D78A5AF.1000808@ewetel.de>
	<AANLkTin7EUJpz4=9gxp3sTXykjybH5B4Uer7noD+Bgct@mail.gmail.com>
	<4D7F251F.2070101@ewetel.de>
Message-ID: <BANLkTi=+i=Ew=Ant8WOgxd4hC_kDxmKetw@mail.gmail.com>

The child process needs to be "reaped" in order to get out of
<defunct> and of course it'll be cleared out when the parent process
finally exits.

<looking at code>

I don't see any reason why it would leave a zombie.  The code
correctly calls waitpid() after the fork in the parent code.

Oh, looks like brian added waitpid on :

dae2cb4a (Brian West             2011-03-29 18:05:05 -0500 1246)

So update and it should work.

On Tue, Mar 15, 2011 at 3:36 AM, Mitja Thomas <mitja.thomas1 at ewetel.de> wrote:
> Hi Micheal,
>
> yeah I think your workaround will work fine and Ill change it that way. But
> this issue might still be a missbehaviour. Either by fs in which case I
> think it might be an interesting information for the Developers OR a wrong
> usage by me in which case this is an interesting information for me :)
>
> Thanks,
> Mitja
>
> IIRC, 'fork' creates a child PID that stays around and waits for the parent
> PID to die. (Don't quote me on that - wait for an expert to chime in.)
> An alternative method would be to have a FS start script that launches your
> make_my_vars.sh script and then launches FS with the appropriate cmd line
> args. You could then drop the 'exec' cmd and then just keep the include
> cmd.
> -MC
>
> On Thu, Mar 10, 2011 at 2:19 AM, Mitja Thomas <mitja.thomas1 at ewetel.de>
> wrote:
>>
>> Hi there,
>>
>> we tried to set up the FreeSWITCH and other Applications, so that we can
>> configure them easier and more centralised.
>> Thus we defined some Environment Variables (using CentOS) which hold often
>> used Configuration Parameter like MySQL IP or FS Event Socket IP.
>> We tried to integrate these Env Variables into the FS conf files by
>> executing a shell Skript in freeswitch.xml via cmd="exec" which prepares an
>> conf file which we include into freeswitch.xml:
>>
>> ? <X-PRE-PROCESS cmd="exec" data="sh conf/make_my_vars.sh"/>
>> ? <X-PRE-PROCESS cmd="include" data="my_vars.xml"/>
>>
>> This works as expected and the pre defined variables in my_vars can be
>> accessed from the other config Files, except that when we start our
>> FreeSWITCH? a zombie child process is spawned.
>>
>> # ps -eaf | grep free
>> ippbx??? 22191 22190? 4 09:41 pts/1??? 00:00:01
>> /opt/app/voip/ippbx/bin/freeswitch -waste -nonat -hp
>> ippbx??? 22197 22191? 0 09:41 pts/1??? 00:00:00 [freeswitch] <defunct>
>>
>> What I wanna know is: Is this a FS missbehaviour or do we use this in a
>> wrong way?
>>
>> make_my_vars.sh:
>> F="conf/my_vars.xml"
>> echo "<!-- Auto generated file for getting *NIX evironment variables -->"
>> > $F
>> echo "<include>" >> $F
>>
>> fs_ip=`printenv MY_FS_IP`
>> if test -n "$fs_ip"
>> then
>> ??????? echo '<X-PRE-PROCESS cmd="set" data="my_fs_ip='$fs_ip'"/>' >> $F
>> fi
>> ...
>> echo "</include>" >> $F
>>
>> my_vars.xml (after FS startup):
>> <!-- Auto generated file for getting *NIX evironment variables -->
>> <include>
>> <X-PRE-PROCESS cmd="set" data="my_fs_ip=***.***.***.***"/>
>> ...
>> </include>
>>
>> Regards
>> Mitja
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From rupa at rupa.com  Wed Apr 20 00:31:13 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Apr 2011 15:31:13 -0500
Subject: [Freeswitch-users] Outbound caller id for a specific gateway?
In-Reply-To: <AANLkTimnorNHZa_3KqeqgecKjgHo4PLQWsm8+xH_FJMQ@mail.gmail.com>
References: <AANLkTin=M3MJKB5QA86iL+hkhkZncTQ8eiKQJM2t6ap5@mail.gmail.com>
	<AANLkTimqg7TRBULWR2__kxu32K4QuHQi1ChgsAZGiW+G@mail.gmail.com>
	<AANLkTimnorNHZa_3KqeqgecKjgHo4PLQWsm8+xH_FJMQ@mail.gmail.com>
Message-ID: <BANLkTimiyfg=AJefCzOoW6uHEHJ7WCzP9A@mail.gmail.com>

On Tue, Mar 22, 2011 at 2:49 AM, Dmitry Bely <dmitry.bely at gmail.com> wrote:
> On Tue, Mar 22, 2011 at 8:02 AM, Michael Collins <msc at freeswitch.org> wrote:
>> How about set the caller id in vars.xml:
>> <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
>> Set it to the most common value and then you only have to do something in
>> bridges and originates that need a CID different from what you set in
>> vars.xml...
>
> I have a number of gateways that require different CIDs. So there is
> no "most common" value. I have managed to make a proper dialplan but
> originate is still tedious... I just wonder if there is more clean way
> (something like caller-id-in-from gateway parameter)

mod_lcr can manipulate your CID on a per route (gateway) basis.  It
may be overkill for just playing with CID but at some point you are
going to want to conditionally route based on rates and ... well, then
you already have mod_lcr setup and configured. :)


-- 
-Rupa


From wstephen80 at gmail.com  Wed Apr 20 00:31:52 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Tue, 19 Apr 2011 22:31:52 +0200
Subject: [Freeswitch-users] Port mapping disabled warning in Freeswitch
	log
In-Reply-To: <BANLkTinBmhahLkcbvctXtd2O5REX-BUkuQ@mail.gmail.com>
References: <BANLkTi=SXobYbHp5ueY8qK+o1CrdMQnd7A@mail.gmail.com>
	<BANLkTinBmhahLkcbvctXtd2O5REX-BUkuQ@mail.gmail.com>
Message-ID: <BANLkTim0k45KkwcwLukWW55G7+LLaFuGvQ@mail.gmail.com>

Ok, thank you Rupa.
My doubt was: why FS try to add a nat port mapping when there is no nat?
But if this is only a log problem, then ok, I'll do a git pull.

Stephen

On Tue, Apr 19, 2011 at 9:22 PM, Rupa Schomaker <rupa at rupa.com> wrote:

> I lowered it to INFO
>
> On Tue, Apr 19, 2011 at 11:04 AM, Stephen Wilde <wstephen80 at gmail.com>
> wrote:
> > If I set log level to warning, I see in the log many row as:
> >  [WARNING] switch_nat.c:589 NAT port mapping disabled
> > My freeswitch is running as: "./freeswitch -nonat"
> > because it has a public IP and all connected gateways have also a public
> IP.
> > To avoid this warning I have removed the parameter "-nonat" but if I do
> > "show nat_map" in fs_cli I see "0 total.".
> > I have always started Freeswitch with "-nonat", any advise to solve that?
> >
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From clive at lansink.co.nz  Wed Apr 20 00:09:24 2011
From: clive at lansink.co.nz (Clive Lansink)
Date: Wed, 20 Apr 2011 08:09:24 +1200
Subject: [Freeswitch-users] Playing tones
Message-ID: <E1QCHF0-0008Vz-JX@mx6.orcon.net.nz>

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Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20110420/e0560e45/attachment.pl 

From rupa at rupa.com  Wed Apr 20 00:58:13 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Apr 2011 15:58:13 -0500
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
Message-ID: <BANLkTi=n4WBCyXu4NkHPm-umdV3qxdNPDQ@mail.gmail.com>

On Fri, Mar 4, 2011 at 11:44 AM, David Villasmil
<david.villasmil.work at gmail.com> wrote:

> the prepaid side is done with nibble, yes. But i don't like it too much, i
> might just deduct the balance when the call disconnects and let the
> authorization block new calls, so the balance might go under 0 a little...

I'm curious -- what are the issues with nibblebill?  Is it something
we can improve or something more fundamental?

-- 
-Rupa


From rhuddleston at gmail.com  Wed Apr 20 01:03:55 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Tue, 19 Apr 2011 17:03:55 -0400
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <BANLkTi=n4WBCyXu4NkHPm-umdV3qxdNPDQ@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=n4WBCyXu4NkHPm-umdV3qxdNPDQ@mail.gmail.com>
Message-ID: <0e9901cbfed5$4bdc9eb0$e395dc10$@com>

Ditto here too

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
Schomaker
Sent: Tuesday, April 19, 2011 4:58 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Complete wholesale app in freeswitch

On Fri, Mar 4, 2011 at 11:44 AM, David Villasmil
<david.villasmil.work at gmail.com> wrote:

> the prepaid side is done with nibble, yes. But i don't like it too much, i
> might just deduct the balance when the call disconnects and let the
> authorization block new calls, so the balance might go under 0 a little...

I'm curious -- what are the issues with nibblebill?  Is it something
we can improve or something more fundamental?

-- 
-Rupa

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



From anton.vazir at gmail.com  Wed Apr 20 01:16:55 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 20 Apr 2011 02:16:55 +0500
Subject: [Freeswitch-users] Download link to Skyopen
In-Reply-To: <4D906E17.4000609@greatiam.com>
References: <4D906E17.4000609@greatiam.com>
Message-ID: <BANLkTi=Eik2MjC+4ouQz9wmtf7E1MufRwQ@mail.gmail.com>

Seems this topic is either forbidden or not at anyones interest.
Personally, I have only found 2.0.72 as mentioned verison in wiki and
emails and 2.0.0.72 as available in the internet. At the same time no
proof that there is a mistake in WIKI for the version number mentioned
as 2.0.72 instead 2.0.0.72 or proof of it's correctness...


From covici at ccs.covici.com  Wed Apr 20 01:32:53 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 19 Apr 2011 17:32:53 -0400
Subject: [Freeswitch-users] mod_cepstral not working
Message-ID: <8977.1303248773@ccs.covici.com>

Hi.  I am having trouble compiling mod_cepstral after moving to a 64-bit
operating system and libraries, although maybe this has nothing to do
with the problem.

When the module tries to load, I get 
2011-04-19 17:31:12.145808 [CRIT] switch_loadable_module.c:928 Error
Loading module /usr/local/freeswitch/mod/mod_cepstral.so
**/usr/local/freeswitch/mod/mod_cepstral.so: undefined symbol:
swift_port_set_voice_by_name**

Now swift does work by itself, so I am not sure what is happening here.

Thanks in advance for any assistance.

-- 
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 rupa at rupa.com  Wed Apr 20 02:03:42 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Apr 2011 17:03:42 -0500
Subject: [Freeswitch-users] Port mapping disabled warning in Freeswitch
	log
In-Reply-To: <BANLkTim0k45KkwcwLukWW55G7+LLaFuGvQ@mail.gmail.com>
References: <BANLkTi=SXobYbHp5ueY8qK+o1CrdMQnd7A@mail.gmail.com>
	<BANLkTinBmhahLkcbvctXtd2O5REX-BUkuQ@mail.gmail.com>
	<BANLkTim0k45KkwcwLukWW55G7+LLaFuGvQ@mail.gmail.com>
Message-ID: <BANLkTikDPbL0LWpKfyY=mj3rDNZ0GN0J=Q@mail.gmail.com>

On Tue, Apr 19, 2011 at 3:31 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:
> Ok, thank you Rupa.
> My doubt was: why FS try to add a nat port mapping when there is no nat?
> But if this is only a log problem, then ok, I'll do a git pull.
> Stephen
>

The log is informational.  I guess you could call it an impedance
mismatch between the caller and the callee. The function that calls
the mapping function doesn't know or care if nat is setup.  The called
mapping function logs a warning (now info) saying "hey dummy, nat
isn't setup so I'm not going to do a nat map".

So lowering to info is the easiest, maybe even debug is more
appropriate.  I can be easily convinced of that.

-- 
-Rupa


From avi at avimarcus.net  Wed Apr 20 04:47:41 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 20 Apr 2011 03:47:41 +0300
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <0e9901cbfed5$4bdc9eb0$e395dc10$@com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=n4WBCyXu4NkHPm-umdV3qxdNPDQ@mail.gmail.com>
	<0e9901cbfed5$4bdc9eb0$e395dc10$@com>
Message-ID: <BANLkTinPNrXFo1mCXvicgisxj2BWBgseog@mail.gmail.com>

I found 3 things:
1) double-bill bug on the new version:
http://jira.freeswitch.org/browse/FS-3153 (maybe older version too..?)
2) 0 balance ended up locking the gateway with stuck channels, I never
figured out how to fix that
3) Lots of sql queries, or maybe I was just bothered by SQL spam in the log?
and I'm barely running any concurrent calls, but it seems nibble makes a LOT
of sql queries. Extra selects at the start and end of the call, I think.

-Avi

On Wed, Apr 20, 2011 at 12:03 AM, Robert Huddleston
<rhuddleston at gmail.com>wrote:

> Ditto here too
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
> Schomaker
> Sent: Tuesday, April 19, 2011 4:58 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Complete wholesale app in freeswitch
>
> On Fri, Mar 4, 2011 at 11:44 AM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
>
> > the prepaid side is done with nibble, yes. But i don't like it too much,
> i
> > might just deduct the balance when the call disconnects and let the
> > authorization block new calls, so the balance might go under 0 a
> little...
>
> I'm curious -- what are the issues with nibblebill?  Is it something
> we can improve or something more fundamental?
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From daletrub at gmail.com  Wed Apr 20 05:24:53 2011
From: daletrub at gmail.com (Dale Trub)
Date: Tue, 19 Apr 2011 18:24:53 -0700
Subject: [Freeswitch-users] conference play / stop / PAUSE/RESUME?
In-Reply-To: <BANLkTin=Ps=ri8mVyH0HTgFe2pB4oD_Sxg@mail.gmail.com>
References: <BANLkTi=9672+WLZON0teCJLa-CFsg4E88w@mail.gmail.com>
	<BANLkTimgE00BQ9AC9O8jOtBtKz3Mv4je=Q@mail.gmail.com>
	<BANLkTin=Ps=ri8mVyH0HTgFe2pB4oD_Sxg@mail.gmail.com>
Message-ID: <BANLkTik2nQ5S3sS6Zmnu81yQiARFeJPRMw@mail.gmail.com>

Wow that's fantastic.

> seek:<+[samples]>|<-[samples]>

What are the parameters for samples?

>Many thanks,
> MC

Thanks to you and and Anthony!  (And fine: Brian too ... :-)

-Dale

On Mon, Apr 18, 2011 at 1:04 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Apr 18, 2011 at 9:30 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> if you need this level of granularity, I suggest instead of the
>> conference play stuff that is built in, that you use the originate
>> command to call a new channel into the conference to play the file
>> then use the uuid_fileman api command to manipulate the playback.
>>
>
> FYI,
> I just added the basic uuid_fileman stuff to the wiki:
> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman
>
> I don't have actual examples yet - I just combed through the source code to
> pull out the various commands that are available. For those who like to dig
> into the source code: uuid_fileman in mod_commands.c is essentially a
> wrapper/helper that calls switch_ivr_process_fh() from switch_ivr.c in the
> core. The commands are all in that function. I was able to deduce most of
> the args and their meanings by looking at the source code. If you have used
> these functions and can contribute some simple examples that would be very
> helpful.
>
> Many thanks,
> 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 daletrub at gmail.com  Wed Apr 20 05:26:54 2011
From: daletrub at gmail.com (Dale Trub)
Date: Tue, 19 Apr 2011 18:26:54 -0700
Subject: [Freeswitch-users] conference play / stop / PAUSE/RESUME?
In-Reply-To: <BANLkTik2nQ5S3sS6Zmnu81yQiARFeJPRMw@mail.gmail.com>
References: <BANLkTi=9672+WLZON0teCJLa-CFsg4E88w@mail.gmail.com>
	<BANLkTimgE00BQ9AC9O8jOtBtKz3Mv4je=Q@mail.gmail.com>
	<BANLkTin=Ps=ri8mVyH0HTgFe2pB4oD_Sxg@mail.gmail.com>
	<BANLkTik2nQ5S3sS6Zmnu81yQiARFeJPRMw@mail.gmail.com>
Message-ID: <BANLkTimmux47TbH2eDc3gByR9jEeCVD8Fw@mail.gmail.com>

ps. I'll update the wiki with your answer ...

On Tue, Apr 19, 2011 at 6:24 PM, Dale Trub <daletrub at gmail.com> wrote:

> Wow that's fantastic.
>
> > seek:<+[samples]>|<-[samples]>
>
> What are the parameters for samples?
>
> >Many thanks,
> > MC
>
> Thanks to you and and Anthony!  (And fine: Brian too ... :-)
>
> -Dale
>
> On Mon, Apr 18, 2011 at 1:04 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Apr 18, 2011 at 9:30 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> if you need this level of granularity, I suggest instead of the
>>> conference play stuff that is built in, that you use the originate
>>> command to call a new channel into the conference to play the file
>>> then use the uuid_fileman api command to manipulate the playback.
>>>
>>
>> FYI,
>> I just added the basic uuid_fileman stuff to the wiki:
>> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_fileman
>>
>> I don't have actual examples yet - I just combed through the source code
>> to pull out the various commands that are available. For those who like to
>> dig into the source code: uuid_fileman in mod_commands.c is essentially a
>> wrapper/helper that calls switch_ivr_process_fh() from switch_ivr.c in the
>> core. The commands are all in that function. I was able to deduce most of
>> the args and their meanings by looking at the source code. If you have used
>> these functions and can contribute some simple examples that would be very
>> helpful.
>>
>> Many thanks,
>> 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 rhuddleston at gmail.com  Wed Apr 20 05:55:32 2011
From: rhuddleston at gmail.com (rhuddleston at gmail.com)
Date: Tue, 19 Apr 2011 21:55:32 -0400
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <BANLkTinPNrXFo1mCXvicgisxj2BWBgseog@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=n4WBCyXu4NkHPm-umdV3qxdNPDQ@mail.gmail.com>
	<0e9901cbfed5$4bdc9eb0$e395dc10$@com>
	<BANLkTinPNrXFo1mCXvicgisxj2BWBgseog@mail.gmail.com>
Message-ID: <6A44240C-064C-4D9A-8189-0CA212B86E8B@gmail.com>

I was planning on using it production w/ mod_lcr but found that mod_lcr doesnt give me what I need completely. Considering writing my own wholesale.
I was considering hacking A2Billing but if Im going to spend effort then id rather write my own

On Apr 19, 2011, at 8:47 PM, Avi Marcus <avi at avimarcus.net> wrote:

> I found 3 things:
> 1) double-bill bug on the new version: http://jira.freeswitch.org/browse/FS-3153 (maybe older version too..?)
> 2) 0 balance ended up locking the gateway with stuck channels, I never figured out how to fix that
> 3) Lots of sql queries, or maybe I was just bothered by SQL spam in the log? and I'm barely running any concurrent calls, but it seems nibble makes a LOT of sql queries. Extra selects at the start and end of the call, I think.
> 
> -Avi
> 
> On Wed, Apr 20, 2011 at 12:03 AM, Robert Huddleston <rhuddleston at gmail.com> wrote:
> Ditto here too
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
> Schomaker
> Sent: Tuesday, April 19, 2011 4:58 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Complete wholesale app in freeswitch
> 
> On Fri, Mar 4, 2011 at 11:44 AM, David Villasmil
> <david.villasmil.work at gmail.com> wrote:
> 
> > the prepaid side is done with nibble, yes. But i don't like it too much, i
> > might just deduct the balance when the call disconnects and let the
> > authorization block new calls, so the balance might go under 0 a little...
> 
> I'm curious -- what are the issues with nibblebill?  Is it something
> we can improve or something more fundamental?
> 
> --
> -Rupa
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From rupa at rupa.com  Wed Apr 20 06:05:21 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Apr 2011 21:05:21 -0500
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <BANLkTinPNrXFo1mCXvicgisxj2BWBgseog@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=n4WBCyXu4NkHPm-umdV3qxdNPDQ@mail.gmail.com>
	<0e9901cbfed5$4bdc9eb0$e395dc10$@com>
	<BANLkTinPNrXFo1mCXvicgisxj2BWBgseog@mail.gmail.com>
Message-ID: <BANLkTinRg49bqSdu=RXLtbYzsZOeCzNF2A@mail.gmail.com>

1) I'll see if I can look into this when I get a chance.  I added
myself as a watcher and if darren doesn't get to it i'll look at it.

2) I've never seen this.  but I don't run significant enough volume to
notice maybe?  Was it reproducible?

3) This is a matter of tuning the nibble bill rate.  There is a double
sql update at the end of a call that is there due to a fix I put in at
one point.  I can look at removing the hangup hook at just work in
cs_reporting.  Though honestly I can't imagine that the second update
really is hurting hte sql server.  It is (generally) an exact
duplicate or a very small offset calculation.

Ultimately there are many ways to skin the cat and if you've found a
better way for you thats fine.  But I want to make sure that any
glaring issues are addressed.  Thanks for the feedback.

On Tue, Apr 19, 2011 at 7:47 PM, Avi Marcus <avi at avimarcus.net> wrote:
> I found 3 things:
> 1) double-bill bug on the new version:
> http://jira.freeswitch.org/browse/FS-3153 (maybe older version too..?)
> 2) 0 balance ended up locking the gateway with stuck channels, I never
> figured out how to fix that
> 3) Lots of sql queries, or maybe I was just bothered by SQL spam in the log?
> and I'm barely running any concurrent calls, but it seems nibble makes a LOT
> of sql queries. Extra selects at the start and end of the call, I think.
> -Avi
>
> On Wed, Apr 20, 2011 at 12:03 AM, Robert Huddleston <rhuddleston at gmail.com>
> wrote:
>>
>> Ditto here too
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
>> Schomaker
>> Sent: Tuesday, April 19, 2011 4:58 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Complete wholesale app in freeswitch
>>
>> On Fri, Mar 4, 2011 at 11:44 AM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>>
>> > the prepaid side is done with nibble, yes. But i don't like it too much,
>> > i
>> > might just deduct the balance when the call disconnects and let the
>> > authorization block new calls, so the balance might go under 0 a
>> > little...
>>
>> I'm curious -- what are the issues with nibblebill? ?Is it something
>> we can improve or something more fundamental?
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From rupa at rupa.com  Wed Apr 20 06:05:48 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Tue, 19 Apr 2011 21:05:48 -0500
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <6A44240C-064C-4D9A-8189-0CA212B86E8B@gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=n4WBCyXu4NkHPm-umdV3qxdNPDQ@mail.gmail.com>
	<0e9901cbfed5$4bdc9eb0$e395dc10$@com>
	<BANLkTinPNrXFo1mCXvicgisxj2BWBgseog@mail.gmail.com>
	<6A44240C-064C-4D9A-8189-0CA212B86E8B@gmail.com>
Message-ID: <BANLkTimPaXNuKGjd5FE+e7gRrdm=W8Et+A@mail.gmail.com>

What additional functionality did you need?

On Tue, Apr 19, 2011 at 8:55 PM,  <rhuddleston at gmail.com> wrote:
> I was planning on using it production w/ mod_lcr but found that mod_lcr
> doesnt give me what I need completely. Considering writing my own wholesale.
> I was considering hacking A2Billing but if Im going to spend effort then id
> rather write my own
>
> On Apr 19, 2011, at 8:47 PM, Avi Marcus <avi at avimarcus.net> wrote:
>
> I found 3 things:
> 1) double-bill bug on the new version:
> http://jira.freeswitch.org/browse/FS-3153 (maybe older version too..?)
> 2) 0 balance ended up locking the gateway with stuck channels, I never
> figured out how to fix that
> 3) Lots of sql queries, or maybe I was just bothered by SQL spam in the log?
> and I'm barely running any concurrent calls, but it seems nibble makes a LOT
> of sql queries. Extra selects at the start and end of the call, I think.
> -Avi
>
> On Wed, Apr 20, 2011 at 12:03 AM, Robert Huddleston <rhuddleston at gmail.com>
> wrote:
>>
>> Ditto here too
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa
>> Schomaker
>> Sent: Tuesday, April 19, 2011 4:58 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] Complete wholesale app in freeswitch
>>
>> On Fri, Mar 4, 2011 at 11:44 AM, David Villasmil
>> <david.villasmil.work at gmail.com> wrote:
>>
>> > the prepaid side is done with nibble, yes. But i don't like it too much,
>> > i
>> > might just deduct the balance when the call disconnects and let the
>> > authorization block new calls, so the balance might go under 0 a
>> > little...
>>
>> I'm curious -- what are the issues with nibblebill? ?Is it something
>> we can improve or something more fundamental?
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From haloha201 at gmail.com  Wed Apr 20 04:58:36 2011
From: haloha201 at gmail.com (haloha)
Date: Wed, 20 Apr 2011 07:58:36 +0700
Subject: [Freeswitch-users] Download link to Skyopen
In-Reply-To: <BANLkTi=Eik2MjC+4ouQz9wmtf7E1MufRwQ@mail.gmail.com>
References: <4D906E17.4000609@greatiam.com>
	<BANLkTi=Eik2MjC+4ouQz9wmtf7E1MufRwQ@mail.gmail.com>
Message-ID: <BANLkTinhHimVxS5nSr9e+8Pa7nztssLZUw@mail.gmail.com>

Hi

try this link if it work for you
http://www.fshare.vn/file/883OBXSPUR/


hope it help

Ha`

On 4/20/11, Anton VG <anton.vazir at gmail.com> wrote:
> Seems this topic is either forbidden or not at anyones interest.
> Personally, I have only found 2.0.72 as mentioned verison in wiki and
> emails and 2.0.0.72 as available in the internet. At the same time no
> proof that there is a mistake in WIKI for the version number mentioned
> as 2.0.72 instead 2.0.0.72 or proof of it's correctness...
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From msc at freeswitch.org  Wed Apr 20 06:17:01 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 19 Apr 2011 19:17:01 -0700
Subject: [Freeswitch-users] mod_cepstral not working
In-Reply-To: <8977.1303248773@ccs.covici.com>
References: <8977.1303248773@ccs.covici.com>
Message-ID: <BANLkTinzpiYVtcAkeJeHV0QGx_VGE80ymA@mail.gmail.com>

Go ahead and open a Jira on this one.
-MC

On Tue, Apr 19, 2011 at 2:32 PM, <covici at ccs.covici.com> wrote:

> Hi.  I am having trouble compiling mod_cepstral after moving to a 64-bit
> operating system and libraries, although maybe this has nothing to do
> with the problem.
>
> When the module tries to load, I get
> 2011-04-19 17:31:12.145808 [CRIT] switch_loadable_module.c:928 Error
> Loading module /usr/local/freeswitch/mod/mod_cepstral.so
> **/usr/local/freeswitch/mod/mod_cepstral.so: undefined symbol:
> swift_port_set_voice_by_name**
>
> Now swift does work by itself, so I am not sure what is happening here.
>
> Thanks in advance for any assistance.
>
> --
> 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
>
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From yehavi.bourvine at gmail.com  Wed Apr 20 07:04:36 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 20 Apr 2011 06:04:36 +0300
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
 with productivity license.
In-Reply-To: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local>
References: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local>
Message-ID: <BANLkTimUkXz84DHh6bac4ognmYL1WzqdiQ@mail.gmail.com>

Hello Michel,

  Version 3.1.7 is very old. Try the 3.3.0 version (3.3.1F loses BLF after a
while).

                       __Yehavi:

 2011/4/19 Michel Py <michel at arneill-py.sacramento.ca.us>

>  Hi guys,
>
> I'm reading the following page:
> http://wiki.freeswitch.org/wiki/Polycom_Presence_Setup
> and I'm trying to make it work the "attendant" way.
>
> The phone is a 601 with 3.1.7.0134. I think the productivity license is
> installed correctly, both because I see the phone loading the file in the
> logs and that I enabled corporate directory in sip_317.cfg and I see it in
> menu - settings - basic - preferences.
>
> I made the mac-address-settings.cfg file as outlined, and none of the
> attendant things even displays on the phone.
>
> What am I missing there? Where do I start the troubleshooting?
>
> Thanks
> Michel.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 covici at ccs.covici.com  Wed Apr 20 08:19:57 2011
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 20 Apr 2011 00:19:57 -0400
Subject: [Freeswitch-users] mod_cepstral not working
In-Reply-To: <BANLkTinzpiYVtcAkeJeHV0QGx_VGE80ymA@mail.gmail.com>
References: <8977.1303248773@ccs.covici.com>
	<BANLkTinzpiYVtcAkeJeHV0QGx_VGE80ymA@mail.gmail.com>
Message-ID: <31139.1303273197@ccs.covici.com>

OK, the bug number is 3255.

Thanks.

Michael Collins <msc at freeswitch.org> wrote:

> Go ahead and open a Jira on this one.
> -MC
> 
> On Tue, Apr 19, 2011 at 2:32 PM, <covici at ccs.covici.com> wrote:
> 
> > Hi.  I am having trouble compiling mod_cepstral after moving to a 64-bit
> > operating system and libraries, although maybe this has nothing to do
> > with the problem.
> >
> > When the module tries to load, I get
> > 2011-04-19 17:31:12.145808 [CRIT] switch_loadable_module.c:928 Error
> > Loading module /usr/local/freeswitch/mod/mod_cepstral.so
> > **/usr/local/freeswitch/mod/mod_cepstral.so: undefined symbol:
> > swift_port_set_voice_by_name**
> >
> > Now swift does work by itself, so I am not sure what is happening here.
> >
> > Thanks in advance for any assistance.
> >
> > --
> > 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
> >
> 
> ----------------------------------------------------
> 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 dome at tel.co.th  Wed Apr 20 08:24:41 2011
From: dome at tel.co.th (dome at tel.co.th)
Date: Wed, 20 Apr 2011 11:24:41 +0700
Subject: [Freeswitch-users] SBC for carrier
Message-ID: <BANLkTinannjPrtj9oX=R7ugRydrJ-m6nfA@mail.gmail.com>

Dear all,
    I'm freeswitch user :)  but looking for carrier grade SBC for my
customer. like a acme , nexttone. they talking about 120,000
concurrent. so if someone know about this please send me directly.

Thanks.

Dome C.


From andrew.keil at askinteractive.net  Wed Apr 20 09:10:37 2011
From: andrew.keil at askinteractive.net (Andrew Keil)
Date: Wed, 20 Apr 2011 15:10:37 +1000
Subject: [Freeswitch-users] Where is mod_spidermonkey_etpan?
Message-ID: <C42A22554C0CF44DAC0AA1F132B86DE657F7EA38F3@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 attempting to send an e-mail from inside a javascript file.  Looking on your website I found the example: http://wiki.freeswitch.org/wiki/Examples_answermachine

This uses: use("etpan");


With the comment: Note: this script uses mod_spidermonkey_etpan which must be compiled and loaded.

My question is where is mod_spidermonkey_etpan ??  I cannot find it anywhere inside the freeswitch directories.

If there is no solution, is there another way you can recommend to send an e-mail from from within a javascript file?  Or even from an XML Dialplan would be OK.

Looking forward to your response.

Andrew Keil

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From gmaruzz at gmail.com  Wed Apr 20 09:23:04 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Wed, 20 Apr 2011 07:23:04 +0200
Subject: [Freeswitch-users] Download link to Skyopen
In-Reply-To: <BANLkTinhHimVxS5nSr9e+8Pa7nztssLZUw@mail.gmail.com>
References: <4D906E17.4000609@greatiam.com>
	<BANLkTi=Eik2MjC+4ouQz9wmtf7E1MufRwQ@mail.gmail.com>
	<BANLkTinhHimVxS5nSr9e+8Pa7nztssLZUw@mail.gmail.com>
Message-ID: <BANLkTikysBSN5ccqUUJCXOYY_rw2iZTyXw@mail.gmail.com>

2.0.0.72
wikipage fixed

On Wed, Apr 20, 2011 at 2:58 AM, haloha <haloha201 at gmail.com> wrote:
> Hi
>
> try this link if it work for you
> http://www.fshare.vn/file/883OBXSPUR/
>
>
> hope it help
>
> Ha`
>
> On 4/20/11, Anton VG <anton.vazir at gmail.com> wrote:
>> Seems this topic is either forbidden or not at anyones interest.
>> Personally, I have only found 2.0.72 as mentioned verison in wiki and
>> emails and 2.0.0.72 as available in the internet. At the same time no
>> proof that there is a mistake in WIKI for the version number mentioned
>> as 2.0.72 instead 2.0.0.72 or proof of it's correctness...
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sadhika at gmail.com  Wed Apr 20 10:20:26 2011
From: sadhika at gmail.com (Sadhika Sharma)
Date: Wed, 20 Apr 2011 11:50:26 +0530
Subject: [Freeswitch-users] Help with rxfax debug "T4 expires in
	phaseT30_PHASE_B_RX"
In-Reply-To: <5A5BB7C35D0E47CABC729B3F01D4F264@e1705>
References: <BANLkTimW6WXQ+_Ge+ASG5u1bVe_j36pTHw@mail.gmail.com>
	<BANLkTikAz7Rcci6g_95mmJD7xAm=iHbZTA@mail.gmail.com>
	<4DAD87B1.7020102@coppice.org>
	<BANLkTikjeWYgw5jJf0Gr0dyFj4_BGd01Zw@mail.gmail.com>
	<BANLkTinUgd2xApkHjbaWjdJHZpyXvshwJg@mail.gmail.com>
	<5A5BB7C35D0E47CABC729B3F01D4F264@e1705>
Message-ID: <BANLkTi=hgf9q99_90ZsavQ1fSu4=cfqEdA@mail.gmail.com>

Hi,

Once again I am stating my problem:
I am using freeswitch for sending and receiving faxes. Incoming and outgoing
faxes goes through seperate PSTN lines.
Whenever a call lands on freeswitch machine, I execute rxfax(). I get about
5000 fax calls every month, out of which about 800 transmissions fail due to
hangup cause 548 'Disconnected after permitted retries'

Logs of a failed attempt are posted here.
<http://pastebin.com/FTXjkFrP>(This log does not explicitly contain
the failed rxfax() execution. There
are other freeswitch operations happening simultaneously.)

Thanks in advance for your help.
Sadhika

On Tue, Apr 19, 2011 at 7:49 PM, Madovsky <infos at madovsky.org> wrote:

>  JIRA
>
> ----- Original Message -----
> *From:* Sadhika Sharma <sadhika at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, April 19, 2011 10:13 AM
> *Subject:* Re: [Freeswitch-users] Help with rxfax debug "T4 expires in
> phaseT30_PHASE_B_RX"
>
> Is there a better way to attach logs? I am not sure if attaching them with
> email is good enough.
> Also how do i prefix uuid to mod_spandsp logs?
>
> On Tue, Apr 19, 2011 at 7:15 PM, Sadhika Sharma <sadhika at gmail.com> wrote:
>
>> Thanks Steve for replying,
>>
>> I will explain the problem in clearer terms. Whenever a call lands on
>> freeswitch machine, I exceute rxfax(). I get about 5000 fax calls every
>> month, out of which about 800 transmissions fail due to hangup cause 548
>> 'Disconnected after permitted retries' Complete log is attached.
>>
>>
>> On Tue, Apr 19, 2011 at 6:31 PM, Steve Underwood <steveu at coppice.org>wrote:
>>
>>> Sadhika,
>>>
>>> If you post only a fragment of a log, and describe only half of what
>>> happens, people will generally ignore what you post. I guess they don't
>>> like being treated like they are mind readers.
>>>
>>> The error seems pretty clear. You aren't getting anything from the far
>>> end. Since you give no clue as to what you are doing, beyond trying to
>>> receive FAX, that's about as detailed an analysis as you are going to
>>> get.
>>>
>>> Steve
>>>
>>>
>>> On 04/19/2011 05:43 PM, Sadhika Sharma wrote:
>>> > Anybody having any idea about this?
>>> >
>>> > On Fri, Apr 15, 2011 at 7:36 PM, Sadhika Sharma <sadhika at gmail.com
>>>  > <mailto:sadhika at gmail.com>> wrote:
>>> >
>>> >     Hi,
>>> >
>>> >     I need help in debugging hangup cause 548, 'Disconnected after
>>> >     permitted retries' while attempting rxfax().
>>> >
>>> >     Logs show that T4 expires in phase T30_PHASE_B_RX.
>>> >     T30 PHASE B concerns exchange of DIS/DCS messages. After sending
>>> >     DIS, mod spandsp keeps waiting for DCS and exhausts its retries.
>>> >
>>> >     Please guide me how to debug the root cause of loss of DCS
>>> >     message. Logs are mentioned below:
>>> >
>>> >     2011-03-07 15:53:48.675180 [DEBUG] switch_ivr.c:557
>>> >     OpenZAP/4:1/43851609 Command Execute
>>> >     rxfax(/srv/fax/in/fcf15a4c-48a4-11e0-919e-7bf53bcc3f81.tiff)
>>> >     2011-03-07 15:53:53.334614 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>>> >     Send complete in phase T30_PHASE_B_TX, state 17
>>> >     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>>> >     DIS:
>>> >     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW
>>> >     T.30   .... ...0= Store and forward Internet fax (T.37): Not set
>>> >     2011-03-07 15:53:53.335619 [DEBUG] mod_spandsp_fax.c:293 FLOW
>>> >     T.30   .... .0..= Real-time Internet fax (T.38): Not set
>>> >     :
>>> >     :
>>> >     2011-03-07 15:53:53.875238 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>>> >     Start T4
>>> >     2011-03-07 15:53:53.895315 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>>> >     HDLC signal status is Carrier up (-2) in state 17
>>> >     2011-03-07 15:53:53.915402 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>>> >     HDLC signal status is Carrier down (-1) in state 17
>>> >     2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>>> >     T4 expired in phase T30_PHASE_B_RX, state 17
>>> >     2011-03-07 15:53:57.335032 [DEBUG] mod_spandsp_fax.c:293 FLOW T.30
>>> >     Retry number 1
>>> >
>>> >
>>> >     Please guide me how to debug the root cause of loss of DCS message.
>>> >     Thanks,
>>> >     Sadhika
>>> >
>>> >
>>> >
>>> >
>>> > --
>>> > Sadhika
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sadhika
>>
>
>
>
> --
> Sadhika
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sadhika
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From kbdfck at gmail.com  Wed Apr 20 10:58:01 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 20 Apr 2011 10:58:01 +0400
Subject: [Freeswitch-users] Disable IP change TRAP?
Message-ID: <BANLkTik17Qtp-NqetDn=X+V3QCh1shD-bQ@mail.gmail.com>

Hi All!
How do I disable ip change trap and following profile restart?

I have two profiles, bound to separate ip addresses on eth0, both SIP and
RTP addresses. But when freeswitch detect IP change (about once a hour), it
tries to apply changes, but actually nothing changed as profiles are bound
to fixed ip addresses.

What can be done to make FS ignore this traps or at least not to touch
working profiles?

-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From fieldpeak at gmail.com  Wed Apr 20 11:27:20 2011
From: fieldpeak at gmail.com (Charles)
Date: Wed, 20 Apr 2011 15:27:20 +0800
Subject: [Freeswitch-users] FS -route to next GW if the first GW full
References: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>
Message-ID: <4dae8ae9.823d2b0a.02b3.155f@mx.google.com>


i have two media GWs connected with FS for PSTN calls, FS route the first one in normal case, once the first GW get full (all of channels were occupied), i need FS route to the second GW. 
i found once the first GW get full, it will reply '404 not found' to FS, can FS route to the second one according to this condition or any other alternative? 

although i know there is mod_distributor for load balancing, however, i need if only first one full then route to second one, so it looks this module not suitable for this senario...

Thanks.

Regards,
Charles
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From darknesslabs at gmail.com  Wed Apr 20 08:35:35 2011
From: darknesslabs at gmail.com (Karol)
Date: Wed, 20 Apr 2011 00:35:35 -0400
Subject: [Freeswitch-users] Rejected by acl "net_list_5". Falling back to
	Digest auth.
Message-ID: <BANLkTimk-sjF2DjqMMTu5WpW7taizHHAjw@mail.gmail.com>

OK, I am having some issues here. What I have going on here, is 2 phones
(one aastra 6730i and 1 nortel 1535) at home. Using the suppiled router by
verzion fios (actiontec), I connect to a vps out on the internet. This box
is running the latest stable freeswitch (1.0.7). I keep getting a net_list_5
error, I've tried using UPNP and STUN server. Nothing has helped, I've also
reset my router thinking something is wrong. Below is the errors, I see if
trying to make a call. Phone to phones works and it should work.

[DEBUG] sofia.c:6551 IP xxx.xxx.xxx.xxx Rejected by acl "net_list_5".
Falling back to Digest auth.

[WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile
'sipinterface_1' for [110 at voip.lightnologies.com] from ip xxx.xxx.xxx.xxx

Anyone have any ideas??

Thanks in advance.
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From michel at arneill-py.sacramento.ca.us  Wed Apr 20 07:42:11 2011
From: michel at arneill-py.sacramento.ca.us (Michel Py)
Date: Tue, 19 Apr 2011 20:42:11 -0700
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
	with productivity license.
In-Reply-To: <BANLkTimUkXz84DHh6bac4ognmYL1WzqdiQ@mail.gmail.com>
References: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local>
	<BANLkTimUkXz84DHh6bac4ognmYL1WzqdiQ@mail.gmail.com>
Message-ID: <471D76419F9EF642962323D13DF1DF69011E4A@newserver.arneill-py.local>

Hello Yehavi,

> Yehavi Bourvine wrote:
> Version 3.1.7 is very old. Try the 3.3.0 version
> (3.3.1F loses BLF after a while).

I would, if I could get it to load on a 601. I actually tried: I renamed a 3.2 file with the name that the 601 expects, no luck (it says platform does not support the software).

I have read contradictory information whether "enhanced BLF" is actually working on pre-3.2. I think I understand the difference between "buddies" and "attendant" but again, if I am posting here is because I don't understand as much as I think I do :P

If I am chasing a wild goose, please tell me.

Michel.


Michel Py <michel at arneill-py.sacramento.ca.us>
Hi guys,
?
I'm reading the following page:
http://wiki.freeswitch.org/wiki/Polycom_Presence_Setup
and I'm trying to make it work the "attendant" way.
?
The phone is a 601 with 3.1.7.0134. I think the productivity license is installed correctly, both because I see the phone loading the file in the logs and that I enabled corporate directory in sip_317.cfg and I see it in menu - settings - basic - preferences.
?
I made the mac-address-settings.cfg file as outlined, and none of the attendant things even displays on the phone.
?
What am I missing there? Where do I start the troubleshooting?
?
Thanks
Michel.
?
?
?

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



From skchopperguy at gmail.com  Wed Apr 20 09:40:57 2011
From: skchopperguy at gmail.com (Skyler)
Date: Tue, 19 Apr 2011 22:40:57 -0700
Subject: [Freeswitch-users] SBC for carrier
In-Reply-To: <BANLkTinannjPrtj9oX=R7ugRydrJ-m6nfA@mail.gmail.com>
References: <BANLkTinannjPrtj9oX=R7ugRydrJ-m6nfA@mail.gmail.com>
Message-ID: <059f01cbff1d$86cb0e60$94612b20$@com>

Why reply directly? I'd like to know too ;)

 

 My 0.02 would be a dual Quad-Core with at least 8gb Ram running OpenSips or
Kamailio in stateless or transactional stateful mode. If you want it to
handle NAT and RTP as well that's easily done but more RAM would be required
most likely. Of course FS distributed to handle the backend ;)

 

Skyler

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
dome at tel.co.th
Sent: Tuesday, April 19, 2011 9:25 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] SBC for carrier

 

Dear all,
    I'm freeswitch user :)  but looking for carrier grade SBC for my
customer. like a acme , nexttone. they talking about 120,000
concurrent. so if someone know about this please send me directly.

Thanks.

Dome C.

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

  _____  

No virus found in this message.
Checked by AVG - www.avg.com
Version: 10.0.1209 / Virus Database: 1500/3584 - Release Date: 04/19/11

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From mitja.thomas1 at ewetel.de  Wed Apr 20 11:17:52 2011
From: mitja.thomas1 at ewetel.de (Mitja Thomas)
Date: Wed, 20 Apr 2011 09:17:52 +0200
Subject: [Freeswitch-users] [freeswitch] <defunct> when using cmd="exec"
 in freeswitch.xml
In-Reply-To: <BANLkTi=+i=Ew=Ant8WOgxd4hC_kDxmKetw@mail.gmail.com>
References: <4D78A5AF.1000808@ewetel.de>	<AANLkTin7EUJpz4=9gxp3sTXykjybH5B4Uer7noD+Bgct@mail.gmail.com>	<4D7F251F.2070101@ewetel.de>
	<BANLkTi=+i=Ew=Ant8WOgxd4hC_kDxmKetw@mail.gmail.com>
Message-ID: <4DAE88A0.5050404@ewetel.de>

Yepp thanks,

I saw that they fixed because someelse posted a similiar question and 
Anthony told him there was a fix for it. I tried it and it works fine.
BTW: Sorry that I didnt opened a bug report on jira, but I wasnt sure at 
the time if it was a bug or just a misuse on my part.

Regards
Mitja

> The child process needs to be "reaped" in order to get out of
> <defunct>  and of course it'll be cleared out when the parent process
> finally exits.
>
> <looking at code>
>
> I don't see any reason why it would leave a zombie.  The code
> correctly calls waitpid() after the fork in the parent code.
>
> Oh, looks like brian added waitpid on :
>
> dae2cb4a (Brian West             2011-03-29 18:05:05 -0500 1246)
>
> So update and it should work.
>
> On Tue, Mar 15, 2011 at 3:36 AM, Mitja Thomas<mitja.thomas1 at ewetel.de>  wrote:
>> Hi Micheal,
>>
>> yeah I think your workaround will work fine and Ill change it that way. But
>> this issue might still be a missbehaviour. Either by fs in which case I
>> think it might be an interesting information for the Developers OR a wrong
>> usage by me in which case this is an interesting information for me :)
>>
>> Thanks,
>> Mitja
>>
>> IIRC, 'fork' creates a child PID that stays around and waits for the parent
>> PID to die. (Don't quote me on that - wait for an expert to chime in.)
>> An alternative method would be to have a FS start script that launches your
>> make_my_vars.sh script and then launches FS with the appropriate cmd line
>> args. You could then drop the 'exec' cmd and then just keep the include
>> cmd.
>> -MC
>>
>> On Thu, Mar 10, 2011 at 2:19 AM, Mitja Thomas<mitja.thomas1 at ewetel.de>
>> wrote:
>>> Hi there,
>>>
>>> we tried to set up the FreeSWITCH and other Applications, so that we can
>>> configure them easier and more centralised.
>>> Thus we defined some Environment Variables (using CentOS) which hold often
>>> used Configuration Parameter like MySQL IP or FS Event Socket IP.
>>> We tried to integrate these Env Variables into the FS conf files by
>>> executing a shell Skript in freeswitch.xml via cmd=xec" which prepares an
>>> conf file which we include into freeswitch.xml:
>>>
>>>    <X-PRE-PROCESS cmd=xec" data="sh conf/make_my_vars.sh"/>
>>>    <X-PRE-PROCESS cmd=nclude" data="my_vars.xml"/>
>>>
>>> This works as expected and the pre defined variables in my_vars can be
>>> accessed from the other config Files, except that when we start our
>>> FreeSWITCH  a zombie child process is spawned.
>>>
>>> # ps -eaf | grep free
>>> ippbx    22191 22190  4 09:41 pts/1    00:00:01
>>> /opt/app/voip/ippbx/bin/freeswitch -waste -nonat -hp
>>> ippbx    22197 22191  0 09:41 pts/1    00:00:00 [freeswitch]<defunct>
>>>
>>> What I wanna know is: Is this a FS missbehaviour or do we use this in a
>>> wrong way?
>>>
>>> make_my_vars.sh:
>>> F=onf/my_vars.xml"
>>> echo "<!-- Auto generated file for getting *NIX evironment variables -->"
>>>> $F
>>> echo "<include>">>  $F
>>>
>>> fs_ip=rintenv MY_FS_IP`
>>> if test -n "$fs_ip"
>>> then
>>>          echo '<X-PRE-PROCESS cmd=et" data="my_fs_ip='$fs_ip'"/>'>>  $F
>>> fi
>>> ...
>>> echo "</include>">>  $F
>>>
>>> my_vars.xml (after FS startup):
>>> <!-- Auto generated file for getting *NIX evironment variables -->
>>> <include>
>>> <X-PRE-PROCESS cmd=et" data="my_fs_ip=***.***.***.***"/>
>>> ...
>>> </include>
>>>
>>> Regards
>>> Mitja
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>


-- 
Mitja Thomas
GE FD
L?sungsmanagement

Telefon: 0441 8000 - 4916
E-Mail: mitja.thomas at ewe.de
___________________________________

EWE TEL GmbH
Cloppenburger Stra?e 310
26133 Oldenburg

Handelsregister Amtsgericht Oldenburg HRB 3723
Vorsitzender des Aufsichtsrates: Dr. Werner Brinker
Gesch?ftsf?hrung: Konrad Meier (Vorsitzender), Dirk Brameier,
Ulf Heggenberger, Norbert Westfal
Homepage: http://www.ewetel.de

Wir stehen f?r starken Service. Versprochen!



From anton.vazir at gmail.com  Wed Apr 20 11:52:09 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 20 Apr 2011 12:52:09 +0500
Subject: [Freeswitch-users] Download link to Skyopen
In-Reply-To: <BANLkTikysBSN5ccqUUJCXOYY_rw2iZTyXw@mail.gmail.com>
References: <4D906E17.4000609@greatiam.com>
	<BANLkTi=Eik2MjC+4ouQz9wmtf7E1MufRwQ@mail.gmail.com>
	<BANLkTinhHimVxS5nSr9e+8Pa7nztssLZUw@mail.gmail.com>
	<BANLkTikysBSN5ccqUUJCXOYY_rw2iZTyXw@mail.gmail.com>
Message-ID: <BANLkTi=yrEoE5=c+5h0da8=LtiK-_BD+ug@mail.gmail.com>

Thanks so much, Giovanni, this helps a lot!

2011/4/20 Giovanni Maruzzelli <gmaruzz at gmail.com>:
> 2.0.0.72
> wikipage fixed
>
> On Wed, Apr 20, 2011 at 2:58 AM, haloha <haloha201 at gmail.com> wrote:
>> Hi
>>
>> try this link if it work for you
>> http://www.fshare.vn/file/883OBXSPUR/
>>
>>
>> hope it help
>>
>> Ha`
>>
>> On 4/20/11, Anton VG <anton.vazir at gmail.com> wrote:
>>> Seems this topic is either forbidden or not at anyones interest.
>>> Personally, I have only found 2.0.72 as mentioned verison in wiki and
>>> emails and 2.0.0.72 as available in the internet. At the same time no
>>> proof that there is a mistake in WIKI for the version number mentioned
>>> as 2.0.72 instead 2.0.0.72 or proof of it's correctness...
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From steveayre at gmail.com  Wed Apr 20 12:00:44 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 20 Apr 2011 09:00:44 +0100
Subject: [Freeswitch-users] Rejected by acl "net_list_5". Falling back
 to Digest auth.
In-Reply-To: <BANLkTimk-sjF2DjqMMTu5WpW7taizHHAjw@mail.gmail.com>
References: <BANLkTimk-sjF2DjqMMTu5WpW7taizHHAjw@mail.gmail.com>
Message-ID: <BANLkTinBZhQgAjBiuG4o-XX45vQMiwk1RA@mail.gmail.com>

>
> [DEBUG] sofia.c:6551 IP xxx.xxx.xxx.xxx Rejected by acl "net_list_5".
> Falling back to Digest auth.
>

That's normal and not an error message (it says debug). It's telling you the
phone isn't on the net_list_5 acl, so won't be allowed in based on IP. It's
falling back to Digest (password) authentication.

[WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile
> 'sipinterface_1' for [110 at voip.lightnologies.com] from ip xxx.xxx.xxx.xxx
>

That is also not an error, per-say. It's one of a number of messages that
are logged for use with fail2ban, so that you can automatically block people
trying to hack into or DOS your server through repeated connection attempts.

See: http://wiki.freeswitch.org/wiki/Fail2ban

If you don't want to use fail2ban you can hide those messages by putting
<param name="log-auth-failures" value="false"/> on your SIP profiles.


-Steve





On 20 April 2011 05:35, Karol <darknesslabs at gmail.com> wrote:

> OK, I am having some issues here. What I have going on here, is 2 phones
> (one aastra 6730i and 1 nortel 1535) at home. Using the suppiled router by
> verzion fios (actiontec), I connect to a vps out on the internet. This box
> is running the latest stable freeswitch (1.0.7). I keep getting a net_list_5
> error, I've tried using UPNP and STUN server. Nothing has helped, I've also
> reset my router thinking something is wrong. Below is the errors, I see if
> trying to make a call. Phone to phones works and it should work.
>
> [DEBUG] sofia.c:6551 IP xxx.xxx.xxx.xxx Rejected by acl "net_list_5".
> Falling back to Digest auth.
>
> [WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile
> 'sipinterface_1' for [110 at voip.lightnologies.com] from ip xxx.xxx.xxx.xxx
>
> Anyone have any ideas??
>
> 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 steveayre at gmail.com  Wed Apr 20 12:10:21 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 20 Apr 2011 09:10:21 +0100
Subject: [Freeswitch-users] FS -route to next GW if the first GW full
In-Reply-To: <4dae8ae9.823d2b0a.02b3.155f@mx.google.com>
References: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>
	<4dae8ae9.823d2b0a.02b3.155f@mx.google.com>
Message-ID: <BANLkTino_bR7=BvhvMdhOWmbMVXcwFQJ5Q@mail.gmail.com>

If you know the number of calls the GW can handle in advance, you can limit
the call attempts before sending the call to the gateway:
http://wiki.freeswitch.org/wiki/Limit
There are several ways to use that.

You can reroute calls to a 2nd gateway on error:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover

There are channel variables that let you control what to consider a
reroutable error and what is a give-up error:
http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
http://wiki.freeswitch.org/wiki/Channel_Variables#failure_causes
This might also be useful, particularly with mod_limit:
http://wiki.freeswitch.org/wiki/Channel_Variables#transfer_on_fail

You could use mod_lcr to get a list of all the GWs, but in a random order.
That'd let you load balance (randomly) but reroute when required without
duplicates unlike mod_distributor.

Hopefully that's enough building blocks to give you somewhere to start...

-Steve



On 20 April 2011 08:27, Charles <fieldpeak at gmail.com> wrote:

>
> i have two media GWs connected with FS for PSTN calls, FS route the first
> one in normal case, once the first GW get full (all of channels were
> occupied), i need FS route to the second GW.
> i found once the first GW get full, it will reply '404 not found' to FS,
> can FS route to the second one according to this condition or any other
> alternative?
>
> although i know there is mod_distributor for load balancing, however, i
> need if only first one full then route to second one, so it looks this
> module not suitable for this senario...
>
> 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
>
>
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From steveayre at gmail.com  Wed Apr 20 12:14:23 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 20 Apr 2011 09:14:23 +0100
Subject: [Freeswitch-users] Outbound caller id for a specific gateway?
In-Reply-To: <AANLkTimnorNHZa_3KqeqgecKjgHo4PLQWsm8+xH_FJMQ@mail.gmail.com>
References: <AANLkTin=M3MJKB5QA86iL+hkhkZncTQ8eiKQJM2t6ap5@mail.gmail.com>
	<AANLkTimqg7TRBULWR2__kxu32K4QuHQi1ChgsAZGiW+G@mail.gmail.com>
	<AANLkTimnorNHZa_3KqeqgecKjgHo4PLQWsm8+xH_FJMQ@mail.gmail.com>
Message-ID: <BANLkTikYBgua9M3ya8wvJ8UjFMU5EEjQHg@mail.gmail.com>

How about you set the channel variable in the gateway definition for the
outbound direction:
http://wiki.freeswitch.org/wiki/Sofia.conf.xml#Variables

That way it will be automatically set on all calls going through that
gateway without you needing to change your dialplan.

-Steve




On 22 March 2011 07:49, Dmitry Bely <dmitry.bely at gmail.com> wrote:

> On Tue, Mar 22, 2011 at 8:02 AM, Michael Collins <msc at freeswitch.org>
> wrote:
> > How about set the caller id in vars.xml:
> > <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
> > Set it to the most common value and then you only have to do something in
> > bridges and originates that need a CID different from what you set in
> > vars.xml...
>
> I have a number of gateways that require different CIDs. So there is
> no "most common" value. I have managed to make a proper dialplan but
> originate is still tedious... I just wonder if there is more clean way
> (something like caller-id-in-from gateway parameter)
>
> > -MC
> >
> > On Sat, Mar 19, 2011 at 4:27 AM, Dmitry Bely <dmitry.bely at gmail.com>
> wrote:
> >>
> >> My VoIP provider requires a specific caller ID set for an outbound
> >> call otherwise the call is rejected. Currently I set it just before
> >> bridge
> >>
> >>      <action application="set" data="effective_caller_id_name=..."/>
> >>      <action application="bridge" data="..."/>
> >>
> >> But it's tedious as there is a number of bridge commands in the
> >> dialplan and I still have to explicitly specify the caller id for
> >> "originate" command in the FreeSWITCH console. Is it possible to force
> >> an outbound caller id on a gateway basis?
>
> - Dmitry Bely
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 leon at scarlet-internet.nl  Wed Apr 20 14:59:55 2011
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Wed, 20 Apr 2011 12:59:55 +0200
Subject: [Freeswitch-users] CallsIN counter on gateway
Message-ID: <10717914-15BF-4D58-9A77-8FAE698661D8@scarlet-internet.nl>

Hi all,

I have a gateway that sends me invites with the callerid in the from: header and the destination number in the to: header. I'd like these inbound invites to be matched to the proper gateway (that is used for outbound calls) so that I can use params of the gateway like context, but also so the CallsIN counter gets increased for each inbound call.

Is this at all possible ? From what I understand from sofia.c, the gateway->ib_calls is only increased when an incoming request contains a gw= param or when destination_number is prefixed by gw+, or is there another way ?

thanks & kind regards,

Leon de Rooij

From Prometheus001 at gmx.net  Wed Apr 20 16:08:08 2011
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 20 Apr 2011 14:08:08 +0200
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
Message-ID: <4DAECCA8.1050203@gmx.net>

Hello,

I am trying to use the mod_callcenter functionality. This works nicely
so far so thank you to everybody involved for programming this nice module!
But I am stuck somehow with uuid-standby. Can anybody explain how
uuid-standby works?

Another question: In the Asterisk based callcenter solution named
"Vicidial", an agent can be held permanently in a conference, waiting
for calls who are bridged to his uuid in the conference. Can this be
haviour be done with mod_callcenter?


Best regards
Peter


From alex8207744 at gmail.com  Wed Apr 20 16:48:29 2011
From: alex8207744 at gmail.com (Alex Zhou)
Date: Wed, 20 Apr 2011 20:48:29 +0800
Subject: [Freeswitch-users] Program Crash! What should i do ?
Message-ID: <BANLkTik3t52A_H3PJp1L7bivxhQ74bWhUQ@mail.gmail.com>

Hi ! Master

I wrote a auto dial out program named waihu5.  It's main function is
fetch phone number from Oracle DB using unixODBC, then call a tuxedo
services to verify the phone number.When the verify is passed, update
database status for this phone number . Finally ,when receive the
HANG-UP event for this phone number, get Hangup-Cause and again update
database status for this phone number.

It was runing perfect this morning for 1.5 hour,and processed and
dialed out about 10000 phone number. But this afternoon,when it  has
finished 16000 phone number,it's crashed!

I just don't know how to fix it?Can you help me.

Here is crash info:http://pastebin.com/3xTk0GgN

I have no dump file .I think since it was crashed ,it will crash again
.Then i will get crash file,because i have about 90000 phone number to
dial out. So if you know where is the problem ,please tell me.

Thanks a lot!

My english is so bad,i wish you can understand what i said. Thanks again.


From peter.olsson at visionutveckling.se  Wed Apr 20 17:35:47 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 20 Apr 2011 15:35:47 +0200
Subject: [Freeswitch-users] Program Crash! What should i do ?
In-Reply-To: <BANLkTik3t52A_H3PJp1L7bivxhQ74bWhUQ@mail.gmail.com>
References: <BANLkTik3t52A_H3PJp1L7bivxhQ74bWhUQ@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58C50EE9C0@cooper>

If I understand you correctly it's not an FS crash - it's in your own software?

/Peter

-----Ursprungligt meddelande-----
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Alex Zhou
Skickat: den 20 april 2011 14:48
Till: freeswitch-users at lists.freeswitch.org
?mne: [Freeswitch-users] Program Crash! What should i do ?

Hi ! Master

I wrote a auto dial out program named waihu5.  It's main function is
fetch phone number from Oracle DB using unixODBC, then call a tuxedo
services to verify the phone number.When the verify is passed, update
database status for this phone number . Finally ,when receive the
HANG-UP event for this phone number, get Hangup-Cause and again update
database status for this phone number.

It was runing perfect this morning for 1.5 hour,and processed and
dialed out about 10000 phone number. But this afternoon,when it  has
finished 16000 phone number,it's crashed!

I just don't know how to fix it?Can you help me.

Here is crash info:http://pastebin.com/3xTk0GgN

I have no dump file .I think since it was crashed ,it will crash again
.Then i will get crash file,because i have about 90000 phone number to
dial out. So if you know where is the problem ,please tell me.

Thanks a lot!

My english is so bad,i wish you can understand what i said. Thanks again.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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:4daed73732768906810856!



From rupa at rupa.com  Wed Apr 20 17:57:43 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 20 Apr 2011 08:57:43 -0500
Subject: [Freeswitch-users] Disable IP change TRAP?
In-Reply-To: <BANLkTik17Qtp-NqetDn=X+V3QCh1shD-bQ@mail.gmail.com>
References: <BANLkTik17Qtp-NqetDn=X+V3QCh1shD-bQ@mail.gmail.com>
Message-ID: <BANLkTi=cx=TbokKLF7vG45bZiQBBqY12WA@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address

On Wed, Apr 20, 2011 at 1:58 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> Hi All!
> How do I disable ip change trap and following profile restart?
> I have two profiles, bound to separate ip addresses on eth0, both SIP and
> RTP addresses. But when freeswitch detect IP change (about once a hour), it
> tries to apply changes, but actually nothing changed as profiles are bound
> to fixed ip addresses.
> What can be done to make FS ignore this traps or at least not to touch
> working profiles?
> --
> 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
>
>



-- 
-Rupa


From jeff at jefflenk.com  Wed Apr 20 18:37:27 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 20 Apr 2011 07:37:27 -0700 (PDT)
Subject: [Freeswitch-users] Where is mod_spidermonkey_etpan?
In-Reply-To: <C42A22554C0CF44DAC0AA1F132B86DE657F7EA38F3@ASKSERVER.asksbs.local>
References: <C42A22554C0CF44DAC0AA1F132B86DE657F7EA38F3@ASKSERVER.asksbs.local>
Message-ID: <1303310247416-6290956.post@n2.nabble.com>

That module was removed sometime ago - patches excepted to add it back if you
are willing to help maintain it.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Where-is-mod-spidermonkey-etpan-tp6289652p6290956.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From curriegrad2004 at gmail.com  Wed Apr 20 18:44:55 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Wed, 20 Apr 2011 07:44:55 -0700
Subject: [Freeswitch-users] Download link to Skyopen
In-Reply-To: <BANLkTi=yrEoE5=c+5h0da8=LtiK-_BD+ug@mail.gmail.com>
References: <4D906E17.4000609@greatiam.com>
	<BANLkTi=Eik2MjC+4ouQz9wmtf7E1MufRwQ@mail.gmail.com>
	<BANLkTinhHimVxS5nSr9e+8Pa7nztssLZUw@mail.gmail.com>
	<BANLkTikysBSN5ccqUUJCXOYY_rw2iZTyXw@mail.gmail.com>
	<BANLkTi=yrEoE5=c+5h0da8=LtiK-_BD+ug@mail.gmail.com>
Message-ID: <BANLkTi=fooPTYR+CAaCqgEFO3n2ie7Um-w@mail.gmail.com>

There's always the option of Sip2Sis, which worked pretty well for
what I need it to do

On Wed, Apr 20, 2011 at 12:52 AM, Anton VG <anton.vazir at gmail.com> wrote:
> Thanks so much, Giovanni, this helps a lot!
>
> 2011/4/20 Giovanni Maruzzelli <gmaruzz at gmail.com>:
>> 2.0.0.72
>> wikipage fixed
>>
>> On Wed, Apr 20, 2011 at 2:58 AM, haloha <haloha201 at gmail.com> wrote:
>>> Hi
>>>
>>> try this link if it work for you
>>> http://www.fshare.vn/file/883OBXSPUR/
>>>
>>>
>>> hope it help
>>>
>>> Ha`
>>>
>>> On 4/20/11, Anton VG <anton.vazir at gmail.com> wrote:
>>>> Seems this topic is either forbidden or not at anyones interest.
>>>> Personally, I have only found 2.0.72 as mentioned verison in wiki and
>>>> emails and 2.0.0.72 as available in the internet. At the same time no
>>>> proof that there is a mistake in WIKI for the version number mentioned
>>>> as 2.0.72 instead 2.0.0.72 or proof of it's correctness...
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From fieldpeak at gmail.com  Wed Apr 20 18:58:46 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Wed, 20 Apr 2011 22:58:46 +0800
Subject: [Freeswitch-users] FS -route to next GW if the first GW full
In-Reply-To: <BANLkTino_bR7=BvhvMdhOWmbMVXcwFQJ5Q@mail.gmail.com>
References: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>
	<4dae8ae9.823d2b0a.02b3.155f@mx.google.com>
	<BANLkTino_bR7=BvhvMdhOWmbMVXcwFQJ5Q@mail.gmail.com>
Message-ID: <BANLkTi=Pi9_gkv-WbkyijJH6-Kj1VgEuvg@mail.gmail.com>

Hi Steve,

Thanks for your so rich stuff provided.

however, i tried to use error code to route as below, it failed (did
not route the next GW when recevied UNALLOCATED_NUMBER), can you
please point out is there any place wrong in the dailplan? attached is
the log, thanks.

<extension name="9_GW" >

   <condition field="destination_number" expression="^(9\d+)$">
      <action application="set" data="continue_on_fail=true"/>
     <action application="set" data="hangup_after_bridge=false"/>
      <action application="bridge" data="sofia/internal/$1 at 192.168.200.201"/>
 </condition>

   <condition field="brige_hangup_cause" expression="UNALLOCATED_NUMBER">
    <action application="bridge" data="sofia/internal/$1 at 192.168.200.202"/>
   </condition>

 </extension>

Regards,
Charles

2011/4/20, Steven Ayre <steveayre at gmail.com>:
> If you know the number of calls the GW can handle in advance, you can limit
> the call attempts before sending the call to the gateway:
> http://wiki.freeswitch.org/wiki/Limit
> There are several ways to use that.
>
> You can reroute calls to a 2nd gateway on error:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover
>
> There are channel variables that let you control what to consider a
> reroutable error and what is a give-up error:
> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
> http://wiki.freeswitch.org/wiki/Channel_Variables#failure_causes
> This might also be useful, particularly with mod_limit:
> http://wiki.freeswitch.org/wiki/Channel_Variables#transfer_on_fail
>
> You could use mod_lcr to get a list of all the GWs, but in a random order.
> That'd let you load balance (randomly) but reroute when required without
> duplicates unlike mod_distributor.
>
> Hopefully that's enough building blocks to give you somewhere to start...
>
> -Steve
>
>
>
> On 20 April 2011 08:27, Charles <fieldpeak at gmail.com> wrote:
>
>>
>> i have two media GWs connected with FS for PSTN calls, FS route the first
>> one in normal case, once the first GW get full (all of channels were
>> occupied), i need FS route to the second GW.
>> i found once the first GW get full, it will reply '404 not found' to FS,
>> can FS route to the second one according to this condition or any other
>> alternative?
>>
>> although i know there is mod_distributor for load balancing, however, i
>> need if only first one full then route to second one, so it looks this
>> module not suitable for this senario...
>>
>> 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
>>
>>
>
-------------- next part --------------
log:
vswitch at mypc> 2011-04-20 22:52:22.496001 [DEBUG] sofia.c:6539 IP 192.168.200.100 Rejected by acl "172.28.0.0/16". Falling back to Digest auth.
2011-04-20 22:52:22.496001 [WARNING] sofia_reg.c:1246 SIP auth challenge (INVITE) on sofia profile 'internal' for [9123 at 192.168.200.100] from ip 192.1
68.200.100
2011-04-20 22:52:22.519003 [DEBUG] sofia.c:6539 IP 192.168.200.100 Rejected by acl "172.28.0.0/16". Falling back to Digest auth.
2011-04-20 22:52:22.526003 [NOTICE] switch_channel.c:812 New Channel sofia/internal/1002 at 192.168.200.100 [e709297d-aae8-46b3-b75e-68b2ab63164a]
2011-04-20 22:52:22.532003 [DEBUG] sofia.c:4760 Channel sofia/internal/1002 at 192.168.200.100 entering state [received][100]
2011-04-20 22:52:22.532003 [DEBUG] sofia.c:4771 Remote SDP:
v=0
o=- 12947784742458999 1 IN IP4 192.168.200.100
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.200.100
t=0 0
a=ice-ufrag:260f31
a=ice-pwd:2110abb4ecb2953f4600b8f332fdb1d3
m=audio 61294 RTP/AVP 0 97 8 3 101
a=rtpmap:97 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=candidate:1 1 UDP 659136 192.168.200.100 61294 typ host
a=candidate:1 2 UDP 659134 192.168.200.100 61295 typ host
2011-04-20 22:52:22.532003 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002 at 192.168.200.100) Running State Change CS_NEW
2011-04-20 22:52:22.533003 [DEBUG] switch_core_state_machine.c:343 (sofia/internal/1002 at 192.168.200.100) State NEW
2011-04-20 22:52:22.533003 [DEBUG] sofia.c:4916 (sofia/internal/1002 at 192.168.200.100) State Change CS_NEW -> CS_INIT
2011-04-20 22:52:22.533003 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 at 192.168.200.100 [BREAK]
2011-04-20 22:52:22.534003 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002 at 192.168.200.100) Running State Change CS_INIT
2011-04-20 22:52:22.534003 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/1002 at 192.168.200.100) State INIT
2011-04-20 22:52:22.534003 [DEBUG] mod_sofia.c:84 sofia/internal/1002 at 192.168.200.100 SOFIA INIT
2011-04-20 22:52:22.534003 [DEBUG] mod_sofia.c:124 (sofia/internal/1002 at 192.168.200.100) State Change CS_INIT -> CS_ROUTING
2011-04-20 22:52:22.534003 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 at 192.168.200.100 [BREAK]
2011-04-20 22:52:22.534003 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/1002 at 192.168.200.100) State INIT going to sleep
2011-04-20 22:52:22.534003 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002 at 192.168.200.100) Running State Change CS_ROUTING
2011-04-20 22:52:22.534003 [DEBUG] switch_channel.c:1668 (sofia/internal/1002 at 192.168.200.100) Callstate Change DOWN -> RINGING
2011-04-20 22:52:22.535004 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1002 at 192.168.200.100) State ROUTING
2011-04-20 22:52:22.535004 [DEBUG] mod_sofia.c:147 sofia/internal/1002 at 192.168.200.100 SOFIA ROUTING
2011-04-20 22:52:22.535004 [DEBUG] switch_core_state_machine.c:77 sofia/internal/1002 at 192.168.200.100 Standard ROUTING
2011-04-20 22:52:22.535004 [INFO] mod_dialplan_xml.c:331 Processing 1002 <1002>->9123 in context default
Dialplan: sofia/internal/1002 at 192.168.200.100 parsing [default->unloop] continue=false
Dialplan: sofia/internal/1002 at 192.168.200.100 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/1002 at 192.168.200.100 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/1002 at 192.168.200.100 parsing [default->Local_Extension] continue=false
Dialplan: sofia/internal/1002 at 192.168.200.100 Regex (FAIL) [Local_Extension] destination_number(9123) =~ /^(1\d{3})$/ break=on-false
Dialplan: sofia/internal/1002 at 192.168.200.100 parsing [default->global-intercept] continue=false
Dialplan: sofia/internal/1002 at 192.168.200.100 Regex (FAIL) [global-intercept] destination_number(9123) =~ /^\*1$/ break=on-false
Dialplan: sofia/internal/1002 at 192.168.200.100 parsing [default->intercept-ext] continue=false
Dialplan: sofia/internal/1002 at 192.168.200.100 Regex (FAIL) [intercept-ext] destination_number(9123) =~ /^\*2(\d{4})$/ break=on-false
Dialplan: sofia/internal/1002 at 192.168.200.100 parsing [default->vmain] continue=false
Dialplan: sofia/internal/1002 at 192.168.200.100 Regex (FAIL) [vmain] destination_number(9123) =~ /^vmain$|^\*3$/ break=on-false
Dialplan: sofia/internal/1002 at 192.168.200.100 parsing [default->9_GW] continue=false
Dialplan: sofia/internal/1002 at 192.168.200.100 Regex (PASS) [9_GW] destination_number(9123) =~ /^(9\d+)$/ break=on-false
Dialplan: sofia/internal/1002 at 192.168.200.100 Action set(continue_on_fail=true)
Dialplan: sofia/internal/1002 at 192.168.200.100 Action set(hangup_after_bridge=false)
Dialplan: sofia/internal/1002 at 192.168.200.100 Action bridge(sofia/internal/9123 at 192.168.200.201)
Dialplan: sofia/internal/1002 at 192.168.200.100 Regex (FAIL) [9_GW] brige_hangup_cause() =~ /UNALLOCATED_NUMBER/ break=on-false
Dialplan: sofia/internal/1002 at 192.168.200.100 parsing [default->conf999] continue=false
Dialplan: sofia/internal/1002 at 192.168.200.100 Regex (FAIL) [conf999] destination_number(9123) =~ /^999$/ break=on-false
Dialplan: sofia/internal/1002 at 192.168.200.100 parsing [default->DISA] continue=false
Dialplan: sofia/internal/1002 at 192.168.200.100 Regex (FAIL) [DISA] destination_number(9123) =~ /^\*(3472)$/ break=on-false
Dialplan: sofia/internal/1002 at 192.168.200.100 parsing [default->IVR] continue=false
Dialplan: sofia/internal/1002 at 192.168.200.100 Regex (FAIL) [IVR] destination_number(9123) =~ /^888$/ break=on-false
Dialplan: sofia/internal/1002 at 192.168.200.100 parsing [default->Recordings] continue=false
Dialplan: sofia/internal/1002 at 192.168.200.100 Regex (FAIL) [Recordings] destination_number(9123) =~ /^\*(732673)$/ break=on-false
2011-04-20 22:52:22.540004 [DEBUG] switch_core_state_machine.c:119 (sofia/internal/1002 at 192.168.200.100) State Change CS_ROUTING -> CS_EXECUTE
2011-04-20 22:52:22.540004 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 at 192.168.200.100 [BREAK]
2011-04-20 22:52:22.540004 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/1002 at 192.168.200.100) State ROUTING going to sleep
2011-04-20 22:52:22.540004 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002 at 192.168.200.100) Running State Change CS_EXECUTE
2011-04-20 22:52:22.540004 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/1002 at 192.168.200.100) State EXECUTE
2011-04-20 22:52:22.540004 [DEBUG] mod_sofia.c:240 sofia/internal/1002 at 192.168.200.100 SOFIA EXECUTE
2011-04-20 22:52:22.540004 [DEBUG] switch_core_state_machine.c:157 sofia/internal/1002 at 192.168.200.100 Standard EXECUTE
EXECUTE sofia/internal/1002 at 192.168.200.100 set(continue_on_fail=true)
2011-04-20 22:52:22.540004 [DEBUG] mod_dptools.c:1060 sofia/internal/1002 at 192.168.200.100 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1002 at 192.168.200.100 set(hangup_after_bridge=false)
2011-04-20 22:52:22.541004 [DEBUG] mod_dptools.c:1060 sofia/internal/1002 at 192.168.200.100 SET [hangup_after_bridge]=[false]
EXECUTE sofia/internal/1002 at 192.168.200.100 bridge(sofia/internal/9123 at 192.168.200.201)
2011-04-20 22:52:22.542004 [NOTICE] switch_channel.c:812 New Channel sofia/internal/9123 at 192.168.200.201 [c817db5a-b2fd-445d-91d5-1bcf67291207]
2011-04-20 22:52:22.543004 [DEBUG] mod_sofia.c:4300 (sofia/internal/9123 at 192.168.200.201) State Change CS_NEW -> CS_INIT
2011-04-20 22:52:22.543004 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/9123 at 192.168.200.201 [BREAK]
2011-04-20 22:52:22.545004 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/9123 at 192.168.200.201) Running State Change CS_INIT
2011-04-20 22:52:22.545004 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/9123 at 192.168.200.201) State INIT
2011-04-20 22:52:22.545004 [DEBUG] mod_sofia.c:84 sofia/internal/9123 at 192.168.200.201 SOFIA INIT
2011-04-20 22:52:22.546004 [DEBUG] sofia_glue.c:1757 sofia/internal/9123 at 192.168.200.201 Patched SDP
---
v=0
o=- 12947784742458999 1 IN IP4 192.168.200.100
s=CounterPath X-Lite 4.0
c=IN IP4 192.168.200.100
t=0 0
a=ice-ufrag:260f31
a=ice-pwd:2110abb4ecb2953f4600b8f332fdb1d3
m=audio 61294 RTP/AVP 0 97 8 3 101
a=rtpmap:97 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=candidate:1 1 UDP 659136 192.168.200.100 61294 typ host
a=candidate:1 2 UDP 659134 192.168.200.100 61295 typ host
+++
v=0
o=FreeSWITCH 0459653255 0459653256 IN IP4 192.168.200.100
s=FreeSWITCH
c=IN IP4 192.168.200.100
t=0 0
a=ice-ufrag:260f31
a=ice-pwd:2110abb4ecb2953f4600b8f332fdb1d3
m=audio 26146 RTP/AVP 0 97 8 3 101
a=rtpmap:97 SPEEX/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=candidate:1 1 UDP 659136 192.168.200.100 61294 typ host
a=candidate:1 2 UDP 659134 192.168.200.100 61295 typ host
2011-04-20 22:52:22.546004 [DEBUG] mod_sofia.c:124 (sofia/internal/9123 at 192.168.200.201) State Change CS_INIT -> CS_ROUTING
2011-04-20 22:52:22.546004 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/9123 at 192.168.200.201 [BREAK]
2011-04-20 22:52:22.546004 [DEBUG] switch_core_state_machine.c:361 (sofia/internal/9123 at 192.168.200.201) State INIT going to sleep
2011-04-20 22:52:22.546004 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/9123 at 192.168.200.201) Running State Change CS_ROUTING
2011-04-20 22:52:22.546004 [DEBUG] switch_channel.c:1668 (sofia/internal/9123 at 192.168.200.201) Callstate Change DOWN -> RINGING
2011-04-20 22:52:22.546004 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/9123 at 192.168.200.201) State ROUTING
2011-04-20 22:52:22.546004 [DEBUG] mod_sofia.c:147 sofia/internal/9123 at 192.168.200.201 SOFIA ROUTING
2011-04-20 22:52:22.546004 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/9123 at 192.168.200.201) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2011-04-20 22:52:22.546004 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/9123 at 192.168.200.201 [BREAK]
2011-04-20 22:52:22.546004 [DEBUG] switch_core_state_machine.c:364 (sofia/internal/9123 at 192.168.200.201) State ROUTING going to sleep
2011-04-20 22:52:22.546004 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/9123 at 192.168.200.201) Running State Change CS_CONSUME_MEDIA
2011-04-20 22:52:22.546004 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/9123 at 192.168.200.201) State CONSUME_MEDIA
2011-04-20 22:52:22.546004 [DEBUG] switch_core_state_machine.c:383 (sofia/internal/9123 at 192.168.200.201) State CONSUME_MEDIA going to sleep
2011-04-20 22:52:22.547004 [DEBUG] sofia.c:4760 Channel sofia/internal/9123 at 192.168.200.201 entering state [calling][0]
2011-04-20 22:52:22.571006 [DEBUG] sofia.c:4760 Channel sofia/internal/9123 at 192.168.200.201 entering state [terminated][404]
2011-04-20 22:52:22.571006 [DEBUG] switch_channel.c:2563 (sofia/internal/9123 at 192.168.200.201) Callstate Change RINGING -> HANGUP
2011-04-20 22:52:22.571006 [NOTICE] sofia.c:5406 Hangup sofia/internal/9123 at 192.168.200.201 [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER]
2011-04-20 22:52:22.571006 [DEBUG] switch_channel.c:2579 Send signal sofia/internal/9123 at 192.168.200.201 [KILL]
2011-04-20 22:52:22.571006 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/9123 at 192.168.200.201 [BREAK]
2011-04-20 22:52:22.573006 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/9123 at 192.168.200.201) Running State Change CS_HANGUP
2011-04-20 22:52:22.575006 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/9123 at 192.168.200.201) State HANGUP
2011-04-20 22:52:22.575006 [DEBUG] mod_sofia.c:451 sofia/internal/9123 at 192.168.200.201 Overriding SIP cause 404 with 404 from the other leg
2011-04-20 22:52:22.575006 [DEBUG] mod_sofia.c:457 Channel sofia/internal/9123 at 192.168.200.201 hanging up, cause: UNALLOCATED_NUMBER
2011-04-20 22:52:22.579006 [DEBUG] switch_core_state_machine.c:46 sofia/internal/9123 at 192.168.200.201 Standard HANGUP, cause: UNALLOCATED_NUMBER
2011-04-20 22:52:22.579006 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/9123 at 192.168.200.201) State HANGUP going to sleep
2011-04-20 22:52:22.579006 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/9123 at 192.168.200.201) State Change CS_HANGUP -> CS_REPORTING
2011-04-20 22:52:22.579006 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/9123 at 192.168.200.201 [BREAK]
2011-04-20 22:52:22.579006 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/9123 at 192.168.200.201) Running State Change CS_REPORTING
2011-04-20 22:52:22.579006 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/9123 at 192.168.200.201) State REPORTING
2011-04-20 22:52:22.579006 [DEBUG] switch_core_state_machine.c:53 sofia/internal/9123 at 192.168.200.201 Standard REPORTING, cause: UNALLOCATED_NUMBER
2011-04-20 22:52:22.579006 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/9123 at 192.168.200.201) State REPORTING going to sleep
2011-04-20 22:52:22.579006 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/9123 at 192.168.200.201) State Change CS_REPORTING -> CS_DESTROY
2011-04-20 22:52:22.579006 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/9123 at 192.168.200.201 [BREAK]
2011-04-20 22:52:22.579006 [DEBUG] switch_core_session.c:1288 Session 14 (sofia/internal/9123 at 192.168.200.201) Locked, Waiting on external entities
2011-04-20 22:52:22.582006 [DEBUG] switch_ivr_originate.c:3492 Originate Resulted in Error Cause: 1 [UNALLOCATED_NUMBER]
2011-04-20 22:52:22.582006 [INFO] mod_dptools.c:2640 Originate Failed.  Cause: UNALLOCATED_NUMBER
2011-04-20 22:52:22.582006 [NOTICE] switch_core_session.c:1306 Session 14 (sofia/internal/9123 at 192.168.200.201) Ended
2011-04-20 22:52:22.582006 [NOTICE] switch_core_state_machine.c:189 sofia/internal/1002 at 192.168.200.100 has executed the last dialplan instruction, ha
nging up.
2011-04-20 22:52:22.582006 [DEBUG] switch_channel.c:2563 (sofia/internal/1002 at 192.168.200.100) Callstate Change RINGING -> HANGUP
2011-04-20 22:52:22.582006 [NOTICE] switch_core_state_machine.c:191 Hangup sofia/internal/1002 at 192.168.200.100 [CS_EXECUTE] [NORMAL_CLEARING]
2011-04-20 22:52:22.582006 [DEBUG] switch_channel.c:2579 Send signal sofia/internal/1002 at 192.168.200.100 [KILL]
2011-04-20 22:52:22.582006 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 at 192.168.200.100 [BREAK]
2011-04-20 22:52:22.582006 [DEBUG] switch_core_state_machine.c:371 (sofia/internal/1002 at 192.168.200.100) State EXECUTE going to sleep
2011-04-20 22:52:22.582006 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002 at 192.168.200.100) Running State Change CS_HANGUP
2011-04-20 22:52:22.582006 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/9123 at 192.168.200.201 [CS_DESTROY]
2011-04-20 22:52:22.585006 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1002 at 192.168.200.100) State HANGUP
2011-04-20 22:52:22.585006 [DEBUG] mod_sofia.c:451 sofia/internal/1002 at 192.168.200.100 Overriding SIP cause 480 with 404 from the other leg
2011-04-20 22:52:22.585006 [DEBUG] mod_sofia.c:457 Channel sofia/internal/1002 at 192.168.200.100 hanging up, cause: NORMAL_CLEARING
2011-04-20 22:52:22.585006 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/9123 at 192.168.200.201) Callstate Change HANGUP -> DOWN
2011-04-20 22:52:22.585006 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/9123 at 192.168.200.201) Running State Change CS_DESTROY
2011-04-20 22:52:22.586006 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/9123 at 192.168.200.201) State DESTROY
2011-04-20 22:52:22.586006 [DEBUG] mod_sofia.c:362 sofia/internal/9123 at 192.168.200.201 SOFIA DESTROY
2011-04-20 22:52:22.586006 [DEBUG] switch_core_state_machine.c:60 sofia/internal/9123 at 192.168.200.201 Standard DESTROY
2011-04-20 22:52:22.586006 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/9123 at 192.168.200.201) State DESTROY going to sleep
2011-04-20 22:52:22.591007 [DEBUG] mod_sofia.c:519 Responding to INVITE with: 404
2011-04-20 22:52:22.591007 [DEBUG] switch_core_state_machine.c:46 sofia/internal/1002 at 192.168.200.100 Standard HANGUP, cause: NORMAL_CLEARING
2011-04-20 22:52:22.591007 [DEBUG] switch_core_state_machine.c:565 (sofia/internal/1002 at 192.168.200.100) State HANGUP going to sleep
2011-04-20 22:52:22.591007 [DEBUG] switch_core_state_machine.c:356 (sofia/internal/1002 at 192.168.200.100) State Change CS_HANGUP -> CS_REPORTING
2011-04-20 22:52:22.591007 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 at 192.168.200.100 [BREAK]
2011-04-20 22:52:22.591007 [DEBUG] switch_core_state_machine.c:325 (sofia/internal/1002 at 192.168.200.100) Running State Change CS_REPORTING
2011-04-20 22:52:22.591007 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1002 at 192.168.200.100) State REPORTING
2011-04-20 22:52:22.785018 [DEBUG] switch_core_state_machine.c:53 sofia/internal/1002 at 192.168.200.100 Standard REPORTING, cause: NORMAL_CLEARING
2011-04-20 22:52:22.785018 [DEBUG] switch_core_state_machine.c:625 (sofia/internal/1002 at 192.168.200.100) State REPORTING going to sleep
2011-04-20 22:52:22.786018 [DEBUG] switch_core_state_machine.c:350 (sofia/internal/1002 at 192.168.200.100) State Change CS_REPORTING -> CS_DESTROY
2011-04-20 22:52:22.786018 [DEBUG] switch_core_session.c:1116 Send signal sofia/internal/1002 at 192.168.200.100 [BREAK]
2011-04-20 22:52:22.786018 [DEBUG] switch_core_session.c:1288 Session 13 (sofia/internal/1002 at 192.168.200.100) Locked, Waiting on external entities
2011-04-20 22:52:22.786018 [NOTICE] switch_core_session.c:1306 Session 13 (sofia/internal/1002 at 192.168.200.100) Ended
2011-04-20 22:52:22.786018 [NOTICE] switch_core_session.c:1308 Close Channel sofia/internal/1002 at 192.168.200.100 [CS_DESTROY]
2011-04-20 22:52:22.786018 [DEBUG] switch_core_state_machine.c:454 (sofia/internal/1002 at 192.168.200.100) Callstate Change HANGUP -> DOWN
2011-04-20 22:52:22.786018 [DEBUG] switch_core_state_machine.c:457 (sofia/internal/1002 at 192.168.200.100) Running State Change CS_DESTROY
2011-04-20 22:52:22.786018 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1002 at 192.168.200.100) State DESTROY
2011-04-20 22:52:22.786018 [DEBUG] mod_sofia.c:362 sofia/internal/1002 at 192.168.200.100 SOFIA DESTROY
2011-04-20 22:52:22.786018 [DEBUG] switch_core_state_machine.c:60 sofia/internal/1002 at 192.168.200.100 Standard DESTROY
2011-04-20 22:52:22.786018 [DEBUG] switch_core_state_machine.c:467 (sofia/internal/1002 at 192.168.200.100) State DESTROY going to sleep

From me at nevian.org  Wed Apr 20 18:59:04 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Wed, 20 Apr 2011 18:59:04 +0400
Subject: [Freeswitch-users] error loading mod_snmp
Message-ID: <1123621303311544@web68.yandex.ru>

Hi

2011-04-20 18:14:19.923216 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_snmp.so
**/usr/lib/libnssutil3.so: undefined symbol: PR_GetDirectorySeparator**

Last GIT, centos 5.5 32/64bit

I'm googled and found few  links to analogous problem with mod_spidermonkey and mod_xml_cURL. The only cure back to 2008 is to rebuild libcurl myself as it broken in Fedora..
Is this right or I'm doing something obvious wrong?

-- 
wbr,
Serge


From msc at freeswitch.org  Wed Apr 20 19:03:28 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Apr 2011 08:03:28 -0700
Subject: [Freeswitch-users] conference play / stop / PAUSE/RESUME?
In-Reply-To: <BANLkTik2nQ5S3sS6Zmnu81yQiARFeJPRMw@mail.gmail.com>
References: <BANLkTi=9672+WLZON0teCJLa-CFsg4E88w@mail.gmail.com>
	<BANLkTimgE00BQ9AC9O8jOtBtKz3Mv4je=Q@mail.gmail.com>
	<BANLkTin=Ps=ri8mVyH0HTgFe2pB4oD_Sxg@mail.gmail.com>
	<BANLkTik2nQ5S3sS6Zmnu81yQiARFeJPRMw@mail.gmail.com>
Message-ID: <BANLkTikoFXfx3Lj1igFZXR488VyBk5MD_g@mail.gmail.com>

On Tue, Apr 19, 2011 at 6:24 PM, Dale Trub <daletrub at gmail.com> wrote:

> Wow that's fantastic.
>
> > seek:<+[samples]>|<-[samples]>
>
> What are the parameters for samples?
>

Samples are the literally the number of samples in the file to jump forward
or backward. In an 8kHz file, 8000 samples would represent one second, in a
16kHz file 16000 samples would be one second, etc.

-MC
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From kris at kriskinc.com  Wed Apr 20 19:07:38 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 20 Apr 2011 11:07:38 -0400
Subject: [Freeswitch-users] SBC for carrier
In-Reply-To: <059f01cbff1d$86cb0e60$94612b20$@com>
References: <BANLkTinannjPrtj9oX=R7ugRydrJ-m6nfA@mail.gmail.com>
	<059f01cbff1d$86cb0e60$94612b20$@com>
Message-ID: <BANLkTi=Ni4Gnvs-emoSxPFMsqqygyZMSsQ@mail.gmail.com>

As a load balancer OpenSIPS or Kamailio could handle 120,000 calls
(depending on call setup rate) on a single machine.  You could then
direct the calls to a FreeSWITCH cluster for the full header
manipulation, RTP proxying, NAT traversal, transcoding, etc, etc that
commercial SBCs typically offer.  The number of calls per FreeSWITCH
machine will vary greatly depending on hardware and specific feature
set (transcoding is a killer, obviously).

Of course you'll want more than one machine (or group of machines) for
redundancy.

On Wed, Apr 20, 2011 at 1:40 AM, Skyler <skchopperguy at gmail.com> wrote:
> Why reply directly? I?d like to know too ;)
>
>
>
> ?My 0.02 would be a dual Quad-Core with at least 8gb Ram running OpenSips or
> Kamailio in stateless or transactional stateful mode. If you want it to
> handle NAT and RTP as well that?s easily done but more RAM would be required
> most likely. Of course FS distributed to handle the backend ;)
>
>
>
> Skyler
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> dome at tel.co.th
> Sent: Tuesday, April 19, 2011 9:25 PM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] SBC for carrier
>
>
>
> Dear all,
> ??? I'm freeswitch user :)? but looking for carrier grade SBC for my
> customer. like a acme , nexttone. they talking about 120,000
> concurrent. so if someone know about this please send me directly.
>
> Thanks.
>
> Dome C.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> ________________________________
>
> No virus found in this message.
> Checked by AVG - www.avg.com
> Version: 10.0.1209 / Virus Database: 1500/3584 - Release Date: 04/19/11
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 kris at kriskinc.com  Wed Apr 20 19:14:04 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 20 Apr 2011 11:14:04 -0400
Subject: [Freeswitch-users] FS -route to next GW if the first GW full
In-Reply-To: <BANLkTi=Pi9_gkv-WbkyijJH6-Kj1VgEuvg@mail.gmail.com>
References: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>
	<4dae8ae9.823d2b0a.02b3.155f@mx.google.com>
	<BANLkTino_bR7=BvhvMdhOWmbMVXcwFQJ5Q@mail.gmail.com>
	<BANLkTi=Pi9_gkv-WbkyijJH6-Kj1VgEuvg@mail.gmail.com>
Message-ID: <BANLkTin0fO=4N81-1iVsLotcR9F2-Ftzww@mail.gmail.com>

Try this:

<extension name="9_GW" >
  <condition field="destination_number" expression="^(9\d+)$">
    <action application="set" data="continue_on_fail=true"/>
    <action application="set" data="failure_causes=UNALLOCATED_NUMBER"/>
    <action application="set" data="hangup_after_bridge=true"/>
    <action application="bridge" data="sofia/internal/$1 at 192.168.200.201"/>
    <action application="bridge" data="sofia/internal/$1 at 192.168.200.202"/>
 </condition>
</extension>

On Wed, Apr 20, 2011 at 10:58 AM, fieldpeak <fieldpeak at gmail.com> wrote:
> Hi Steve,
>
> Thanks for your so rich stuff provided.
>
> however, i tried to use error code to route as below, it failed (did
> not route the next GW when recevied UNALLOCATED_NUMBER), can you
> please point out is there any place wrong in the dailplan? attached is
> the log, thanks.
>
> <extension name="9_GW" >
>
> ? <condition field="destination_number" expression="^(9\d+)$">
> ? ? ?<action application="set" data="continue_on_fail=true"/>
> ? ? <action application="set" data="hangup_after_bridge=false"/>
> ? ? ?<action application="bridge" data="sofia/internal/$1 at 192.168.200.201"/>
> ?</condition>
>
> ? <condition field="brige_hangup_cause" expression="UNALLOCATED_NUMBER">
> ? ?<action application="bridge" data="sofia/internal/$1 at 192.168.200.202"/>
> ? </condition>
>
> ?</extension>
>
> Regards,
> Charles
>
> 2011/4/20, Steven Ayre <steveayre at gmail.com>:
>> If you know the number of calls the GW can handle in advance, you can limit
>> the call attempts before sending the call to the gateway:
>> http://wiki.freeswitch.org/wiki/Limit
>> There are several ways to use that.
>>
>> You can reroute calls to a 2nd gateway on error:
>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover
>>
>> There are channel variables that let you control what to consider a
>> reroutable error and what is a give-up error:
>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>> http://wiki.freeswitch.org/wiki/Channel_Variables#failure_causes
>> This might also be useful, particularly with mod_limit:
>> http://wiki.freeswitch.org/wiki/Channel_Variables#transfer_on_fail
>>
>> You could use mod_lcr to get a list of all the GWs, but in a random order.
>> That'd let you load balance (randomly) but reroute when required without
>> duplicates unlike mod_distributor.
>>
>> Hopefully that's enough building blocks to give you somewhere to start...
>>
>> -Steve
>>
>>
>>
>> On 20 April 2011 08:27, Charles <fieldpeak at gmail.com> wrote:
>>
>>>
>>> i have two media GWs connected with FS for PSTN calls, FS route the first
>>> one in normal case, once the first GW get full (all of channels were
>>> occupied), i need FS route to the second GW.
>>> i found once the first GW get full, it will reply '404 not found' to FS,
>>> can FS route to the second one according to this condition or any other
>>> alternative?
>>>
>>> although i know there is mod_distributor for load balancing, however, i
>>> need if only first one full then route to second one, so it looks this
>>> module not suitable for this senario...
>>>
>>> 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
>
>



-- 
Kristian Kielhofner


From wstephen80 at gmail.com  Wed Apr 20 19:16:30 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Wed, 20 Apr 2011 17:16:30 +0200
Subject: [Freeswitch-users] FS -route to next GW if the first GW full
In-Reply-To: <BANLkTi=Pi9_gkv-WbkyijJH6-Kj1VgEuvg@mail.gmail.com>
References: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>
	<4dae8ae9.823d2b0a.02b3.155f@mx.google.com>
	<BANLkTino_bR7=BvhvMdhOWmbMVXcwFQJ5Q@mail.gmail.com>
	<BANLkTi=Pi9_gkv-WbkyijJH6-Kj1VgEuvg@mail.gmail.com>
Message-ID: <BANLkTin4iTimDJwxL_dxmiuCsnysQAKQfA@mail.gmail.com>

By default, when a "condition" block is completed (in your case after the
first bridge) then execution of dialplan breaks and the inbound call is
dropped.
You can change this default behaviour with: break="never" in the first
conditional block.
In this case also the second conditional block is always executed.
So your first conditional block will be:

<condition field="destination_number" expression="^(9\d+)$" break="never">
 ....

Stephen

On Wed, Apr 20, 2011 at 4:58 PM, fieldpeak <fieldpeak at gmail.com> wrote:

> Hi Steve,
>
> Thanks for your so rich stuff provided.
>
> however, i tried to use error code to route as below, it failed (did
> not route the next GW when recevied UNALLOCATED_NUMBER), can you
> please point out is there any place wrong in the dailplan? attached is
> the log, thanks.
>
> <extension name="9_GW" >
>
>   <condition field="destination_number" expression="^(9\d+)$">
>      <action application="set" data="continue_on_fail=true"/>
>     <action application="set" data="hangup_after_bridge=false"/>
>      <action application="bridge" data="sofia/internal/$1 at 192.168.200.201
> "/>
>  </condition>
>
>   <condition field="brige_hangup_cause" expression="UNALLOCATED_NUMBER">
>    <action application="bridge" data="sofia/internal/$1 at 192.168.200.202"/>
>   </condition>
>
>  </extension>
>
> Regards,
> Charles
>
> 2011/4/20, Steven Ayre <steveayre at gmail.com>:
> > If you know the number of calls the GW can handle in advance, you can
> limit
> > the call attempts before sending the call to the gateway:
> > http://wiki.freeswitch.org/wiki/Limit
> > There are several ways to use that.
> >
> > You can reroute calls to a 2nd gateway on error:
> >
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover
> >
> > There are channel variables that let you control what to consider a
> > reroutable error and what is a give-up error:
> > http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
> > http://wiki.freeswitch.org/wiki/Channel_Variables#failure_causes
> > This might also be useful, particularly with mod_limit:
> > http://wiki.freeswitch.org/wiki/Channel_Variables#transfer_on_fail
> >
> > You could use mod_lcr to get a list of all the GWs, but in a random
> order.
> > That'd let you load balance (randomly) but reroute when required without
> > duplicates unlike mod_distributor.
> >
> > Hopefully that's enough building blocks to give you somewhere to start...
> >
> > -Steve
> >
> >
> >
> > On 20 April 2011 08:27, Charles <fieldpeak at gmail.com> wrote:
> >
> >>
> >> i have two media GWs connected with FS for PSTN calls, FS route the
> first
> >> one in normal case, once the first GW get full (all of channels were
> >> occupied), i need FS route to the second GW.
> >> i found once the first GW get full, it will reply '404 not found' to FS,
> >> can FS route to the second one according to this condition or any other
> >> alternative?
> >>
> >> although i know there is mod_distributor for load balancing, however, i
> >> need if only first one full then route to second one, so it looks this
> >> module not suitable for this senario...
> >>
> >> 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
>
>
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From kbdfck at gmail.com  Wed Apr 20 19:26:56 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Wed, 20 Apr 2011 19:26:56 +0400
Subject: [Freeswitch-users] Disable IP change TRAP?
In-Reply-To: <BANLkTi=cx=TbokKLF7vG45bZiQBBqY12WA@mail.gmail.com>
References: <BANLkTik17Qtp-NqetDn=X+V3QCh1shD-bQ@mail.gmail.com>
	<BANLkTi=cx=TbokKLF7vG45bZiQBBqY12WA@mail.gmail.com>
Message-ID: <BANLkTin22CcSkWexVfJ2bcXG6BZNf0gV6Q@mail.gmail.com>

Thanks! I've already bound profiles to static IP, but didn't know about
<param name="auto-restart" value="false"/>.

2011/4/20 Rupa Schomaker <rupa at rupa.com>

>
> http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address
>
> On Wed, Apr 20, 2011 at 1:58 AM, Dmitry Sytchev <kbdfck at gmail.com> wrote:
> > Hi All!
> > How do I disable ip change trap and following profile restart?
> > I have two profiles, bound to separate ip addresses on eth0, both SIP and
> > RTP addresses. But when freeswitch detect IP change (about once a hour),
> it
> > tries to apply changes, but actually nothing changed as profiles are
> bound
> > to fixed ip addresses.
> > What can be done to make FS ignore this traps or at least not to touch
> > working profiles?
> > --
> > 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
> >
> >
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Best regards,

Dmitry Sytchev,
IT Engineer
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From Nabble at slickdeals.endjunk.com  Wed Apr 20 19:44:38 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 20 Apr 2011 08:44:38 -0700 (PDT)
Subject: [Freeswitch-users] Possible Google Voice -> FreeSWITCH directly?
Message-ID: <1303314278570-6291203.post@n2.nabble.com>

I have noticed additional e-mail addresses entered into a GMail account will
show up on GV Phone TAB as a Google Chat lines. In this case, I wonder if an
FS system is hosted on an e-mail server can be used to interact with GV to
place/receive GV calls. If not, then perhaps it is now a good project for FS
developers to find out what protocol/handshake is used by GV to forward an
incoming call to a Google Chat address and create/write a mod_gv to handle
GV calls directly on an FS system. Anyone?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Possible-Google-Voice-FreeSWITCH-directly-tp6291203p6291203.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Wed Apr 20 20:42:38 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Apr 2011 09:42:38 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today - VoIP
	Abuse/PBX Honeypot
Message-ID: <BANLkTin1viwge_kGSLEmh+M5SJq_jPUBqQ@mail.gmail.com>

Hello all! Our agenda is here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_04_20

At about 2PM EDT (11AM PDT) J. Oquendo will join us to talk about PBX
Honeypot and VoIP Abuse projects. We will still start the conference at 1PM
EDT/10AM PDT and do our agenda items until J. joins us.

Talk to you soon!

-Michael
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From msc at freeswitch.org  Wed Apr 20 20:49:59 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Apr 2011 09:49:59 -0700
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
 with productivity license.
In-Reply-To: <BANLkTimUkXz84DHh6bac4ognmYL1WzqdiQ@mail.gmail.com>
References: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local>
	<BANLkTimUkXz84DHh6bac4ognmYL1WzqdiQ@mail.gmail.com>
Message-ID: <BANLkTikzB_BzKstXrsY0dYTUiY80JNWA3A@mail.gmail.com>

On Tue, Apr 19, 2011 at 8:04 PM, Yehavi Bourvine
<yehavi.bourvine at gmail.com>wrote:

> Hello Michel,
>
>   Version 3.1.7 is very old. Try the 3.3.0 version (3.3.1F loses BLF after
> a while).
>

You can't do 3.3.1 on the 501 and 601 phones - they are oldies. Polycom has
"legacy" firmware for those old phones and you're stuck w/ what they give
you, which is indeed 3.1.7.

-MC
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From msc at freeswitch.org  Wed Apr 20 20:51:57 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 20 Apr 2011 09:51:57 -0700
Subject: [Freeswitch-users] Program Crash! What should i do ?
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58C50EE9C0@cooper>
References: <BANLkTik3t52A_H3PJp1L7bivxhQ74bWhUQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58C50EE9C0@cooper>
Message-ID: <BANLkTimcFC5fe+ivwdoefqMa5OX2Pt-0qA@mail.gmail.com>

On Wed, Apr 20, 2011 at 6:35 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> If I understand you correctly it's not an FS crash - it's in your own
> software?
>
> /Peter
>
>
Indeed, I think he needs to consult the "waihu dev list" :D
-MC
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From roger.castaldo at gmail.com  Wed Apr 20 21:13:04 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Wed, 20 Apr 2011 13:13:04 -0400
Subject: [Freeswitch-users] Possible Google Voice -> FreeSWITCH directly?
In-Reply-To: <1303314278570-6291203.post@n2.nabble.com>
References: <1303314278570-6291203.post@n2.nabble.com>
Message-ID: <BANLkTikgjM1RKKMqigmRRwur6K5gBrw91A@mail.gmail.com>

refer to a previous post on the freeswitch website
http://www.freeswitch.org/node/280

There should also be documentation on how to get freeswitch to link to
google voice in the wiki.

On Wed, Apr 20, 2011 at 11:44 AM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

> I have noticed additional e-mail addresses entered into a GMail account
> will
> show up on GV Phone TAB as a Google Chat lines. In this case, I wonder if
> an
> FS system is hosted on an e-mail server can be used to interact with GV to
> place/receive GV calls. If not, then perhaps it is now a good project for
> FS
> developers to find out what protocol/handshake is used by GV to forward an
> incoming call to a Google Chat address and create/write a mod_gv to handle
> GV calls directly on an FS system. Anyone?
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Possible-Google-Voice-FreeSWITCH-directly-tp6291203p6291203.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 boris at tagnet.ru  Wed Apr 20 21:21:02 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Wed, 20 Apr 2011 23:21:02 +0600
Subject: [Freeswitch-users] LUA directory script
Message-ID: <4DAF15FE.1050100@tagnet.ru>

Hello!

     May somebody help? I want to integrate directory with sql db via 
LUA script. By I can't authenticate user. Same user defined with xml 
file authenticates without problems.

The FS request is:

2011-04-20 23:15:34.543486 [ALERT] switch_cpp.cpp:1190 key_name name
2011-04-20 23:15:34.543486 [ALERT] switch_cpp.cpp:1190 key_value 
pbx.tagnet.hn
2011-04-20 23:15:34.543486 [ALERT] switch_cpp.cpp:1190 tag_name domain
2011-04-20 23:15:34.543486 [ALERT] switch_cpp.cpp:1190 section directory
2011-04-20 23:15:34.543486 [ALERT] switch_cpp.cpp:1190 'Event-Name: 
REQUEST_PARAMS
Core-UUID: df5b6d29-49f1-41d3-8eb0-aa7ad7ee611d
FreeSWITCH-Hostname: ipats.tagnet.ru
FreeSWITCH-IPv4: 80.64.16.19
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2011-04-20%2023%3A15%3A34
Event-Date-GMT: Wed,%2020%20Apr%202011%2017%3A15%3A34%20GMT
Event-Date-Timestamp: 1303319734543486
Event-Calling-File: sofia_reg.c
Event-Calling-Function: sofia_reg_parse_auth
Event-Calling-Line-Number: 2102
action: sip_auth
sip_profile: ipbx
sip_user_agent: SIPPER%20for%20PhonerLite
sip_auth_username: 73435230045
sip_auth_realm: pbx.tagnet.hn
sip_auth_nonce: 413ca6a6-84b8-41b0-89e4-43c7ed17abad
sip_auth_uri: sip%3Apbx.tagnet.hn
sip_contact_user: 73435230045
sip_contact_host: 192.168.3.253
sip_to_user: 73435230045
sip_to_host: pbx.tagnet.hn
sip_from_user: 73435230045
sip_from_host: pbx.tagnet.hn
sip_request_host: pbx.tagnet.hn
sip_auth_qop: auth
sip_auth_cnonce: 234abcc436e2667097e7fe6eia53e8dd
sip_auth_nc: 00000001
sip_auth_response: d388a1da67bdaa9206d0f6430f998c54
sip_auth_method: REGISTER
key: id
user: 73435230045
domain: pbx.tagnet.hn
ip: 192.168.3.253


The response is:

<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
<domain name="pbx.tagnet.hn">
<params><param name="dial-string" 
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
</params>
<variables><variable name="accountcode" value="ats001"/>
<variable name="domain" value="pbx.tagnet.hn"/>
<variable name="domain_name" value="pbx.tagnet.hn"/>
<variable name="user_context" value="pre_routing"/>
<variable name="v_tagnet_ats_dstport" value="0"/>
<variable name="v_tagnet_ats_srcport" value="0"/>
<variable name="v_tagnet_context" value="home.tagnet.hn"/>
</variables>
<groups>
<group name="default">
<users>
<user id="73435230045">
<params>
<param name="password" value="01234"/>
</params>
<variables>
<variable name="toll_allow" value="internal,local,domestic,international"/>
<variable name="user_context" value="pre_routing"/>
<variable name="v_tagnet_context" value="home.tagnet.hn"/>
</variables>
</user>
</users>
</group>
</groups>
</domain>
</document>

-- 
With respect,
Boris




From boris at tagnet.ru  Wed Apr 20 21:31:26 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Wed, 20 Apr 2011 23:31:26 +0600
Subject: [Freeswitch-users] LUA directory script
In-Reply-To: <4DAF15FE.1050100@tagnet.ru>
References: <4DAF15FE.1050100@tagnet.ru>
Message-ID: <4DAF186E.5000604@tagnet.ru>

Hello!

     Found I forgot <section name="directory">. Now user authenticates 
and another problem... my call is placed into the public context and 
should be pre_routing. And also there should be INVITE event, isn't? But 
I can't see it...

-- 
With respect,
Boris



From brian at freeswitch.org  Wed Apr 20 21:45:52 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Apr 2011 12:45:52 -0500
Subject: [Freeswitch-users] Polycom TLS
Message-ID: <1C72069C-FEA2-46D6-A220-9AE365489823@freeswitch.org>

Anyone here using Polycom with TLS on 3.3.1.0933?  I have a bug that I can't nail down nor can I get all the data required due to the phone locking up and rebooting before the complete tech dump is sent to syslog.  Just want to see if anyone else is having the issue where the phone will lockup/reboot.

Thanks,
Brian



From brian at freeswitch.org  Wed Apr 20 22:03:01 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Apr 2011 13:03:01 -0500
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
	with productivity license.
In-Reply-To: <BANLkTimUkXz84DHh6bac4ognmYL1WzqdiQ@mail.gmail.com>
References: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local>
	<BANLkTimUkXz84DHh6bac4ognmYL1WzqdiQ@mail.gmail.com>
Message-ID: <33A1CF05-5BEB-4902-BFE1-C3388FC9C6B5@freeswitch.org>

3.3.1F  also crashes left and right on TLS.

/b

On Apr 19, 2011, at 10:04 PM, Yehavi Bourvine wrote:

> Hello Michel,
> 
>  Version 3.1.7 is very old. Try the 3.3.0 version (3.3.1F loses BLF after a
> while).



From brian at freeswitch.org  Wed Apr 20 22:03:31 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Apr 2011 13:03:31 -0500
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
	with productivity license.
In-Reply-To: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local>
References: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local>
Message-ID: <E41BA92D-036B-4532-8754-EF37ED61D73C@freeswitch.org>

You should NEVER edit these files.  You should enable the features in the phone config.

/b

On Apr 19, 2011, at 2:39 PM, Michel Py wrote:

>  I enabled corporate directory in sip_317.cfg a



From yehavi.bourvine at gmail.com  Wed Apr 20 22:13:32 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Wed, 20 Apr 2011 21:13:32 +0300
Subject: [Freeswitch-users] Polycom TLS
In-Reply-To: <1C72069C-FEA2-46D6-A220-9AE365489823@freeswitch.org>
References: <1C72069C-FEA2-46D6-A220-9AE365489823@freeswitch.org>
Message-ID: <BANLkTim-pfo_0Mj0yO939BQ_DJ+gM_A61w@mail.gmail.com>

Hello Brian.

  I have a Polycom-450 running 3.3.1F and have some other issues with TLS:
It loses connection after a while and regains it back after a few seconds. I
did not have the time yet to produce debugging from the Freeswitch side.

                        __Yehavi:

2011/4/20 Brian West <brian at freeswitch.org>

> Anyone here using Polycom with TLS on 3.3.1.0933?  I have a bug that I
> can't nail down nor can I get all the data required due to the phone locking
> up and rebooting before the complete tech dump is sent to syslog.  Just want
> to see if anyone else is having the issue where the phone will
> lockup/reboot.
>
> Thanks,
> Brian
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From Nabble at slickdeals.endjunk.com  Wed Apr 20 22:31:48 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Wed, 20 Apr 2011 11:31:48 -0700 (PDT)
Subject: [Freeswitch-users] Possible Google Voice -> FreeSWITCH directly?
In-Reply-To: <BANLkTikgjM1RKKMqigmRRwur6K5gBrw91A@mail.gmail.com>
References: <1303314278570-6291203.post@n2.nabble.com>
	<BANLkTikgjM1RKKMqigmRRwur6K5gBrw91A@mail.gmail.com>
Message-ID: <1303324308583-6291781.post@n2.nabble.com>


Roger Castaldo wrote:
> refer to a previous post on the freeswitch website
> http://www.freeswitch.org/node/280
That uses a mod_dingaling (Gtalk -> FS) and is not a direct GV -> FS.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Possible-Google-Voice-FreeSWITCH-directly-tp6291203p6291781.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From freeswitch at peely.com  Wed Apr 20 22:32:00 2011
From: freeswitch at peely.com (peely)
Date: Wed, 20 Apr 2011 11:32:00 -0700 (PDT)
Subject: [Freeswitch-users] continue_on_fail for condition only where 100
	not received?
Message-ID: <1303324320582-6291782.post@n2.nabble.com>

Hi,

I can't seem to get a continue_on_fail code which would only retry if no
response whatsoever is received from an endpoint.

Does anyone know how I could route to a secondary endpoint ONLY if the
initial endpoint does not respond with 100? I guess as a secondary part of
the question I would need to know if I can change the INVITE timer to a
smaller amount so the second device is attempted in less than 60 seconds.

Basically I have a bunch of routing devices within the network I want to
route to with a failove option, these devices onward route so I trust if
that device is OK not to route to the second device, given these devices log
sessions themselves it's vital that if one device receives an INVITE none of
the others do for the same session.


Thanks,




Neil.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/continue-on-fail-for-condition-only-where-100-not-received-tp6291782p6291782.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From brian at freeswitch.org  Wed Apr 20 22:51:52 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Apr 2011 13:51:52 -0500
Subject: [Freeswitch-users] Polycom TLS
In-Reply-To: <BANLkTim-pfo_0Mj0yO939BQ_DJ+gM_A61w@mail.gmail.com>
References: <1C72069C-FEA2-46D6-A220-9AE365489823@freeswitch.org>
	<BANLkTim-pfo_0Mj0yO939BQ_DJ+gM_A61w@mail.gmail.com>
Message-ID: <42657A07-296F-4CBC-B12C-B4F845AA5D65@freeswitch.org>

I'm seeing that exact thing.

/b

On Apr 20, 2011, at 1:13 PM, Yehavi Bourvine wrote:

> Hello Brian.
> 
>  I have a Polycom-450 running 3.3.1F and have some other issues with TLS:
> It loses connection after a while and regains it back after a few seconds. I
> did not have the time yet to produce debugging from the Freeswitch side.
> 
>                        __Yehavi:



From brian at freeswitch.org  Wed Apr 20 22:53:37 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Apr 2011 13:53:37 -0500
Subject: [Freeswitch-users] Polycom TLS
In-Reply-To: <BANLkTim-pfo_0Mj0yO939BQ_DJ+gM_A61w@mail.gmail.com>
References: <1C72069C-FEA2-46D6-A220-9AE365489823@freeswitch.org>
	<BANLkTim-pfo_0Mj0yO939BQ_DJ+gM_A61w@mail.gmail.com>
Message-ID: <1E6E3DB8-F7B3-422B-A1D9-3F89B703661F@freeswitch.org>

What TLS Version are you using in your sofia profile?

/b



From steveayre at gmail.com  Wed Apr 20 23:19:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 20 Apr 2011 20:19:01 +0100
Subject: [Freeswitch-users] Possible Google Voice -> FreeSWITCH directly?
In-Reply-To: <1303324308583-6291781.post@n2.nabble.com>
References: <1303314278570-6291203.post@n2.nabble.com>
	<BANLkTikgjM1RKKMqigmRRwur6K5gBrw91A@mail.gmail.com>
	<1303324308583-6291781.post@n2.nabble.com>
Message-ID: <BANLkTikqrNUMkDQGt3Z7SpGVBZmSN2h_-Q@mail.gmail.com>

"Google Voice permits Voice Over IP (VoIP) connections through Gmail or
Google Talk, but offers no simple way to communicate with users of other
VoIP services, e.g. by direct connection between IP addresses or SIP
gateway."
http://en.wikipedia.org/wiki/Google_voice#VoIP_services

The only way for FS to use GV is via Google Talk.

-Steve


On 20 April 2011 19:31, mazilo <Nabble at slickdeals.endjunk.com> wrote:

>
> Roger Castaldo wrote:
> > refer to a previous post on the freeswitch website
> > http://www.freeswitch.org/node/280
> That uses a mod_dingaling (Gtalk -> FS) and is not a direct GV -> FS.
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Possible-Google-Voice-FreeSWITCH-directly-tp6291203p6291781.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 andrew.keil at askinteractive.net  Wed Apr 20 23:47:05 2011
From: andrew.keil at askinteractive.net (Andrew Keil)
Date: Thu, 21 Apr 2011 05:47:05 +1000
Subject: [Freeswitch-users] Where is mod_spidermonkey_etpan?
In-Reply-To: <1303310247416-6290956.post@n2.nabble.com>
References: <C42A22554C0CF44DAC0AA1F132B86DE657F7EA38F3@ASKSERVER.asksbs.local>
	<1303310247416-6290956.post@n2.nabble.com>
Message-ID: <C42A22554C0CF44DAC0AA1F132B86DE657F7EA38F9@ASKSERVER.asksbs.local>

Understood.  I have since found and used msmtp.exe (http://msmtp.sourceforge.net/) instead (via a system call) and that seems to work fine for my purposes at the moment.

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk
Sent: Thursday, 21 April 2011 12:37 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Where is mod_spidermonkey_etpan?

That module was removed sometime ago - patches excepted to add it back if you are willing to help maintain it.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Where-is-mod-spidermonkey-etpan-tp6289652p6290956.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
 

__________ Information from ESET NOD32 Antivirus, version of virus signature database 6059 (20110420) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
 


From fraserredmond at gmail.com  Thu Apr 21 00:04:34 2011
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 20 Apr 2011 16:04:34 -0400
Subject: [Freeswitch-users] one-way audio problem on some inbound
 gateways, but not others (and no outbound gateways)
In-Reply-To: <4A1D272B-E12C-4CF5-9149-4DED32E88CCB@freeswitch.org>
References: <BANLkTinuTg+DwXQ5_fU1K-mE+oqYgH7apQ@mail.gmail.com>
	<4A1D272B-E12C-4CF5-9149-4DED32E88CCB@freeswitch.org>
Message-ID: <BANLkTi=yyL4f-q65Du8e0VRy27y0zAoi6Q@mail.gmail.com>

Thanks Brian, I'd appreciate you looking - I don't know what to look for in
the sip traces (could be worth documenting some pointers in the wiki?)

The sip trace is here:
http://pastebin.freeswitch.org/16136

I pressed enter a few times in the console before and after it connected to
the extension, so about lines 490-600 is the relevant part.

I also captured a pcap, in case that is of interest - let me know and I'll
email it directly.

Thanks,
Fraser




On Sun, Apr 17, 2011 at 12:39 AM, Brian Weat <brian at freeswitch.org> wrote:

> have a sip trace?
>
> Sent from my iPhone
>
> On Apr 16, 2011, at 11:34 PM, Fraser Redmond <fraserredmond at gmail.com>
> wrote:
>
> > Any ideas of what to try?
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 rodrigo.ferrari at cellmidia.com.br  Thu Apr 21 00:21:58 2011
From: rodrigo.ferrari at cellmidia.com.br (Rodrigo Ferrari)
Date: Wed, 20 Apr 2011 17:21:58 -0300
Subject: [Freeswitch-users] Khomp GSM not receiving messages
In-Reply-To: <BANLkTik84Ga-cUwt69=7P4RtAWHcfOca-Q@mail.gmail.com>
References: <BANLkTik84Ga-cUwt69=7P4RtAWHcfOca-Q@mail.gmail.com>
Message-ID: <BANLkTi=zKAT2EiWAzAAG-_vie5PiDtsKDg@mail.gmail.com>

Hello friends,

Sorry for my bad english.

I installed a khomp GSM board, set up 2 chips, b0c0 and b0c1. At the
dialplan I configured this situation:

    <context name="khomp-00-00">
      <extension name="exemplo_1">
        <condition field="destination_number" expression="^s$">
          <action application="log" data="DEBUG Recieving CALL ..." />
          <action application="log" data="De ${caller_id_number}"/>

          <action application="curl" data="
http://app.cellmidia.com.br/receiver/receiver.php headers post
teste=${caller_id_number}&teste2=Chamada"/>
          <action application="info"/>

        </condition>
      </extension>
    </context>

    <context name="khomp-sms-00-00">
      <extension name="sms">
        <condition field="destination_number" expression="^s$">
          <action application="curl" data="
http://app.cellmidia.com.br/receiver/receiver.php headers post
teste=SMS&teste2=Entrante"/>
          <action application="info"/>
        </condition>
      </extension>
    </context>

If I call it will do what is set to do, but if I send a message, it will not
operate, there is no log set or any event marked as happining. Is there some
configuration that I miss? Some setting to my khomp board be able to receive
messages.

Thanks, best regard's.

Rodrigo Ferrari.
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From brian at freeswitch.org  Thu Apr 21 01:46:46 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Apr 2011 16:46:46 -0500
Subject: [Freeswitch-users] one-way audio problem on some inbound
	gateways, but not others (and no outbound gateways)
In-Reply-To: <BANLkTi=yyL4f-q65Du8e0VRy27y0zAoi6Q@mail.gmail.com>
References: <BANLkTinuTg+DwXQ5_fU1K-mE+oqYgH7apQ@mail.gmail.com>
	<4A1D272B-E12C-4CF5-9149-4DED32E88CCB@freeswitch.org>
	<BANLkTi=yyL4f-q65Du8e0VRy27y0zAoi6Q@mail.gmail.com>
Message-ID: <1A68D020-D97F-43E6-B83B-E3C762DAD665@freeswitch.org>

firewall issue? mic doesn't work?  can you get a pcap of all traffic?

/b

On Apr 20, 2011, at 3:04 PM, Fraser Redmond wrote:

> Thanks Brian, I'd appreciate you looking - I don't know what to look for in
> the sip traces (could be worth documenting some pointers in the wiki?)
> 
> The sip trace is here:
> http://pastebin.freeswitch.org/16136
> 
> I pressed enter a few times in the console before and after it connected to
> the extension, so about lines 490-600 is the relevant part.
> 
> I also captured a pcap, in case that is of interest - let me know and I'll
> email it directly.
> 
> Thanks,
> Fraser

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From Prometheus001 at gmx.net  Thu Apr 21 03:00:48 2011
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 21 Apr 2011 01:00:48 +0200
Subject: [Freeswitch-users] SPIT attack and how to strike back
Message-ID: <4DAF65A0.3070600@gmx.net>

Hello all,

I would like to share this with you as you may have also been affected
by this threat.

Yesterday we received a SPIT attack to our Freeswitch servers. We had
about 50 register requests/sec. We noticed this as we saw a slight
increase in the load of the Freeswitch servers. Fortunately Freeswitch
can handle a huge amount of register requests so we had no denial of
service.

You can identify this attack by finding the following in the Register
message:
    User-Agent: friendly-scanner

How to get rid of it:
The attacker used Sipvicious (friendly-scanner). Sipvicious itself has a
nice tool "svcrash.py" wich can send a malformed packet back to the
attacker which crashes their own Sipvicious tool. You can issue this tool by
  python svcrash.py -d <host of attacker> -p <port of attacker>
You will need port 5060 on your machine to work. But there is also a
workaround for that. svcrash.py will show how to overcome this if your
port 5060 is not available.
Download it here
http://sipvicious.googlecode.com/files/sipvicious-0.2.6.tar.gz and
unpack it to a folder of your choice.

I wrote a small Ruby script to send the packet back to a port range, as
our attacker used some dozens of ports to send.
Here is the script (Install ruby first by "apt-get install ruby" e.g. on
Debian based systems). Put it into the sipvicious directory
kill_ports.rb:

#!/usr/bin/env ruby
host=ARGV[0]
start_port=ARGV[1].to_i
end_port=ARGV[2].to_i
start_port.upto(end_port) do |port|
  cmd="python svcrash.py -d #{host} -p #{port}"
  p cmd
  erg=`#{cmd}`
  p erg
end

You now can run it by
./kill_ports.rb <ip> <from_port> <to_port>

By using this tool we got rid of most of the SPIT messages. But after a
while they started again to attack us from different ports.

The next step is: Why not automate this by trying to identify host and
port automatically and send back the svcrash.py packet to the sender's port?

First install the pcap library
    apt-get install libpcap-dev libpcap-ruby

Then I wrote the following tool to automate this, it makes use of the
kill_ports.rb above:
strike_back.rb:

#!/usr/bin/env ruby
# I used some code from http://snippets.dzone.com/posts/show/5931
require 'pcaplet'
require 'logger'
require 'timeout'
@timeout=3600 # max runtime: 1 hour

@logfile='strike_back.log'
class AuditLogger < Logger
  def format_message(severity, timestamp, progname, msg)
    puts msg
    "#{msg}\n"
  end
end

logfile = File.open(@logfile, 'a')
LOGGER = AuditLogger.new(logfile)
LOGGER.level = Logger::INFO
search="friendly-scanner"
puts"Searching for '#{ search}' in SIP packets"
$network = Pcaplet.new('-s 1500')
$filter = Pcap::Filter.new('udp and dst port 5060', $network.capture)
$network.add_filter($filter)
puts "Logfile: #{@logfile}"
puts "Starting capture..."
begin
  Timeout.timeout(@timeout) do # 3600 sec
    for p in $network
        header= "#{Time.now.strftime("%Y-%m-%d %H:%M:%S")}
#{p.src}:#{p.sport} => #{p.dst}:#{p.dport}"
        if $filter =~ p
            #puts "simple search"
            if p.udp_data.index(search)
              LOGGER.info "Kill Friendly scanner #{p.src} with Source
Port #{p.sport}"
              cmd="./kill_ports.rb #{p.src} #{p.sport} #{p.sport}"
              erg=`#{cmd}`
              p erg
              LOGGER.info header
              LOGGER.info p.udp_data
            end
        end
    end
  end
rescue Timeout::Error
  logfile.flush
  puts "Timeout - finished."
end

There may be a better way to code this, but at least it worked. After
about 15min the number of attacks went to 0.

Disclaimer: You can damage other systems by using these tools. So be
carefull and use at your own risks. Do not use this tool for attacking
other systems!

Best regards
Peter


From fraserredmond at gmail.com  Thu Apr 21 03:30:54 2011
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 20 Apr 2011 19:30:54 -0400
Subject: [Freeswitch-users] one-way audio problem on some inbound
 gateways, but not others (and no outbound gateways)
In-Reply-To: <1A68D020-D97F-43E6-B83B-E3C762DAD665@freeswitch.org>
References: <BANLkTinuTg+DwXQ5_fU1K-mE+oqYgH7apQ@mail.gmail.com>
	<4A1D272B-E12C-4CF5-9149-4DED32E88CCB@freeswitch.org>
	<BANLkTi=yyL4f-q65Du8e0VRy27y0zAoi6Q@mail.gmail.com>
	<1A68D020-D97F-43E6-B83B-E3C762DAD665@freeswitch.org>
Message-ID: <BANLkTi=p7=xZ_GziBTYegM=J3MkEOFRVCw@mail.gmail.com>

No, I tried turning off the firewall, and as I said in the OP it works with
one of our other gateways.

Mic works on that one gateway, and during the calls where the audio isn't
transmitted the mic-indicator goes up and down.

Cheers,
Fraser




On Wed, Apr 20, 2011 at 5:46 PM, Brian West <brian at freeswitch.org> wrote:

> firewall issue? mic doesn't work?  can you get a pcap of all traffic?
>
> /b
>
> On Apr 20, 2011, at 3:04 PM, Fraser Redmond wrote:
>
> Thanks Brian, I'd appreciate you looking - I don't know what to look for in
> the sip traces (could be worth documenting some pointers in the wiki?)
>
> The sip trace is here:
> http://pastebin.freeswitch.org/16136
>
> I pressed enter a few times in the console before and after it connected to
> the extension, so about lines 490-600 is the relevant part.
>
> I also captured a pcap, in case that is of interest - let me know and I'll
> email it directly.
>
> Thanks,
> Fraser
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 fraserredmond at gmail.com  Thu Apr 21 03:37:28 2011
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 20 Apr 2011 19:37:28 -0400
Subject: [Freeswitch-users] one-way audio problem on some inbound
 gateways, but not others (and no outbound gateways)
In-Reply-To: <BANLkTi=p7=xZ_GziBTYegM=J3MkEOFRVCw@mail.gmail.com>
References: <BANLkTinuTg+DwXQ5_fU1K-mE+oqYgH7apQ@mail.gmail.com>
	<4A1D272B-E12C-4CF5-9149-4DED32E88CCB@freeswitch.org>
	<BANLkTi=yyL4f-q65Du8e0VRy27y0zAoi6Q@mail.gmail.com>
	<1A68D020-D97F-43E6-B83B-E3C762DAD665@freeswitch.org>
	<BANLkTi=p7=xZ_GziBTYegM=J3MkEOFRVCw@mail.gmail.com>
Message-ID: <BANLkTinQKCPVJm7ASsaCLpw9SMzOpDytsQ@mail.gmail.com>

Hi Brian,

Here's the pcap file.

BTW, your iphone email has your name spelt wrong ("Brian Weat"), made me
wonder if there was a new Brian at Freeswitch :-)

Cheers,
Fraser




On Wed, Apr 20, 2011 at 7:30 PM, Fraser Redmond <fraserredmond at gmail.com>wrote:

> No, I tried turning off the firewall, and as I said in the OP it works with
> one of our other gateways.
>
> Mic works on that one gateway, and during the calls where the audio isn't
> transmitted the mic-indicator goes up and down.
>
> Cheers,
> Fraser
>
>
>
>
> On Wed, Apr 20, 2011 at 5:46 PM, Brian West <brian at freeswitch.org> wrote:
>
>> firewall issue? mic doesn't work?  can you get a pcap of all traffic?
>>
>> /b
>>
>> On Apr 20, 2011, at 3:04 PM, Fraser Redmond wrote:
>>
>> Thanks Brian, I'd appreciate you looking - I don't know what to look for
>> in
>> the sip traces (could be worth documenting some pointers in the wiki?)
>>
>> The sip trace is here:
>> http://pastebin.freeswitch.org/16136
>>
>> I pressed enter a few times in the console before and after it connected
>> to
>> the extension, so about lines 490-600 is the relevant part.
>>
>> I also captured a pcap, in case that is of interest - let me know and I'll
>> email it directly.
>>
>> Thanks,
>> Fraser
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 sos at sokhapkin.dyndns.org  Thu Apr 21 03:42:45 2011
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Wed, 20 Apr 2011 19:42:45 -0400
Subject: [Freeswitch-users] one-way audio problem on some inbound
	gateways, but not others (and no outbound gateways)
In-Reply-To: <BANLkTi=p7=xZ_GziBTYegM=J3MkEOFRVCw@mail.gmail.com>
References: <BANLkTinuTg+DwXQ5_fU1K-mE+oqYgH7apQ@mail.gmail.com>
	<1A68D020-D97F-43E6-B83B-E3C762DAD665@freeswitch.org>
	<BANLkTi=p7=xZ_GziBTYegM=J3MkEOFRVCw@mail.gmail.com>
Message-ID: <201104201942.45050.sos@sokhapkin.dyndns.org>

No outgoing audio? Usually this happens if SIP ALG is enabled in your router.

On Wednesday 20 April 2011, Fraser Redmond wrote:
> No, I tried turning off the firewall, and as I said in the OP it works with
> one of our other gateways.
> 
> Mic works on that one gateway, and during the calls where the audio isn't
> transmitted the mic-indicator goes up and down.
> 
> Cheers,
> Fraser
> 
> On Wed, Apr 20, 2011 at 5:46 PM, Brian West <brian at freeswitch.org> wrote:
> > firewall issue? mic doesn't work?  can you get a pcap of all traffic?
> > 
> > /b
> > 
> > On Apr 20, 2011, at 3:04 PM, Fraser Redmond wrote:
> > 
> > Thanks Brian, I'd appreciate you looking - I don't know what to look for
> > in the sip traces (could be worth documenting some pointers in the
> > wiki?)
> > 
> > The sip trace is here:
> > http://pastebin.freeswitch.org/16136
> > 
> > I pressed enter a few times in the console before and after it connected
> > to the extension, so about lines 490-600 is the relevant part.
> > 
> > I also captured a pcap, in case that is of interest - let me know and
> > I'll email it directly.
> > 
> > Thanks,
> > Fraser
> > 
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org



From fraserredmond at gmail.com  Thu Apr 21 05:16:14 2011
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Wed, 20 Apr 2011 21:16:14 -0400
Subject: [Freeswitch-users] one-way audio problem on some inbound
 gateways, but not others (and no outbound gateways)
In-Reply-To: <201104201942.45050.sos@sokhapkin.dyndns.org>
References: <BANLkTinuTg+DwXQ5_fU1K-mE+oqYgH7apQ@mail.gmail.com>
	<1A68D020-D97F-43E6-B83B-E3C762DAD665@freeswitch.org>
	<BANLkTi=p7=xZ_GziBTYegM=J3MkEOFRVCw@mail.gmail.com>
	<201104201942.45050.sos@sokhapkin.dyndns.org>
Message-ID: <BANLkTi=5nROvHn=MU0J7xJX3THAvnwZ+4A@mail.gmail.com>

But it's odd that it's only happening with some gateways, but other gateways
get two-way audio fine. That's what really has me puzzled.

It's definitely not a problem at the extension side, as it happens with
multiple phones, and multiple locations. (So not mic, firewall, router)

Cheers,
Fraser




On Wed, Apr 20, 2011 at 7:42 PM, Sergey Okhapkin
<sos at sokhapkin.dyndns.org>wrote:

> No outgoing audio? Usually this happens if SIP ALG is enabled in your
> router.
>
> On Wednesday 20 April 2011, Fraser Redmond wrote:
> > No, I tried turning off the firewall, and as I said in the OP it works
> with
> > one of our other gateways.
> >
> > Mic works on that one gateway, and during the calls where the audio isn't
> > transmitted the mic-indicator goes up and down.
> >
> > Cheers,
> > Fraser
> >
> > On Wed, Apr 20, 2011 at 5:46 PM, Brian West <brian at freeswitch.org>
> wrote:
> > > firewall issue? mic doesn't work?  can you get a pcap of all traffic?
> > >
> > > /b
> > >
> > > On Apr 20, 2011, at 3:04 PM, Fraser Redmond wrote:
> > >
> > > Thanks Brian, I'd appreciate you looking - I don't know what to look
> for
> > > in the sip traces (could be worth documenting some pointers in the
> > > wiki?)
> > >
> > > The sip trace is here:
> > > http://pastebin.freeswitch.org/16136
> > >
> > > I pressed enter a few times in the console before and after it
> connected
> > > to the extension, so about lines 490-600 is the relevant part.
> > >
> > > I also captured a pcap, in case that is of interest - let me know and
> > > I'll email it directly.
> > >
> > > Thanks,
> > > Fraser
> > >
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 brian at freeswitch.org  Thu Apr 21 05:22:44 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 20 Apr 2011 20:22:44 -0500
Subject: [Freeswitch-users] SPIT attack and how to strike back
In-Reply-To: <4DAF65A0.3070600@gmx.net>
References: <4DAF65A0.3070600@gmx.net>
Message-ID: <8906FDAD-11A2-4609-9676-EFFB1179973D@freeswitch.org>

Or 

iptables -I INPUT -j DROP -p udp --dport 5060 -m string --string "friendly-scanner" --algo bm

/b

On Apr 20, 2011, at 6:00 PM, Peter P GMX wrote:

> Hello all,
> 
> I would like to share this with you as you may have also been affected
> by this threat.
> 
> Yesterday we received a SPIT attack to our Freeswitch servers. We had
> about 50 register requests/sec. We noticed this as we saw a slight
> increase in the load of the Freeswitch servers. Fortunately Freeswitch
> can handle a huge amount of register requests so we had no denial of
> service.
> 
> You can identify this attack by finding the following in the Register
> message:
>    User-Agent: friendly-scanner
> 
> How to get rid of it:
> The attacker used Sipvicious (friendly-scanner). Sipvicious itself has a
> nice tool "svcrash.py" wich can send a malformed packet back to the
> attacker which crashes their own Sipvicious tool. You can issue this tool by
>  python svcrash.py -d <host of attacker> -p <port of attacker>
> You will need port 5060 on your machine to work. But there is also a
> workaround for that. svcrash.py will show how to overcome this if your
> port 5060 is not available.
> Download it here
> http://sipvicious.googlecode.com/files/sipvicious-0.2.6.tar.gz and
> unpack it to a folder of your choice.
> 
> I wrote a small Ruby script to send the packet back to a port range, as
> our attacker used some dozens of ports to send.
> Here is the script (Install ruby first by "apt-get install ruby" e.g. on
> Debian based systems). Put it into the sipvicious directory
> kill_ports.rb:
> 
> #!/usr/bin/env ruby
> host=ARGV[0]
> start_port=ARGV[1].to_i
> end_port=ARGV[2].to_i
> start_port.upto(end_port) do |port|
>  cmd="python svcrash.py -d #{host} -p #{port}"
>  p cmd
>  erg=`#{cmd}`
>  p erg
> end
> 
> You now can run it by
> ./kill_ports.rb <ip> <from_port> <to_port>
> 
> By using this tool we got rid of most of the SPIT messages. But after a
> while they started again to attack us from different ports.
> 
> The next step is: Why not automate this by trying to identify host and
> port automatically and send back the svcrash.py packet to the sender's port?
> 
> First install the pcap library
>    apt-get install libpcap-dev libpcap-ruby
> 
> Then I wrote the following tool to automate this, it makes use of the
> kill_ports.rb above:
> strike_back.rb:
> 
> #!/usr/bin/env ruby
> # I used some code from http://snippets.dzone.com/posts/show/5931
> require 'pcaplet'
> require 'logger'
> require 'timeout'
> @timeout=3600 # max runtime: 1 hour
> 
> @logfile='strike_back.log'
> class AuditLogger < Logger
>  def format_message(severity, timestamp, progname, msg)
>    puts msg
>    "#{msg}\n"
>  end
> end
> 
> logfile = File.open(@logfile, 'a')
> LOGGER = AuditLogger.new(logfile)
> LOGGER.level = Logger::INFO
> search="friendly-scanner"
> puts"Searching for '#{ search}' in SIP packets"
> $network = Pcaplet.new('-s 1500')
> $filter = Pcap::Filter.new('udp and dst port 5060', $network.capture)
> $network.add_filter($filter)
> puts "Logfile: #{@logfile}"
> puts "Starting capture..."
> begin
>  Timeout.timeout(@timeout) do # 3600 sec
>    for p in $network
>        header= "#{Time.now.strftime("%Y-%m-%d %H:%M:%S")}
> #{p.src}:#{p.sport} => #{p.dst}:#{p.dport}"
>        if $filter =~ p
>            #puts "simple search"
>            if p.udp_data.index(search)
>              LOGGER.info "Kill Friendly scanner #{p.src} with Source
> Port #{p.sport}"
>              cmd="./kill_ports.rb #{p.src} #{p.sport} #{p.sport}"
>              erg=`#{cmd}`
>              p erg
>              LOGGER.info header
>              LOGGER.info p.udp_data
>            end
>        end
>    end
>  end
> rescue Timeout::Error
>  logfile.flush
>  puts "Timeout - finished."
> end
> 
> There may be a better way to code this, but at least it worked. After
> about 15min the number of attacks went to 0.
> 
> Disclaimer: You can damage other systems by using these tools. So be
> carefull and use at your own risks. Do not use this tool for attacking
> other systems!
> 
> Best regards
> Peter
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From elijah at crankenstein.com  Thu Apr 21 06:13:12 2011
From: elijah at crankenstein.com (elijah)
Date: Wed, 20 Apr 2011 19:13:12 -0700
Subject: [Freeswitch-users] attended transfer to gateway
In-Reply-To: <BANLkTikkF95DCGgpQJ-kKCR-U_=X7fF5+g@mail.gmail.com>
References: <BANLkTinB6uFAWDnq-Ot1tiSPi99pu0Oruw@mail.gmail.com>
	<BANLkTikkF95DCGgpQJ-kKCR-U_=X7fF5+g@mail.gmail.com>
Message-ID: <BANLkTinAXp_u4+0OwNjY5dEDH5YHmFR-fw@mail.gmail.com>

Ok, here's the log: http://pastebin.freeswitch.org/16144
<http://pastebin.freeswitch.org/16144>Any input would be great - I'm not
seeing the condition that is causing 0 to drop the c-leg after an attended
bridge.

thanks,
elijah

On Tue, Apr 19, 2011 at 12:39 PM, Michael Collins <msc at freeswitch.org>wrote:

> Get a console log with full siptrace (on both internal and external
> profiles) and drop it into pastebin. Be sure to specific "FreeSWITCH log"
> for syntax highlighting to make it easier to read. Send the pb link here.
> Hopefully you or another user will be able to parse the logs to see what is
> happening.
>
> -MC
>
> On Tue, Apr 19, 2011 at 11:49 AM, elijah <elijah at crankenstein.com> wrote:
>
>> Attended transfers to an external c-leg is breaking 3-way conference
>> functionality for me. Like this:
>>
>> <action application="att_xfer"
>> data="sofia/gateway/onesource/${attxfer_callthis}"/>
>>
>> The c-leg is established and hanging up the originator's leg will
>> successfully bridge the other 2. However, pressing 0 on the originator,
>> which should establish a 3-way conference, actually hangs up the c-leg. Is
>> there another method by which I should specify the outbound leg for
>> att_xfer?
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 21 06:45:07 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 21 Apr 2011 10:45:07 +0800
Subject: [Freeswitch-users] FS -route to next GW if the first GW full
In-Reply-To: <BANLkTin0fO=4N81-1iVsLotcR9F2-Ftzww@mail.gmail.com>
References: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>
	<4dae8ae9.823d2b0a.02b3.155f@mx.google.com>
	<BANLkTino_bR7=BvhvMdhOWmbMVXcwFQJ5Q@mail.gmail.com>
	<BANLkTi=Pi9_gkv-WbkyijJH6-Kj1VgEuvg@mail.gmail.com>
	<BANLkTin0fO=4N81-1iVsLotcR9F2-Ftzww@mail.gmail.com>
Message-ID: <BANLkTimSG_M+q8G05W7V2SwdUgV0w938xQ@mail.gmail.com>

Thanks All, it is resolved now with below config.

2011/4/20, Kristian Kielhofner <kris at kriskinc.com>:
> Try this:
>
> <extension name="9_GW" >
>   <condition field="destination_number" expression="^(9\d+)$">
>     <action application="set" data="continue_on_fail=true"/>
>     <action application="set" data="failure_causes=UNALLOCATED_NUMBER"/>
>     <action application="set" data="hangup_after_bridge=true"/>
>     <action application="bridge" data="sofia/internal/$1 at 192.168.200.201"/>
>     <action application="bridge" data="sofia/internal/$1 at 192.168.200.202"/>
>  </condition>
> </extension>
>
> On Wed, Apr 20, 2011 at 10:58 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>> Hi Steve,
>>
>> Thanks for your so rich stuff provided.
>>
>> however, i tried to use error code to route as below, it failed (did
>> not route the next GW when recevied UNALLOCATED_NUMBER), can you
>> please point out is there any place wrong in the dailplan? attached is
>> the log, thanks.
>>
>> <extension name="9_GW" >
>>
>>   <condition field="destination_number" expression="^(9\d+)$">
>>      <action application="set" data="continue_on_fail=true"/>
>>     <action application="set" data="hangup_after_bridge=false"/>
>>      <action application="bridge"
>> data="sofia/internal/$1 at 192.168.200.201"/>
>>  </condition>
>>
>>   <condition field="brige_hangup_cause" expression="UNALLOCATED_NUMBER">
>>    <action application="bridge" data="sofia/internal/$1 at 192.168.200.202"/>
>>   </condition>
>>
>>  </extension>
>>
>> Regards,
>> Charles
>>
>> 2011/4/20, Steven Ayre <steveayre at gmail.com>:
>>> If you know the number of calls the GW can handle in advance, you can
>>> limit
>>> the call attempts before sending the call to the gateway:
>>> http://wiki.freeswitch.org/wiki/Limit
>>> There are several ways to use that.
>>>
>>> You can reroute calls to a 2nd gateway on error:
>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover
>>>
>>> There are channel variables that let you control what to consider a
>>> reroutable error and what is a give-up error:
>>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>> http://wiki.freeswitch.org/wiki/Channel_Variables#failure_causes
>>> This might also be useful, particularly with mod_limit:
>>> http://wiki.freeswitch.org/wiki/Channel_Variables#transfer_on_fail
>>>
>>> You could use mod_lcr to get a list of all the GWs, but in a random
>>> order.
>>> That'd let you load balance (randomly) but reroute when required without
>>> duplicates unlike mod_distributor.
>>>
>>> Hopefully that's enough building blocks to give you somewhere to start...
>>>
>>> -Steve
>>>
>>>
>>>
>>> On 20 April 2011 08:27, Charles <fieldpeak at gmail.com> wrote:
>>>
>>>>
>>>> i have two media GWs connected with FS for PSTN calls, FS route the
>>>> first
>>>> one in normal case, once the first GW get full (all of channels were
>>>> occupied), i need FS route to the second GW.
>>>> i found once the first GW get full, it will reply '404 not found' to FS,
>>>> can FS route to the second one according to this condition or any other
>>>> alternative?
>>>>
>>>> although i know there is mod_distributor for load balancing, however, i
>>>> need if only first one full then route to second one, so it looks this
>>>> module not suitable for this senario...
>>>>
>>>> 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
>>
>>
>
>
>
> --
> 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
>


From boris at tagnet.ru  Thu Apr 21 11:01:30 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 21 Apr 2011 13:01:30 +0600
Subject: [Freeswitch-users] LUA pattern again
Message-ID: <4DAFD64A.4090704@tagnet.ru>

Hello!

     May somebody help me how to write regex ^7?(3435)?230\d{3} with LUA 
pattern? I've tried ^7?3435?230%d%d%d and this is wrong.

-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



From denisgalvao at gmail.com  Thu Apr 21 05:13:31 2011
From: denisgalvao at gmail.com (Denis Galvao)
Date: Wed, 20 Apr 2011 22:13:31 -0300
Subject: [Freeswitch-users] SPIT attack and how to strike back
In-Reply-To: <4DAF65A0.3070600@gmx.net>
References: <4DAF65A0.3070600@gmx.net>
Message-ID: <BANLkTinhK8uJu9736W+htsn1DpgcbsFQ+Q@mail.gmail.com>

Nice! Thanks for sharing!

Denis

2011/4/20, Peter P GMX <Prometheus001 at gmx.net>:
> Hello all,
>
> I would like to share this with you as you may have also been affected
> by this threat.
>
> Yesterday we received a SPIT attack to our Freeswitch servers. We had
> about 50 register requests/sec. We noticed this as we saw a slight
> increase in the load of the Freeswitch servers. Fortunately Freeswitch
> can handle a huge amount of register requests so we had no denial of
> service.
>
> You can identify this attack by finding the following in the Register
> message:
>     User-Agent: friendly-scanner
>
> How to get rid of it:
> The attacker used Sipvicious (friendly-scanner). Sipvicious itself has a
> nice tool "svcrash.py" wich can send a malformed packet back to the
> attacker which crashes their own Sipvicious tool. You can issue this tool by
>   python svcrash.py -d <host of attacker> -p <port of attacker>
> You will need port 5060 on your machine to work. But there is also a
> workaround for that. svcrash.py will show how to overcome this if your
> port 5060 is not available.
> Download it here
> http://sipvicious.googlecode.com/files/sipvicious-0.2.6.tar.gz and
> unpack it to a folder of your choice.
>
> I wrote a small Ruby script to send the packet back to a port range, as
> our attacker used some dozens of ports to send.
> Here is the script (Install ruby first by "apt-get install ruby" e.g. on
> Debian based systems). Put it into the sipvicious directory
> kill_ports.rb:
>
> #!/usr/bin/env ruby
> host=ARGV[0]
> start_port=ARGV[1].to_i
> end_port=ARGV[2].to_i
> start_port.upto(end_port) do |port|
>   cmd="python svcrash.py -d #{host} -p #{port}"
>   p cmd
>   erg=`#{cmd}`
>   p erg
> end
>
> You now can run it by
> ./kill_ports.rb <ip> <from_port> <to_port>
>
> By using this tool we got rid of most of the SPIT messages. But after a
> while they started again to attack us from different ports.
>
> The next step is: Why not automate this by trying to identify host and
> port automatically and send back the svcrash.py packet to the sender's port?
>
> First install the pcap library
>     apt-get install libpcap-dev libpcap-ruby
>
> Then I wrote the following tool to automate this, it makes use of the
> kill_ports.rb above:
> strike_back.rb:
>
> #!/usr/bin/env ruby
> # I used some code from http://snippets.dzone.com/posts/show/5931
> require 'pcaplet'
> require 'logger'
> require 'timeout'
> @timeout=3600 # max runtime: 1 hour
>
> @logfile='strike_back.log'
> class AuditLogger < Logger
>   def format_message(severity, timestamp, progname, msg)
>     puts msg
>     "#{msg}\n"
>   end
> end
>
> logfile = File.open(@logfile, 'a')
> LOGGER = AuditLogger.new(logfile)
> LOGGER.level = Logger::INFO
> search="friendly-scanner"
> puts"Searching for '#{ search}' in SIP packets"
> $network = Pcaplet.new('-s 1500')
> $filter = Pcap::Filter.new('udp and dst port 5060', $network.capture)
> $network.add_filter($filter)
> puts "Logfile: #{@logfile}"
> puts "Starting capture..."
> begin
>   Timeout.timeout(@timeout) do # 3600 sec
>     for p in $network
>         header= "#{Time.now.strftime("%Y-%m-%d %H:%M:%S")}
> #{p.src}:#{p.sport} => #{p.dst}:#{p.dport}"
>         if $filter =~ p
>             #puts "simple search"
>             if p.udp_data.index(search)
>               LOGGER.info "Kill Friendly scanner #{p.src} with Source
> Port #{p.sport}"
>               cmd="./kill_ports.rb #{p.src} #{p.sport} #{p.sport}"
>               erg=`#{cmd}`
>               p erg
>               LOGGER.info header
>               LOGGER.info p.udp_data
>             end
>         end
>     end
>   end
> rescue Timeout::Error
>   logfile.flush
>   puts "Timeout - finished."
> end
>
> There may be a better way to code this, but at least it worked. After
> about 15min the number of attacks went to 0.
>
> Disclaimer: You can damage other systems by using these tools. So be
> carefull and use at your own risks. Do not use this tool for attacking
> other systems!
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Enviado do meu celular


From clive at lansink.co.nz  Thu Apr 21 06:38:40 2011
From: clive at lansink.co.nz (Clive Lansink)
Date: Thu, 21 Apr 2011 14:38:40 +1200
Subject: [Freeswitch-users] More on playing tones - can someone try this
	please?
Message-ID: <E1QCjp8-0001HK-Ht@mx5.orcon.net.nz>

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From msc at freeswitch.org  Thu Apr 21 11:42:33 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Apr 2011 00:42:33 -0700
Subject: [Freeswitch-users] LUA pattern again
In-Reply-To: <4DAFD64A.4090704@tagnet.ru>
References: <4DAFD64A.4090704@tagnet.ru>
Message-ID: <BANLkTimB7SXbjRp284hnDQa4Gkj0n4nYpg@mail.gmail.com>

So you want all these to match?
73435230xxx
7230xxx
3435230xxx
230xxx

Just confirming...
-MC

On Thu, Apr 21, 2011 at 12:01 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     May somebody help me how to write regex ^7?(3435)?230\d{3} with LUA
> pattern? I've tried ^7?3435?230%d%d%d and this is wrong.
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435) 230001
>   ???? +7 (3435) 230005
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 21 11:58:35 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 21 Apr 2011 15:58:35 +0800
Subject: [Freeswitch-users] FS- specify installation path
Message-ID: <BANLkTimcjfqUH+HS9_M1ZmTz9q8MKNfM5A@mail.gmail.com>

Can anyone help advise how to specify a path to install the FS,

as per the wiki,
http://wiki.freeswitch.org/wiki/Linux_Quick_Install_Guide,
(./bootstrap.sh, ./configure, make, make install)
it will be always installed at /usr/local/freeswitch.

i want to install to another path, e.g. /usr/test...

Thanks.

Regards,
Charles


From boris at tagnet.ru  Thu Apr 21 11:59:10 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 21 Apr 2011 13:59:10 +0600
Subject: [Freeswitch-users] LUA pattern again
In-Reply-To: <BANLkTimB7SXbjRp284hnDQa4Gkj0n4nYpg@mail.gmail.com>
References: <4DAFD64A.4090704@tagnet.ru>
	<BANLkTimB7SXbjRp284hnDQa4Gkj0n4nYpg@mail.gmail.com>
Message-ID: <4DAFE3CE.5000102@tagnet.ru>

Hello!

After your message I found that and my regex is invalid too :)  No, I need:

73435230xxx
3435230xxx
230xxx


> So you want all these to match?
> 73435230xxx
> 7230xxx
> 3435230xxx
> 230xxx
>
> Just confirming...
> -MC
>
> On Thu, Apr 21, 2011 at 12:01 AM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         May somebody help me how to write regex ^7?(3435)?230\d{3}
>     with LUA
>     pattern? I've tried ^7?3435?230%d%d%d and this is wrong.
>
>     --
>     ? ?????????,
>       ????? ?????????
>       ??? "??????"
>       ???. +7 (3435) 230001
>       ???? +7 (3435) 230005
>
>
>     _______________________________________________
>     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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005

-------------- next part --------------
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From peter.olsson at visionutveckling.se  Thu Apr 21 12:09:01 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 21 Apr 2011 10:09:01 +0200
Subject: [Freeswitch-users] FS- specify installation path
In-Reply-To: <BANLkTimcjfqUH+HS9_M1ZmTz9q8MKNfM5A@mail.gmail.com>
References: <BANLkTimcjfqUH+HS9_M1ZmTz9q8MKNfM5A@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58F1276186@cooper>

./configure prefix=/your/path

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r fieldpeak [fieldpeak at gmail.com]
Skickat: den 21 april 2011 09:58
Till: FreeSWITCH Users Help
?mne: [Freeswitch-users] FS- specify installation path

Can anyone help advise how to specify a path to install the FS,

as per the wiki,
http://wiki.freeswitch.org/wiki/Linux_Quick_Install_Guide,
(./bootstrap.sh, ./configure, make, make install)
it will be always installed at /usr/local/freeswitch.

i want to install to another path, e.g. /usr/test...

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

!DSPAM:4dafe44d32761309613289!



From fieldpeak at gmail.com  Thu Apr 21 12:29:08 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 21 Apr 2011 16:29:08 +0800
Subject: [Freeswitch-users] FS- specify installation path
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58F1276186@cooper>
References: <BANLkTimcjfqUH+HS9_M1ZmTz9q8MKNfM5A@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F1276186@cooper>
Message-ID: <BANLkTinNVbON5fw_M32c2wrNmR4v_SjQCA@mail.gmail.com>

Hi Peter,

Thanks for your so quick reply... i'll try...
is there any difference between ./configure prefix=/your/path and
./configure --prefix=/your/path, the latter one more -- before prefix,
i found it at google.. :)
thanks a lot.

Regards,
Charles

2011/4/21, Peter Olsson <peter.olsson at visionutveckling.se>:
> ./configure prefix=/your/path
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r fieldpeak
> [fieldpeak at gmail.com]
> Skickat: den 21 april 2011 09:58
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] FS- specify installation path
>
> Can anyone help advise how to specify a path to install the FS,
>
> as per the wiki,
> http://wiki.freeswitch.org/wiki/Linux_Quick_Install_Guide,
> (./bootstrap.sh, ./configure, make, make install)
> it will be always installed at /usr/local/freeswitch.
>
> i want to install to another path, e.g. /usr/test...
>
> 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
>
> !DSPAM:4dafe44d32761309613289!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From peter.olsson at visionutveckling.se  Thu Apr 21 12:38:31 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Thu, 21 Apr 2011 10:38:31 +0200
Subject: [Freeswitch-users] FS- specify installation path
In-Reply-To: <BANLkTinNVbON5fw_M32c2wrNmR4v_SjQCA@mail.gmail.com>
References: <BANLkTimcjfqUH+HS9_M1ZmTz9q8MKNfM5A@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F1276186@cooper>,
	<BANLkTinNVbON5fw_M32c2wrNmR4v_SjQCA@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58F1276187@cooper>

Sorry, --prefix it is! :)

./configure --prefix=/your/base/path

/Peter
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r fieldpeak [fieldpeak at gmail.com]
Skickat: den 21 april 2011 10:29
Till: FreeSWITCH Users Help
?mne: Re: [Freeswitch-users] FS- specify installation path

Hi Peter,

Thanks for your so quick reply... i'll try...
is there any difference between ./configure prefix=/your/path and
./configure --prefix=/your/path, the latter one more -- before prefix,
i found it at google.. :)
thanks a lot.

Regards,
Charles

2011/4/21, Peter Olsson <peter.olsson at visionutveckling.se>:
> ./configure prefix=/your/path
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r fieldpeak
> [fieldpeak at gmail.com]
> Skickat: den 21 april 2011 09:58
> Till: FreeSWITCH Users Help
> ?mne: [Freeswitch-users] FS- specify installation path
>
> Can anyone help advise how to specify a path to install the FS,
>
> as per the wiki,
> http://wiki.freeswitch.org/wiki/Linux_Quick_Install_Guide,
> (./bootstrap.sh, ./configure, make, make install)
> it will be always installed at /usr/local/freeswitch.
>
> i want to install to another path, e.g. /usr/test...
>
> 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
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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:4dafec0a32761862512650!



From fieldpeak at gmail.com  Thu Apr 21 12:43:02 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Thu, 21 Apr 2011 16:43:02 +0800
Subject: [Freeswitch-users] FS- specify installation path
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58F1276187@cooper>
References: <BANLkTimcjfqUH+HS9_M1ZmTz9q8MKNfM5A@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F1276186@cooper>
	<BANLkTinNVbON5fw_M32c2wrNmR4v_SjQCA@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F1276187@cooper>
Message-ID: <BANLkTi=CdaLMixJUnqwyAa=pNCQobC_GdQ@mail.gmail.com>

okay, thanks, i just tried without --, it also works, :)

2011/4/21, Peter Olsson <peter.olsson at visionutveckling.se>:
> Sorry, --prefix it is! :)
>
> ./configure --prefix=/your/base/path
>
> /Peter
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r fieldpeak
> [fieldpeak at gmail.com]
> Skickat: den 21 april 2011 10:29
> Till: FreeSWITCH Users Help
> ?mne: Re: [Freeswitch-users] FS- specify installation path
>
> Hi Peter,
>
> Thanks for your so quick reply... i'll try...
> is there any difference between ./configure prefix=/your/path and
> ./configure --prefix=/your/path, the latter one more -- before prefix,
> i found it at google.. :)
> thanks a lot.
>
> Regards,
> Charles
>
> 2011/4/21, Peter Olsson <peter.olsson at visionutveckling.se>:
>> ./configure prefix=/your/path
>>
>> /Peter
>> ________________________________________
>> Fr?n: freeswitch-users-bounces at lists.freeswitch.org
>> [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r fieldpeak
>> [fieldpeak at gmail.com]
>> Skickat: den 21 april 2011 09:58
>> Till: FreeSWITCH Users Help
>> ?mne: [Freeswitch-users] FS- specify installation path
>>
>> Can anyone help advise how to specify a path to install the FS,
>>
>> as per the wiki,
>> http://wiki.freeswitch.org/wiki/Linux_Quick_Install_Guide,
>> (./bootstrap.sh, ./configure, make, make install)
>> it will be always installed at /usr/local/freeswitch.
>>
>> i want to install to another path, e.g. /usr/test...
>>
>> 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
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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:4dafec0a32761862512650!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From boris at tagnet.ru  Thu Apr 21 14:25:40 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 21 Apr 2011 16:25:40 +0600
Subject: [Freeswitch-users] caller_id_number
Message-ID: <4DB00624.2010009@tagnet.ru>

Hello!

     When the call is incoming for my network I want CID is translated 
by my rules. So I translate caller_id_number and set 
effective_caller_id_number. But there is a little problem - when the 
call arrived to a analog phone (for example Panasonic KTX-TCD540) I see 
two number on display - original caller_id_number and 
effective_caller_id_number. Is there a way to not show caller_id_number 
in CID? Or may I modify caller_id_number instead of setting 
effective_caller_id_number?


-- 
Regards,
Boris




From kerem.erciyes at gmail.com  Thu Apr 21 15:10:19 2011
From: kerem.erciyes at gmail.com (Kerem Erciyes)
Date: Thu, 21 Apr 2011 14:10:19 +0300
Subject: [Freeswitch-users] SPIT attack and how to strike back
In-Reply-To: <BANLkTinhK8uJu9736W+htsn1DpgcbsFQ+Q@mail.gmail.com>
References: <4DAF65A0.3070600@gmx.net>
	<BANLkTinhK8uJu9736W+htsn1DpgcbsFQ+Q@mail.gmail.com>
Message-ID: <BANLkTi=24LxwBuW_-SigrMLc69tGk4SVeQ@mail.gmail.com>

That would work until attacker defines a custom user-agent string. There has
been reports of modified SipVicious code using Asterisk PBX as the agent.

On Thu, Apr 21, 2011 at 4:13 AM, Denis Galvao <denisgalvao at gmail.com> wrote:

> Nice! Thanks for sharing!
>
> Denis
>
> 2011/4/20, Peter P GMX <Prometheus001 at gmx.net>:
> > Hello all,
> >
> > I would like to share this with you as you may have also been affected
> > by this threat.
> >
> > Yesterday we received a SPIT attack to our Freeswitch servers. We had
> > about 50 register requests/sec. We noticed this as we saw a slight
> > increase in the load of the Freeswitch servers. Fortunately Freeswitch
> > can handle a huge amount of register requests so we had no denial of
> > service.
> >
> > You can identify this attack by finding the following in the Register
> > message:
> >     User-Agent: friendly-scanner
> >
> > How to get rid of it:
> > The attacker used Sipvicious (friendly-scanner). Sipvicious itself has a
> > nice tool "svcrash.py" wich can send a malformed packet back to the
> > attacker which crashes their own Sipvicious tool. You can issue this tool
> by
> >   python svcrash.py -d <host of attacker> -p <port of attacker>
> > You will need port 5060 on your machine to work. But there is also a
> > workaround for that. svcrash.py will show how to overcome this if your
> > port 5060 is not available.
> > Download it here
> > http://sipvicious.googlecode.com/files/sipvicious-0.2.6.tar.gz and
> > unpack it to a folder of your choice.
> >
> > I wrote a small Ruby script to send the packet back to a port range, as
> > our attacker used some dozens of ports to send.
> > Here is the script (Install ruby first by "apt-get install ruby" e.g. on
> > Debian based systems). Put it into the sipvicious directory
> > kill_ports.rb:
> >
> > #!/usr/bin/env ruby
> > host=ARGV[0]
> > start_port=ARGV[1].to_i
> > end_port=ARGV[2].to_i
> > start_port.upto(end_port) do |port|
> >   cmd="python svcrash.py -d #{host} -p #{port}"
> >   p cmd
> >   erg=`#{cmd}`
> >   p erg
> > end
> >
> > You now can run it by
> > ./kill_ports.rb <ip> <from_port> <to_port>
> >
> > By using this tool we got rid of most of the SPIT messages. But after a
> > while they started again to attack us from different ports.
> >
> > The next step is: Why not automate this by trying to identify host and
> > port automatically and send back the svcrash.py packet to the sender's
> port?
> >
> > First install the pcap library
> >     apt-get install libpcap-dev libpcap-ruby
> >
> > Then I wrote the following tool to automate this, it makes use of the
> > kill_ports.rb above:
> > strike_back.rb:
> >
> > #!/usr/bin/env ruby
> > # I used some code from http://snippets.dzone.com/posts/show/5931
> > require 'pcaplet'
> > require 'logger'
> > require 'timeout'
> > @timeout=3600 # max runtime: 1 hour
> >
> > @logfile='strike_back.log'
> > class AuditLogger < Logger
> >   def format_message(severity, timestamp, progname, msg)
> >     puts msg
> >     "#{msg}\n"
> >   end
> > end
> >
> > logfile = File.open(@logfile, 'a')
> > LOGGER = AuditLogger.new(logfile)
> > LOGGER.level = Logger::INFO
> > search="friendly-scanner"
> > puts"Searching for '#{ search}' in SIP packets"
> > $network = Pcaplet.new('-s 1500')
> > $filter = Pcap::Filter.new('udp and dst port 5060', $network.capture)
> > $network.add_filter($filter)
> > puts "Logfile: #{@logfile}"
> > puts "Starting capture..."
> > begin
> >   Timeout.timeout(@timeout) do # 3600 sec
> >     for p in $network
> >         header= "#{Time.now.strftime("%Y-%m-%d %H:%M:%S")}
> > #{p.src}:#{p.sport} => #{p.dst}:#{p.dport}"
> >         if $filter =~ p
> >             #puts "simple search"
> >             if p.udp_data.index(search)
> >               LOGGER.info "Kill Friendly scanner #{p.src} with Source
> > Port #{p.sport}"
> >               cmd="./kill_ports.rb #{p.src} #{p.sport} #{p.sport}"
> >               erg=`#{cmd}`
> >               p erg
> >               LOGGER.info header
> >               LOGGER.info p.udp_data
> >             end
> >         end
> >     end
> >   end
> > rescue Timeout::Error
> >   logfile.flush
> >   puts "Timeout - finished."
> > end
> >
> > There may be a better way to code this, but at least it worked. After
> > about 15min the number of attacks went to 0.
> >
> > Disclaimer: You can damage other systems by using these tools. So be
> > carefull and use at your own risks. Do not use this tool for attacking
> > other systems!
> >
> > Best regards
> > Peter
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> --
> Enviado do meu celular
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kerem Erciyes - Sistem Danismani
http://keremerciyes.com
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From Nabble at slickdeals.endjunk.com  Thu Apr 21 15:34:15 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 21 Apr 2011 04:34:15 -0700 (PDT)
Subject: [Freeswitch-users] Possible Google Voice -> FreeSWITCH directly?
In-Reply-To: <BANLkTikqrNUMkDQGt3Z7SpGVBZmSN2h_-Q@mail.gmail.com>
References: <1303314278570-6291203.post@n2.nabble.com>
	<BANLkTikgjM1RKKMqigmRRwur6K5gBrw91A@mail.gmail.com>
	<1303324308583-6291781.post@n2.nabble.com>
	<BANLkTikqrNUMkDQGt3Z7SpGVBZmSN2h_-Q@mail.gmail.com>
Message-ID: <1303385655632-6293811.post@n2.nabble.com>


Steven Ayre wrote:
> "Google Voice permits Voice Over IP (VoIP) connections through Gmail or
> Google Talk, but offers no simple way to communicate with users of other
> VoIP services, e.g. by direct connection between IP addresses or SIP
> gateway."
> http://en.wikipedia.org/wiki/Google_voice#VoIP_services
> 
> The only way for FS to use GV is via Google Talk.
Exactly. But, since adding a new e-mail address to a GMail account will show
up as a Google Chat in GV account, this will open up some new horizons in
the field of call tracing to show how GV handshakes directly with Google
Chat on the newly added e-mail address hosted locally. The idea stemmed up
from this is to write up a new FS module to integrate GV directly with FS,
instead of going through GTalk/GMail with mod_dingaling.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Possible-Google-Voice-FreeSWITCH-directly-tp6291203p6293811.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From dujinfang at gmail.com  Thu Apr 21 15:47:30 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 21 Apr 2011 19:47:30 +0800
Subject: [Freeswitch-users] questions on make mod_flite
Message-ID: <EA8825994E814CEE9AF4ED1F851EF40C@gmail.com>

Hi,

I tried to build mod_flite but faild.

git head on a mac book air with MacOSX 10.6.7

make mod_flite

 configure: error: cannot find install-sh or install.sh in "." "./.." "./../.."

touch libs/flite-1.3.99/install-sh

make mod_flite

configure: creating ./config.status
config.status: error: cannot find input file: config/config.in
make[1]: *** [/Users/seven/workspace/freeswitch/freeswitch/libs/flite-1.3.99/Makefile] Error 1
make: *** [all] Error 1

Any ideas ? thanks

-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn

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From avi at avimarcus.net  Thu Apr 21 16:02:32 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 21 Apr 2011 15:02:32 +0300
Subject: [Freeswitch-users] SPIT attack and how to strike back
In-Reply-To: <BANLkTi=24LxwBuW_-SigrMLc69tGk4SVeQ@mail.gmail.com>
References: <4DAF65A0.3070600@gmx.net>
	<BANLkTinhK8uJu9736W+htsn1DpgcbsFQ+Q@mail.gmail.com>
	<BANLkTi=24LxwBuW_-SigrMLc69tGk4SVeQ@mail.gmail.com>
Message-ID: <BANLkTinbMKECpvUopMUqa06nrUWOTZ1jFg@mail.gmail.com>

True. See http://wiki.freeswitch.org/wiki/QoS for other helpful iptable
commands for rate limiting.
Or, http://wiki.freeswitch.org/wiki/Fail2ban#SIP_DOS_Attack for actual
fail2ban-ing too many auth challanges.  (<-- I have this one set)
Or http://etel.wiki.oreilly.com/wiki/index.php/SIP_DoS/DDoS_Mitigation for
lots of cool stuff.

<http://etel.wiki.oreilly.com/wiki/index.php/SIP_DoS/DDoS_Mitigation>But all
of these are passive, to drop packets. This thread was started as a way to
actually impede the scanner...

-Avi


On Thu, Apr 21, 2011 at 2:10 PM, Kerem Erciyes <kerem.erciyes at gmail.com>wrote:

> That would work until attacker defines a custom user-agent string. There
> has been reports of modified SipVicious code using Asterisk PBX as the
> agent.
>
>
> On Thu, Apr 21, 2011 at 4:13 AM, Denis Galvao <denisgalvao at gmail.com>wrote:
>
>> Nice! Thanks for sharing!
>>
>> Denis
>>
>> 2011/4/20, Peter P GMX <Prometheus001 at gmx.net>:
>> > Hello all,
>> >
>> > I would like to share this with you as you may have also been affected
>> > by this threat.
>> >
>> > Yesterday we received a SPIT attack to our Freeswitch servers. We had
>> > about 50 register requests/sec. We noticed this as we saw a slight
>> > increase in the load of the Freeswitch servers. Fortunately Freeswitch
>> > can handle a huge amount of register requests so we had no denial of
>> > service.
>> >
>> > You can identify this attack by finding the following in the Register
>> > message:
>> >     User-Agent: friendly-scanner
>> >
>> > How to get rid of it:
>> > The attacker used Sipvicious (friendly-scanner). Sipvicious itself has a
>> > nice tool "svcrash.py" wich can send a malformed packet back to the
>> > attacker which crashes their own Sipvicious tool. You can issue this
>> tool by
>> >   python svcrash.py -d <host of attacker> -p <port of attacker>
>> > You will need port 5060 on your machine to work. But there is also a
>> > workaround for that. svcrash.py will show how to overcome this if your
>> > port 5060 is not available.
>> > Download it here
>> > http://sipvicious.googlecode.com/files/sipvicious-0.2.6.tar.gz and
>> > unpack it to a folder of your choice.
>> >
>> > I wrote a small Ruby script to send the packet back to a port range, as
>> > our attacker used some dozens of ports to send.
>> > Here is the script (Install ruby first by "apt-get install ruby" e.g. on
>> > Debian based systems). Put it into the sipvicious directory
>> > kill_ports.rb:
>> >
>> > #!/usr/bin/env ruby
>> > host=ARGV[0]
>> > start_port=ARGV[1].to_i
>> > end_port=ARGV[2].to_i
>> > start_port.upto(end_port) do |port|
>> >   cmd="python svcrash.py -d #{host} -p #{port}"
>> >   p cmd
>> >   erg=`#{cmd}`
>> >   p erg
>> > end
>> >
>> > You now can run it by
>> > ./kill_ports.rb <ip> <from_port> <to_port>
>> >
>> > By using this tool we got rid of most of the SPIT messages. But after a
>> > while they started again to attack us from different ports.
>> >
>> > The next step is: Why not automate this by trying to identify host and
>> > port automatically and send back the svcrash.py packet to the sender's
>> port?
>> >
>> > First install the pcap library
>> >     apt-get install libpcap-dev libpcap-ruby
>> >
>> > Then I wrote the following tool to automate this, it makes use of the
>> > kill_ports.rb above:
>> > strike_back.rb:
>> >
>> > #!/usr/bin/env ruby
>> > # I used some code from http://snippets.dzone.com/posts/show/5931
>> > require 'pcaplet'
>> > require 'logger'
>> > require 'timeout'
>> > @timeout=3600 # max runtime: 1 hour
>> >
>> > @logfile='strike_back.log'
>> > class AuditLogger < Logger
>> >   def format_message(severity, timestamp, progname, msg)
>> >     puts msg
>> >     "#{msg}\n"
>> >   end
>> > end
>> >
>> > logfile = File.open(@logfile, 'a')
>> > LOGGER = AuditLogger.new(logfile)
>> > LOGGER.level = Logger::INFO
>> > search="friendly-scanner"
>> > puts"Searching for '#{ search}' in SIP packets"
>> > $network = Pcaplet.new('-s 1500')
>> > $filter = Pcap::Filter.new('udp and dst port 5060', $network.capture)
>> > $network.add_filter($filter)
>> > puts "Logfile: #{@logfile}"
>> > puts "Starting capture..."
>> > begin
>> >   Timeout.timeout(@timeout) do # 3600 sec
>> >     for p in $network
>> >         header= "#{Time.now.strftime("%Y-%m-%d %H:%M:%S")}
>> > #{p.src}:#{p.sport} => #{p.dst}:#{p.dport}"
>> >         if $filter =~ p
>> >             #puts "simple search"
>> >             if p.udp_data.index(search)
>> >               LOGGER.info "Kill Friendly scanner #{p.src} with Source
>> > Port #{p.sport}"
>> >               cmd="./kill_ports.rb #{p.src} #{p.sport} #{p.sport}"
>> >               erg=`#{cmd}`
>> >               p erg
>> >               LOGGER.info header
>> >               LOGGER.info p.udp_data
>> >             end
>> >         end
>> >     end
>> >   end
>> > rescue Timeout::Error
>> >   logfile.flush
>> >   puts "Timeout - finished."
>> > end
>> >
>> > There may be a better way to code this, but at least it worked. After
>> > about 15min the number of attacks went to 0.
>> >
>> > Disclaimer: You can damage other systems by using these tools. So be
>> > carefull and use at your own risks. Do not use this tool for attacking
>> > other systems!
>> >
>> > Best regards
>> > Peter
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> --
>> Enviado do meu celular
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Kerem Erciyes - Sistem Danismani
> http://keremerciyes.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 Nabble at slickdeals.endjunk.com  Thu Apr 21 16:42:45 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 21 Apr 2011 05:42:45 -0700 (PDT)
Subject: [Freeswitch-users] SPIT attack and how to strike back
In-Reply-To: <8906FDAD-11A2-4609-9676-EFFB1179973D@freeswitch.org>
References: <4DAF65A0.3070600@gmx.net>
	<8906FDAD-11A2-4609-9676-EFFB1179973D@freeswitch.org>
Message-ID: <1303389765583-6293879.post@n2.nabble.com>


Brian West wrote:
> 
> Or 
> 
> iptables -I INPUT -j DROP -p udp --dport 5060 -m string --string
> "friendly-scanner" --algo bm
> 
> /b
Thanks. But, for FS that uses default 5080 port, wouldn't it be better to
use --dport 5080 instead of --dport 5060?

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SPIT-attack-and-how-to-strike-back-tp6292624p6293879.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From vetali100 at gmail.com  Thu Apr 21 16:48:14 2011
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Thu, 21 Apr 2011 15:48:14 +0300
Subject: [Freeswitch-users] SPIT attack and how to strike back
In-Reply-To: <1303389765583-6293879.post@n2.nabble.com>
References: <4DAF65A0.3070600@gmx.net>
	<8906FDAD-11A2-4609-9676-EFFB1179973D@freeswitch.org>
	<1303389765583-6293879.post@n2.nabble.com>
Message-ID: <BANLkTinBWuc_z0T=XxAt_SNiE8dic7FUag@mail.gmail.com>

It is not better.

It is MUST to :)


2011/4/21 mazilo <Nabble at slickdeals.endjunk.com>

>
> Brian West wrote:
> >
> > Or
> >
> > iptables -I INPUT -j DROP -p udp --dport 5060 -m string --string
> > "friendly-scanner" --algo bm
> >
> > /b
> Thanks. But, for FS that uses default 5080 port, wouldn't it be better to
> use --dport 5080 instead of --dport 5060?
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/SPIT-attack-and-how-to-strike-back-tp6292624p6293879.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 yehavi.bourvine at gmail.com  Thu Apr 21 17:01:36 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 21 Apr 2011 16:01:36 +0300
Subject: [Freeswitch-users] Polycom TLS
In-Reply-To: <1E6E3DB8-F7B3-422B-A1D9-3F89B703661F@freeswitch.org>
References: <1C72069C-FEA2-46D6-A220-9AE365489823@freeswitch.org>
	<BANLkTim-pfo_0Mj0yO939BQ_DJ+gM_A61w@mail.gmail.com>
	<1E6E3DB8-F7B3-422B-A1D9-3F89B703661F@freeswitch.org>
Message-ID: <BANLkTi=CreYrYanKjo4K9_GLtofTYYtpnw@mail.gmail.com>

Hello Brian,

  I am using sslv23.

I've tried the phone today with git from last week and git from today, and
it works mostly ok... Once I lost the registration during the registration
renewal but the debug was not on at that time... I'll continue watching it
next week.

BTW, in version 3.3.1F (both UDP and TLS) it loses BLF after a while - the
BLF does not monitor anymore the line's status. With 3.3.0 and same config
files it works ok.

                                    Regards, __Yehavi:

2011/4/20 Brian West <brian at freeswitch.org>

> What TLS Version are you using in your sofia profile?
>
> /b
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From kris at kriskinc.com  Thu Apr 21 17:55:23 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 21 Apr 2011 09:55:23 -0400
Subject: [Freeswitch-users] caller_id_number
In-Reply-To: <4DB00624.2010009@tagnet.ru>
References: <4DB00624.2010009@tagnet.ru>
Message-ID: <BANLkTim73bjLaNqdN=3+5jo-S66o+hyi9g@mail.gmail.com>

It sounds like you're forgetting to also modify caller_id_name, which
is probably the same as caller_id_number (per your carrier).

On Thu, Apr 21, 2011 at 6:25 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ? ? When the call is incoming for my network I want CID is translated
> by my rules. So I translate caller_id_number and set
> effective_caller_id_number. But there is a little problem - when the
> call arrived to a analog phone (for example Panasonic KTX-TCD540) I see
> two number on display - original caller_id_number and
> effective_caller_id_number. Is there a way to not show caller_id_number
> in CID? Or may I modify caller_id_number instead of setting
> effective_caller_id_number?
>
>
> --
> 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
>



-- 
Kristian Kielhofner


From brian at freeswitch.org  Thu Apr 21 18:02:21 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Apr 2011 09:02:21 -0500
Subject: [Freeswitch-users] Polycom TLS
In-Reply-To: <BANLkTi=CreYrYanKjo4K9_GLtofTYYtpnw@mail.gmail.com>
References: <1C72069C-FEA2-46D6-A220-9AE365489823@freeswitch.org>
	<BANLkTim-pfo_0Mj0yO939BQ_DJ+gM_A61w@mail.gmail.com>
	<1E6E3DB8-F7B3-422B-A1D9-3F89B703661F@freeswitch.org>
	<BANLkTi=CreYrYanKjo4K9_GLtofTYYtpnw@mail.gmail.com>
Message-ID: <56C05B17-4FA8-475A-9254-2E7857692E12@freeswitch.org>

I have tried tlsv1 and sslv23 and the phone just goes bonkers.  I have crash dumps and packet captures and submitted them all to Polycom for review.

/b

On Apr 21, 2011, at 8:01 AM, Yehavi Bourvine wrote:

> Hello Brian,
> 
>  I am using sslv23.
> 
> I've tried the phone today with git from last week and git from today, and
> it works mostly ok... Once I lost the registration during the registration
> renewal but the debug was not on at that time... I'll continue watching it
> next week.
> 
> BTW, in version 3.3.1F (both UDP and TLS) it loses BLF after a while - the
> BLF does not monitor anymore the line's status. With 3.3.0 and same config
> files it works ok.
> 
>                                    Regards, __Yehavi:
> 
> 2011/4/20 Brian West <brian at freeswitch.org>
> 
>> What TLS Version are you using in your sofia profile?
> 

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From brian at freeswitch.org  Thu Apr 21 18:04:46 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Apr 2011 09:04:46 -0500
Subject: [Freeswitch-users] Polycom TLS
In-Reply-To: <BANLkTi=CreYrYanKjo4K9_GLtofTYYtpnw@mail.gmail.com>
References: <1C72069C-FEA2-46D6-A220-9AE365489823@freeswitch.org>
	<BANLkTim-pfo_0Mj0yO939BQ_DJ+gM_A61w@mail.gmail.com>
	<1E6E3DB8-F7B3-422B-A1D9-3F89B703661F@freeswitch.org>
	<BANLkTi=CreYrYanKjo4K9_GLtofTYYtpnw@mail.gmail.com>
Message-ID: <31C7EC36-5FF7-4863-A88D-5B6ED2C6B5D3@freeswitch.org>

This is what I get in the logs plus half a tech dump and its a nice fatal error!  :P

Apr 20 10:24:21 192.168.1.28 192.168.1.28 0004f21f3339|0420102421|sip  |2|16121|MakeTlsConnection: SSL_connect OK : TLS Handshake completed successfully        
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|soSystemLogException vector 2 in tPktProSys                                     
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|soSystemLogException running taskShow:                                          
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|################   ################                                             
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|                                                                                
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|  NAME        ENTRY       TID    PRI   STATUS      PC       SP     ERRNO  DELAY 
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|---------- ------------ -------- --- ---------- -------- -------- ------- ----- 
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|tPktProSys 9419695c     958f1aa0 178 SUSPEND    94813c28 958f1370       0     0 
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|                                                                                
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|stack: base 0x958f1aa0  end 0x958e9aa0  size 32752  high 11084  margin 21668    
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|                                                                                
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|options: 0x4                                                                    
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|VX_DEALLOC_STACK                                                                
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|                                                                                
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|VxWorks Events                                                                  
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|--------------                                                                  
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|Events Pended on    : Not Pended                                                
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|Received Events     : 0x0                                                       
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|Options             : N/A                                                       
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|                                                                                
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|$0    =        0   t0    =        0   s0    = 7265636f   t8    =        1       
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|at    = 94cd0000   t1    =     9c00   s1    =        0   t9    = 94cd0000       
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|v0    = 72656370   t2    =     9c01   s2    =        0   k0    =       14       
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|v1    = 95d47c60   t3    =        0   s3    =        0   k1    =        0       
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|a0    = 7265636f   t4    = 95c2946c   s4    =        0   gp    = 94c3ea20       
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|a1    = 7265636f   t5    = de66fedb   s5    =        0   sp    = 958f1370       
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|a2    = 95d47620   t6    =        1   s6    =        0   s8    = 958f1370       
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|a3    =        0   t7    =        0   s7    =        0   ra    = 942541f8       
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|divlo =      1dc   divhi =        0   sr    =     9c01   pc    = 94813c28       
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|soSystemLogException running tt:                                                
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|################   ################                                             
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|94196f2c                      : 94210ce4 (952b8300, 94195564, 458df, 958d1bc0)  
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|94210dfc                      : 94259b68 (95388e10, 1d, 1, 1)                   
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|94259be0                      : 94253a18 (95d47c60, 1d, 1, 9420cf38)            
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|94253b20                      : 94254b00 (95d47c60, 948913bc, 1d, 95d47c60)     
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|94255234                      : 94254190 (952bc9b0, 7265636f, 95d47620, 30)     
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |*|16121|942541f0                      : 94813c20 (3, 958d25e8, 9481c8cc, 958d2530)      
Apr 20 10:24:22 192.168.1.28 192.168.1.28 0004f21f3339|0420102422|sys  |6|16121|soSystemLogException logging complete, rebooting...  



On Apr 21, 2011, at 8:01 AM, Yehavi Bourvine wrote:

> Hello Brian,
> 
>  I am using sslv23.
> 
> I've tried the phone today with git from last week and git from today, and
> it works mostly ok... Once I lost the registration during the registration
> renewal but the debug was not on at that time... I'll continue watching it
> next week.
> 
> BTW, in version 3.3.1F (both UDP and TLS) it loses BLF after a while - the
> BLF does not monitor anymore the line's status. With 3.3.0 and same config
> files it works ok.
> 
>                                    Regards, __Yehavi:
> 
> 2011/4/20 Brian West <brian at freeswitch.org>
> 
>> What TLS Version are you using in your sofia profile?
> 

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From brian at freeswitch.org  Thu Apr 21 18:05:40 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Apr 2011 09:05:40 -0500
Subject: [Freeswitch-users] SPIT attack and how to strike back
In-Reply-To: <1303389765583-6293879.post@n2.nabble.com>
References: <4DAF65A0.3070600@gmx.net>
	<8906FDAD-11A2-4609-9676-EFFB1179973D@freeswitch.org>
	<1303389765583-6293879.post@n2.nabble.com>
Message-ID: <F066B925-691E-4DBC-AEBF-4C7BA57B4D07@freeswitch.org>

the little prick doesn't scan on 5080 yet as far as I know!  :P

/b

On Apr 21, 2011, at 7:42 AM, mazilo wrote:

> Thanks. But, for FS that uses default 5080 port, wouldn't it be better to
> use --dport 5080 instead of --dport 5060?



From brian at freeswitch.org  Thu Apr 21 18:06:13 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Apr 2011 09:06:13 -0500
Subject: [Freeswitch-users] attended transfer to gateway
In-Reply-To: <BANLkTinAXp_u4+0OwNjY5dEDH5YHmFR-fw@mail.gmail.com>
References: <BANLkTinB6uFAWDnq-Ot1tiSPi99pu0Oruw@mail.gmail.com>
	<BANLkTikkF95DCGgpQJ-kKCR-U_=X7fF5+g@mail.gmail.com>
	<BANLkTinAXp_u4+0OwNjY5dEDH5YHmFR-fw@mail.gmail.com>
Message-ID: <65A0D45D-0666-493C-B53A-D9DC882EE77C@freeswitch.org>

There is an example of using this in the default config in features.xml

/b

On Apr 20, 2011, at 9:13 PM, elijah wrote:

> Ok, here's the log: http://pastebin.freeswitch.org/16144
> <http://pastebin.freeswitch.org/16144>Any input would be great - I'm not
> seeing the condition that is causing 0 to drop the c-leg after an attended
> bridge.
> 
> thanks,
> elijah

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From brian at freeswitch.org  Thu Apr 21 18:13:22 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Apr 2011 09:13:22 -0500
Subject: [Freeswitch-users] questions on make mod_flite
In-Reply-To: <EA8825994E814CEE9AF4ED1F851EF40C@gmail.com>
References: <EA8825994E814CEE9AF4ED1F851EF40C@gmail.com>
Message-ID: <23477397-512A-4AC8-A4F9-5DE0BBFF397D@freeswitch.org>

You're not using the apple tools to build with are you?  You have macports or fink installed eh?

/b

On Apr 21, 2011, at 6:47 AM, Seven Du wrote:

> Hi,
> 
> I tried to build mod_flite but faild.
> 
> git head on a mac book air with MacOSX 10.6.7
> 
> make mod_flite
> 
> configure: error: cannot find install-sh or install.sh in "." "./.." "./../.."
> 
> touch libs/flite-1.3.99/install-sh
> 
> make mod_flite
> 
> configure: creating ./config.status
> config.status: error: cannot find input file: config/config.in
> make[1]: *** [/Users/seven/workspace/freeswitch/freeswitch/libs/flite-1.3.99/Makefile] Error 1
> make: *** [all] Error 1
> 
> Any ideas ? thanks
> 
> -- 
> Seven Du
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj: http://www.freeswitch.org.cn
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Apr 21 18:13:56 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Apr 2011 09:13:56 -0500
Subject: [Freeswitch-users] continue_on_fail for condition only where
	100 not received?
In-Reply-To: <1303324320582-6291782.post@n2.nabble.com>
References: <1303324320582-6291782.post@n2.nabble.com>
Message-ID: <245144A7-1F53-400D-A23A-C83DEA96C400@freeswitch.org>

That seems like  rather weird request since the 100 trying isn't required.

/b

On Apr 20, 2011, at 1:32 PM, peely wrote:

> Hi,
> 
> I can't seem to get a continue_on_fail code which would only retry if no
> response whatsoever is received from an endpoint.
> 
> Does anyone know how I could route to a secondary endpoint ONLY if the
> initial endpoint does not respond with 100? I guess as a secondary part of
> the question I would need to know if I can change the INVITE timer to a
> smaller amount so the second device is attempted in less than 60 seconds.
> 
> Basically I have a bunch of routing devices within the network I want to
> route to with a failove option, these devices onward route so I trust if
> that device is OK not to route to the second device, given these devices log
> sessions themselves it's vital that if one device receives an INVITE none of
> the others do for the same session.
> 
> 
> Thanks,
> 
> 
> 
> 
> Neil.



From brian at freeswitch.org  Thu Apr 21 18:14:17 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Apr 2011 09:14:17 -0500
Subject: [Freeswitch-users] LUA directory script
In-Reply-To: <4DAF186E.5000604@tagnet.ru>
References: <4DAF15FE.1050100@tagnet.ru> <4DAF186E.5000604@tagnet.ru>
Message-ID: <5E6FFD89-D94B-4003-993F-C9D6F014EBCC@freeswitch.org>

set the variable user_context

/b

On Apr 20, 2011, at 12:31 PM, Boris Kovalenko wrote:

> Hello!
> 
>     Found I forgot <section name="directory">. Now user authenticates 
> and another problem... my call is placed into the public context and 
> should be pre_routing. And also there should be INVITE event, isn't? But 
> I can't see it...



From brian at freeswitch.org  Thu Apr 21 18:17:47 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Apr 2011 09:17:47 -0500
Subject: [Freeswitch-users] transfer_ringback triggered by "183 Session
	progress"
In-Reply-To: <4DADA951.30708@mastervoice.it>
References: <4DADA951.30708@mastervoice.it>
Message-ID: <C1AC4D08-A4CC-49A1-8331-318DD18DF7FB@freeswitch.org>

Happen to be setting ignore_early_media=true?


On Apr 19, 2011, at 10:25 AM, Irina Ivanova wrote:

> Hi,
> 
> I've noticed that if to set transfer_ringback (to any ringback tone) for 
> already answered call and then to execute the bridge to some external 
> number through the gateway, the ringing is triggered not only by "180 
> Ringing" SIP response, but also when "183 Session progress" is received.
> Does anybody know if there is a way to make transfer_ringback not to be 
> triggered by 183? I need it because in the case when the destination 
> number is busy and provider sends me 183 and then 486 (Busy here) the 
> caller hears one ringback tone and then the busy tone which makes an 
> impression that the called party rejected the call.
> 
> Thanks,
> Irina
> 
> -- 
> ================================================================
> 
> Distinti saluti
> --
> 
> Irina Ivanova
> Settore Sviluppo MasterVoice
> 
> tel: +39 0522 1590100
> fax: +39 0522 331673
> mob: +39 334 6449290
> e-mail: i.ivanova at mastervoice.it
> web: www.mastertraining.it - www.registroelettronico.com
> 
> Master Training S.r.l.
> Sede Legale: via Timolini, 18 - Correggio (RE) - Italy
> Sede Operativa: via Sani, 15 - Reggio Emilia - Italy
> Sede Commerciale: via Sani, 9 - Reggio Emilia - Italy
> 
> ================================================================
> Le informazioni contenute in questa e-mail sono da considerarsi confidenziali e esclusivamente per uso personale dei destinatari sopra indicati. Questo messaggio pu? includere dati personali o sensibili. Qualora questo messaggio fosse da Voi ricevuto per errore vogliate cortesemente darcene notizia a mezzo e-mail e distruggere il messaggio ricevuto erroneamente. Quanto precede ai fini del rispetto del Decreto Legislativo 196/2003 sulla tutela dei dati personali e sensibili.
> This e-mail and any file transmitted with it is intended only for the person or entity to which is addressed and may contain information that is privileged, confidential or otherwise protected from disclosure.Copying, dissemination or use of this e-mail or the information herein by anyone other than the intended recipient is prohibited. If you have received this e-mail by mistake, please notify us immediately by telephone or fax.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Apr 21 18:18:27 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 21 Apr 2011 09:18:27 -0500
Subject: [Freeswitch-users] Freeswitch codec usage
In-Reply-To: <BANLkTim0xU+-jaG=XUfePy11AK2ra96a2g@mail.gmail.com>
References: <BANLkTim0xU+-jaG=XUfePy11AK2ra96a2g@mail.gmail.com>
Message-ID: <6B2917BB-9F6C-4B31-A8F9-93D6F91119EA@freeswitch.org>

In that case I think you'll be stuck with it allocated the whole call.

/b

On Apr 19, 2011, at 10:46 AM, Stephen Wilde wrote:

> In order to evaluate the number of necessary G729 codec, I have a question:
> if there is a sip-to-sip call in G729 and I do a tone_detect, surely it's
> necessary to do a decode operation. When the detect operation is completed
> (and I see in the log a message "removed media bug"), for this call the
> codec resource is released? The codec resource is allocated to the call for
> entire call duration or only during tone_detect phase?
> 
> Stephen



From boris at tagnet.ru  Thu Apr 21 18:27:04 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 21 Apr 2011 20:27:04 +0600
Subject: [Freeswitch-users] LUA directory script
In-Reply-To: <5E6FFD89-D94B-4003-993F-C9D6F014EBCC@freeswitch.org>
References: <4DAF15FE.1050100@tagnet.ru> <4DAF186E.5000604@tagnet.ru>
	<5E6FFD89-D94B-4003-993F-C9D6F014EBCC@freeswitch.org>
Message-ID: <4DB03EB8.7080607@tagnet.ru>

Hello!

I set it. The root of problem - I forgot to remove gateway that was 
authenticated by cidr. And gateway was in public. Now I got LUA 
directory working....

> set the variable user_context
>
> /b
>
> On Apr 20, 2011, at 12:31 PM, Boris Kovalenko wrote:
>
>> Hello!
>>
>>      Found I forgot<section name="directory">. Now user authenticates
>> and another problem... my call is placed into the public context and
>> should be pre_routing. And also there should be INVITE event, isn't? But
>> I can't see 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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



From kris at kriskinc.com  Thu Apr 21 18:31:09 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 21 Apr 2011 10:31:09 -0400
Subject: [Freeswitch-users] continue_on_fail for condition only where
 100 not received?
In-Reply-To: <245144A7-1F53-400D-A23A-C83DEA96C400@freeswitch.org>
References: <1303324320582-6291782.post@n2.nabble.com>
	<245144A7-1F53-400D-A23A-C83DEA96C400@freeswitch.org>
Message-ID: <BANLkTi=dDmN4s0t3SaLt2ig_cDFXo9sgkQ@mail.gmail.com>

When using UDP it's typically a good idea to fire some kind of
provisional response ASAP to prevent the SIP retransmission timers
from freaking out and repeatedly resending the initial INVITE.  That's
where 100 Trying comes in.

On Thu, Apr 21, 2011 at 10:13 AM, Brian West <brian at freeswitch.org> wrote:
> That seems like ?rather weird request since the 100 trying isn't required.
>
> /b

-- 
Kristian Kielhofner


From dujinfang at gmail.com  Thu Apr 21 18:33:11 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 21 Apr 2011 22:33:11 +0800
Subject: [Freeswitch-users] questions on make mod_flite
In-Reply-To: <23477397-512A-4AC8-A4F9-5DE0BBFF397D@freeswitch.org>
References: <EA8825994E814CEE9AF4ED1F851EF40C@gmail.com>
	<23477397-512A-4AC8-A4F9-5DE0BBFF397D@freeswitch.org>
Message-ID: <4F965BE8C6614CEAB24947E1656B0867@gmail.com>

no macports or flink, but I did installed some other softwares with brew, is that related?

Thanks
-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn

On Thursday, April 21, 2011 at 10:13 PM, Brian West wrote: 
> You're not using the apple tools to build with are you? You have macports or fink installed eh?
> 
> /b
> 
> On Apr 21, 2011, at 6:47 AM, Seven Du wrote:
> 
> > Hi,
> > 
> > I tried to build mod_flite but faild.
> > 
> > git head on a mac book air with MacOSX 10.6.7
> > 
> > make mod_flite
> > 
> > configure: error: cannot find install-sh or install.sh in "." "./.." "./../.."
> > 
> > touch libs/flite-1.3.99/install-sh
> > 
> > make mod_flite
> > 
> > configure: creating ./config.status
> > config.status: error: cannot find input file: config/config.in
> > make[1]: *** [/Users/seven/workspace/freeswitch/freeswitch/libs/flite-1.3.99/Makefile] Error 1
> > make: *** [all] Error 1
> > 
> > Any ideas ? thanks
> > 
> > -- 
> > Seven Du
> > About: http://about.me/dujinfang
> > Blog: http://www.dujinfang.com
> > Proj: http://www.freeswitch.org.cn
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Apr 21 18:38:18 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 21 Apr 2011 20:38:18 +0600
Subject: [Freeswitch-users] caller_id_number
In-Reply-To: <BANLkTim73bjLaNqdN=3+5jo-S66o+hyi9g@mail.gmail.com>
References: <4DB00624.2010009@tagnet.ru>
	<BANLkTim73bjLaNqdN=3+5jo-S66o+hyi9g@mail.gmail.com>
Message-ID: <4DB0415A.3020805@tagnet.ru>

Hello!

Yes, you are right. Is the caller_id_name a mandatory variable? May I 
unset it without breaking some features?

> It sounds like you're forgetting to also modify caller_id_name, which
> is probably the same as caller_id_number (per your carrier).
>
> On Thu, Apr 21, 2011 at 6:25 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>>      When the call is incoming for my network I want CID is translated
>> by my rules. So I translate caller_id_number and set
>> effective_caller_id_number. But there is a little problem - when the
>> call arrived to a analog phone (for example Panasonic KTX-TCD540) I see
>> two number on display - original caller_id_number and
>> effective_caller_id_number. Is there a way to not show caller_id_number
>> in CID? Or may I modify caller_id_number instead of setting
>> effective_caller_id_number?
>>
>>
>> --
>> 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
>>
>
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



From kris at kriskinc.com  Thu Apr 21 18:44:35 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 21 Apr 2011 10:44:35 -0400
Subject: [Freeswitch-users] caller_id_number
In-Reply-To: <4DB0415A.3020805@tagnet.ru>
References: <4DB00624.2010009@tagnet.ru>
	<BANLkTim73bjLaNqdN=3+5jo-S66o+hyi9g@mail.gmail.com>
	<4DB0415A.3020805@tagnet.ru>
Message-ID: <BANLkTimJJEiHgo8Qhfh1MiF-tzS7jBNOGQ@mail.gmail.com>

You can manipulate caller_id_name just as you would caller_id_number.

On Thu, Apr 21, 2011 at 10:38 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> Yes, you are right. Is the caller_id_name a mandatory variable? May I
> unset it without breaking some features?
>

-- 
Kristian Kielhofner


From dujinfang at gmail.com  Thu Apr 21 18:47:17 2011
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 21 Apr 2011 22:47:17 +0800
Subject: [Freeswitch-users] questions on make mod_flite
In-Reply-To: <4F965BE8C6614CEAB24947E1656B0867@gmail.com>
References: <EA8825994E814CEE9AF4ED1F851EF40C@gmail.com>
	<23477397-512A-4AC8-A4F9-5DE0BBFF397D@freeswitch.org>
	<4F965BE8C6614CEAB24947E1656B0867@gmail.com>
Message-ID: <1A35F08E4F334D15BE6371D5AF53B761@gmail.com>

Then I downloaded flite-1.3-release

making in ...
making in include ...
making in src ...
making in src/audio ...
make[1]: *** [../build/i386-darwin10.7.0/obj/src/.make_build_dirs] Error 2
make: *** [build/i386-darwin10.7.0/obj//.make_build_dirs] Error 2

then I removed --disable-audio from mod_flite Makefile it passed, then I hit a swab conflict and I commented in string.h according to 
http://jira.freeswitch.org/browse/FS-24 and fix an extra semi-colon.

It seems that some langs are missing in 1.3-release so I commented others and only use kal. And also added a --enable-lang in mod_flite Make file so it looks :

cd $(FLITE_BUILDDIR) && $(DEFAULT_VARS) $(FLITE_DIR)/configure $(DEFAULT_ARGS) --srcdir=$(FLITE_DIR) --with-pic --disable-shared --enable-vox --enable-lang

finally get compiled and installed, however, got a flat namespace error when load in FS

load mod_flite:

2011-04-21 22:44:48.141486 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_flite.so
**dlopen(/usr/local/freeswitch/mod/mod_flite.so, 6): Symbol not found: _delete_wave
Referenced from: /usr/local/freeswitch/mod/mod_flite.so
Expected in: flat namespace
in /usr/local/freeswitch/mod/mod_flite.so**


No idea how to deal with this, any more help is appreciated. 

Thanks,
Seven


-- 
Seven Du
About: http://about.me/dujinfang
Blog: http://www.dujinfang.com
Proj: http://www.freeswitch.org.cn
Sent with Sparrow
On Thursday, April 21, 2011 at 10:33 PM, Seven Du wrote: 
> no macports or flink, but I did installed some other softwares with brew, is that related?
> 
> Thanks
> -- 
> Seven Du
> About: http://about.me/dujinfang
> Blog: http://www.dujinfang.com
> Proj: http://www.freeswitch.org.cn
> 
> On Thursday, April 21, 2011 at 10:13 PM, Brian West wrote:
> > You're not using the apple tools to build with are you? You have macports or fink installed eh?
> > 
> > /b
> > 
> > On Apr 21, 2011, at 6:47 AM, Seven Du wrote:
> > 
> > > Hi,
> > > 
> > > I tried to build mod_flite but faild.
> > > 
> > > git head on a mac book air with MacOSX 10.6.7
> > > 
> > > make mod_flite
> > > 
> > > configure: error: cannot find install-sh or install.sh in "." "./.." "./../.."
> > > 
> > > touch libs/flite-1.3.99/install-sh
> > > 
> > > make mod_flite
> > > 
> > > configure: creating ./config.status
> > > config.status: error: cannot find input file: config/config.in
> > > make[1]: *** [/Users/seven/workspace/freeswitch/freeswitch/libs/flite-1.3.99/Makefile] Error 1
> > > make: *** [all] Error 1
> > > 
> > > Any ideas ? thanks
> > > 
> > > -- 
> > > Seven Du
> > > About: http://about.me/dujinfang
> > > Blog: http://www.dujinfang.com
> > > Proj: http://www.freeswitch.org.cn
> > > 
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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  Thu Apr 21 19:03:25 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Thu, 21 Apr 2011 08:03:25 -0700 (PDT)
Subject: [Freeswitch-users] SPIT attack and how to strike back
In-Reply-To: <F066B925-691E-4DBC-AEBF-4C7BA57B4D07@freeswitch.org>
References: <4DAF65A0.3070600@gmx.net>
	<8906FDAD-11A2-4609-9676-EFFB1179973D@freeswitch.org>
	<1303389765583-6293879.post@n2.nabble.com>
	<F066B925-691E-4DBC-AEBF-4C7BA57B4D07@freeswitch.org>
Message-ID: <1303398205258-6294551.post@n2.nabble.com>


Brian West wrote:
> the little prick doesn't scan on 5080 yet as far as I know!  :P
OK and that makes sense. When I telnet to both port 5060 and 5080 on my FS,
it responded. So, I reckon it is safe and better to include both --dport
5080 and --dport 5060.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/SPIT-attack-and-how-to-strike-back-tp6292624p6294551.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kris at kriskinc.com  Thu Apr 21 19:13:24 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 21 Apr 2011 11:13:24 -0400
Subject: [Freeswitch-users] SPIT attack and how to strike back
In-Reply-To: <1303398205258-6294551.post@n2.nabble.com>
References: <4DAF65A0.3070600@gmx.net>
	<8906FDAD-11A2-4609-9676-EFFB1179973D@freeswitch.org>
	<1303389765583-6293879.post@n2.nabble.com>
	<F066B925-691E-4DBC-AEBF-4C7BA57B4D07@freeswitch.org>
	<1303398205258-6294551.post@n2.nabble.com>
Message-ID: <BANLkTi=7tKDMHzTHqXFzzRtbi=SvvZapEQ@mail.gmail.com>

On Thu, Apr 21, 2011 at 11:03 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>
> Brian West wrote:
>> the little prick doesn't scan on 5080 yet as far as I know! ?:P
> OK and that makes sense. When I telnet to both port 5060 and 5080 on my FS,
> it responded. So, I reckon it is safe and better to include both --dport
> 5080 and --dport 5060.

Keep in mind that sipvicious typically (always?) scans using UDP.
FreeSWITCH supports UDP, TCP and TLS (on 5061 if enabled).  Telnet is
TCP only so it's not a valid test for exposure to UDP only scans using
sipvicious.

For effective blocking of these attacks block TCP and UDP transports
to 5060 and 5080 if using the default config.

-- 
Kristian Kielhofner


From msc at freeswitch.org  Thu Apr 21 19:22:12 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Apr 2011 08:22:12 -0700
Subject: [Freeswitch-users] LUA pattern again
In-Reply-To: <4DAFE3CE.5000102@tagnet.ru>
References: <4DAFD64A.4090704@tagnet.ru>
	<BANLkTimB7SXbjRp284hnDQa4Gkj0n4nYpg@mail.gmail.com>
	<4DAFE3CE.5000102@tagnet.ru>
Message-ID: <BANLkTindcV0yYW=tqA-bThcj4TCsoHrm0A@mail.gmail.com>

Try this pattern:
"(7?3435)?230%d%d%d

-MC

On Thu, Apr 21, 2011 at 12:59 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

>  Hello!
>
> After your message I found that and my regex is invalid too :)  No, I need:
>
> 73435230xxx
> 3435230xxx
> 230xxx
>
>
>
> So you want all these to match?
> 73435230xxx
> 7230xxx
> 3435230xxx
> 230xxx
>
>  Just confirming...
> -MC
>
> On Thu, Apr 21, 2011 at 12:01 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
>
>> Hello!
>>
>>     May somebody help me how to write regex ^7?(3435)?230\d{3} with LUA
>> pattern? I've tried ^7?3435?230%d%d%d and this is wrong.
>>
>> --
>> ? ?????????,
>>   ????? ?????????
>>   ??? "??????"
>>   ???. +7 (3435) 230001
>>   ???? +7 (3435) 230005
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> 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
>
>
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435) 230001
>   ???? +7 (3435) 230005
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 21 19:55:02 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 21 Apr 2011 16:55:02 +0100
Subject: [Freeswitch-users] continue_on_fail for condition only where
 100 not received?
In-Reply-To: <BANLkTi=dDmN4s0t3SaLt2ig_cDFXo9sgkQ@mail.gmail.com>
References: <1303324320582-6291782.post@n2.nabble.com>
	<245144A7-1F53-400D-A23A-C83DEA96C400@freeswitch.org>
	<BANLkTi=dDmN4s0t3SaLt2ig_cDFXo9sgkQ@mail.gmail.com>
Message-ID: <BANLkTi=nuCxO2CHM0nOzpgMG8MS5i0P_tw@mail.gmail.com>

Yes, but it can be skipped - for instance if you're autoanswering you could
skip the 100 and jump straight to the 200. Or similarly if you're generating
ringback jump straight to the 183. There's no reason that 100 is needed
first.

-Steve


On 21 April 2011 15:31, Kristian Kielhofner <kris at kriskinc.com> wrote:

> When using UDP it's typically a good idea to fire some kind of
> provisional response ASAP to prevent the SIP retransmission timers
> from freaking out and repeatedly resending the initial INVITE.  That's
> where 100 Trying comes in.
>
> On Thu, Apr 21, 2011 at 10:13 AM, Brian West <brian at freeswitch.org> wrote:
> > That seems like  rather weird request since the 100 trying isn't
> required.
> >
> > /b
>
> --
> 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 eagle.antonio at gmail.com  Thu Apr 21 20:24:42 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Thu, 21 Apr 2011 16:24:42 +0000
Subject: [Freeswitch-users] Tuning Up Freeswitch
Message-ID: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>

Hello List.

I'm currently integrating an IVR in python together with freeswitch using
mod_python and ESL and my life has been well until ...
The flow of calls went over 80 simultaneous calls.
Now freeswitch starts sending packets with huge delays ( even when
establishing the call , mainly the 200 ) and firing up the IVR with tons of
delay up to 20 seconds.

So i searched the wiki forums and mailing list:

Put freeswitch on a diet , trimmed modules.conf
Played with the ulimit stuff.
Played with the IVRS to reduce load to a minimum and i was able to squeeze
more 5 calls of performance.

The problem is :

Top shows
top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92, 1.76, 1.78
Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0 zombie
Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,  0.5%si,
0.0%st
Mem:   8193336k total,  1639156k used,  6554180k free,   177208k buffers
Swap: 19534904k total,        0k used, 19534904k free,  1062272k cached

  PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
COMMAND

31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85
freeswitch

Freeswitch goes up to 150 % and puff there goes the MOS down to 0.


Some basic System Info :
Debian 6.0 ( i heard the timming module is affected by Debian , but if the
CPU % gets lower than 95% everything will be more stable)
Python 2.5

2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
8 GB of Ram

as you can see 94 % of the "Cpu Power" is sleeping :\


It appears freeswitch is only capable of using let's say "one cpu"/thread ??
Do you guys recommend simply starting more instances or redoing the IVR
stuff.


Hope you guys can help me out.

Thanks
Ant?nio Teixeira
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From msc at freeswitch.org  Thu Apr 21 20:46:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Apr 2011 09:46:30 -0700
Subject: [Freeswitch-users] continue_on_fail for condition only where
 100 not received?
In-Reply-To: <BANLkTi=nuCxO2CHM0nOzpgMG8MS5i0P_tw@mail.gmail.com>
References: <1303324320582-6291782.post@n2.nabble.com>
	<245144A7-1F53-400D-A23A-C83DEA96C400@freeswitch.org>
	<BANLkTi=dDmN4s0t3SaLt2ig_cDFXo9sgkQ@mail.gmail.com>
	<BANLkTi=nuCxO2CHM0nOzpgMG8MS5i0P_tw@mail.gmail.com>
Message-ID: <BANLkTinXAijFk8GxH4u3AL8DNS7Dd-pTVg@mail.gmail.com>

So *relying* on a 100 Trying is dangerous, but keeping an eye out for it is
okay. To the OP (Neil) I'd ask: what is the failure condition when a 100
Trying does not come back? Is that a form of "gateway down"? Is it possibly
to use SIP OPTIONS to test this instead of relying upon a 100?

-MC

On Thu, Apr 21, 2011 at 8:55 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Yes, but it can be skipped - for instance if you're autoanswering you could
> skip the 100 and jump straight to the 200. Or similarly if you're generating
> ringback jump straight to the 183. There's no reason that 100 is needed
> first.
>
> -Steve
>
>
>
> On 21 April 2011 15:31, Kristian Kielhofner <kris at kriskinc.com> wrote:
>
>> When using UDP it's typically a good idea to fire some kind of
>> provisional response ASAP to prevent the SIP retransmission timers
>> from freaking out and repeatedly resending the initial INVITE.  That's
>> where 100 Trying comes in.
>>
>> On Thu, Apr 21, 2011 at 10:13 AM, Brian West <brian at freeswitch.org>
>> wrote:
>> > That seems like  rather weird request since the 100 trying isn't
>> required.
>> >
>> > /b
>>
>> --
>> 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
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 michel at arneill-py.sacramento.ca.us  Thu Apr 21 21:07:35 2011
From: michel at arneill-py.sacramento.ca.us (Michel Py)
Date: Thu, 21 Apr 2011 10:07:35 -0700
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
	with productivity license.
In-Reply-To: <BANLkTikzB_BzKstXrsY0dYTUiY80JNWA3A@mail.gmail.com>
References: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local><BANLkTimUkXz84DHh6bac4ognmYL1WzqdiQ@mail.gmail.com>
	<BANLkTikzB_BzKstXrsY0dYTUiY80JNWA3A@mail.gmail.com>
Message-ID: <471D76419F9EF642962323D13DF1DF69011E4F@newserver.arneill-py.local>

> Michael Collins
> You can't do 3.3.1 on the 501 and 601 phones - they are oldies.
> Polycom has "legacy" firmware for those old phones and you're
> stuck w/ what they give you, which is indeed 3.1.7.?

Indeed :-(

Nevertheless, according to the wiki, I should be able to do the attendant thing.
http://wiki.freeswitch.org/wiki/Polycom_Presence_Setup
and I'm trying to make it work the "attendant" way.

What am I missing there? Where do I start the troubleshooting?

Thanks
Michel.



From elijah at crankenstein.com  Thu Apr 21 21:15:31 2011
From: elijah at crankenstein.com (elijah)
Date: Thu, 21 Apr 2011 10:15:31 -0700
Subject: [Freeswitch-users] attended transfer to gateway
In-Reply-To: <65A0D45D-0666-493C-B53A-D9DC882EE77C@freeswitch.org>
References: <BANLkTinB6uFAWDnq-Ot1tiSPi99pu0Oruw@mail.gmail.com>
	<BANLkTikkF95DCGgpQJ-kKCR-U_=X7fF5+g@mail.gmail.com>
	<BANLkTinAXp_u4+0OwNjY5dEDH5YHmFR-fw@mail.gmail.com>
	<65A0D45D-0666-493C-B53A-D9DC882EE77C@freeswitch.org>
Message-ID: <BANLkTin8Vdg=a3em_2O1bkX3C5qoGLCBBg@mail.gmail.com>

Right. That's the example I used. My config looks like this:

        <action application="bind_meta_app" data="1 b s
execute_extension::ax XML callsdirect"/>

    <extension name="bind_receiver">
      <condition field="destination_number" expression="^ax$">
        <action application="answer"/>
        <action application="read" data="10 10
'tone_stream://%(10000,0,350,440)' attxfer_callthis 30000 *"/>
        <action application="execute_extension" data="attended_xfer XML
callsdirect"/>
      </condition>
    </extension>

    <extension name="attended_xfer">
      <condition field="destination_number" expression="^attended_xfer$"/>
      <condition field="${attxfer_callthis}" expression="^(\d+)$">
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="origination_cancel_key=#"/>
        <action application="att_xfer"
data="sofia/gateway/onesource/${attxfer_callthis}"/>
        <anti-action application="eval" data="cancel transfer"/>
      </condition>
    </extension>


On Thu, Apr 21, 2011 at 7:06 AM, Brian West <brian at freeswitch.org> wrote:

> There is an example of using this in the default config in features.xml
>
> /b
>
> On Apr 20, 2011, at 9:13 PM, elijah wrote:
>
> Ok, here's the log: http://pastebin.freeswitch.org/16144
> <http://pastebin.freeswitch.org/16144>Any input would be great - I'm not
> seeing the condition that is causing 0 to drop the c-leg after an attended
> bridge.
>
> thanks,
> elijah
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 christian.loeschenkohl at xpirio.com  Thu Apr 21 21:26:28 2011
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Thu, 21 Apr 2011 19:26:28 +0200
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
Message-ID: <4DB068C4.90109@xpirio.com>

hi

we did use debian too and had such performance issues (sip packet delays, low cps).
after using centos 64bit (as advised by the devs) all performance problems are gone.

br

On 2011-04-21 18:24, Antonio Teixeira wrote:

> Hello List.
>
> I'm currently integrating an IVR in python together with freeswitch using mod_python and ESL and my life has been well until ...
> The flow of calls went over 80 simultaneous calls.
> Now freeswitch starts sending packets with huge delays ( even when establishing the call , mainly the 200 ) and firing up the IVR with tons of delay up to 20 seconds.
>
> So i searched the wiki forums and mailing list:
>
> Put freeswitch on a diet , trimmed modules.conf
> Played with the ulimit stuff.
> Played with the IVRS to reduce load to a minimum and i was able to squeeze more 5 calls of performance.
>
> The problem is :
>
> Top shows
> top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92, 1.76, 1.78
> Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0 zombie
> Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,  0.5%si,  0.0%st
> Mem:   8193336k total,  1639156k used,  6554180k free,   177208k buffers
> Swap: 19534904k total,        0k used, 19534904k free,  1062272k cached
>
>    PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
> 31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85 freeswitch
>
> Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
>
>
> Some basic System Info :
> Debian 6.0 ( i heard the timming module is affected by Debian , but if the CPU % gets lower than 95% everything will be more stable)
> Python 2.5
>
> 2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
> 8 GB of Ram
>
> as you can see 94 % of the "Cpu Power" is sleeping :\
>
>
> It appears freeswitch is only capable of using let's say "one cpu"/thread ??
> Do you guys recommend simply starting more instances or redoing the IVR stuff.
>
>
> Hope you guys can help me out.
>
> Thanks
> Ant?nio Teixeira
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

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


From yehavi.bourvine at gmail.com  Thu Apr 21 21:28:32 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 21 Apr 2011 20:28:32 +0300
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
 with productivity license.
In-Reply-To: <471D76419F9EF642962323D13DF1DF69011E4F@newserver.arneill-py.local>
References: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local>
	<BANLkTimUkXz84DHh6bac4ognmYL1WzqdiQ@mail.gmail.com>
	<BANLkTikzB_BzKstXrsY0dYTUiY80JNWA3A@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E4F@newserver.arneill-py.local>
Message-ID: <BANLkTimV6oVxz5tGQy60ODtLirL+hXP=6A@mail.gmail.com>

Hello Michel,

  The buddies worked for me a long time ago. Some day it stopped working,
but since the majority of my phones have productivity license I never
bothered to check why it happened.
I suggest a TCPDUMP to see whether the Polycom subscribes to the watched
extensions.

                        __Yehavi:

2011/4/21 Michel Py <michel at arneill-py.sacramento.ca.us>

> > Michael Collins
> > You can't do 3.3.1 on the 501 and 601 phones - they are oldies.
> > Polycom has "legacy" firmware for those old phones and you're
> > stuck w/ what they give you, which is indeed 3.1.7.
>
> Indeed :-(
>
> Nevertheless, according to the wiki, I should be able to do the attendant
> thing.
> http://wiki.freeswitch.org/wiki/Polycom_Presence_Setup
> and I'm trying to make it work the "attendant" way.
>
>  What am I missing there? Where do I start the troubleshooting?
>
> Thanks
> Michel.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Apr 21 21:41:48 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Thu, 21 Apr 2011 23:41:48 +0600
Subject: [Freeswitch-users] LUA pattern again
In-Reply-To: <BANLkTindcV0yYW=tqA-bThcj4TCsoHrm0A@mail.gmail.com>
References: <4DAFD64A.4090704@tagnet.ru>	<BANLkTimB7SXbjRp284hnDQa4Gkj0n4nYpg@mail.gmail.com>	<4DAFE3CE.5000102@tagnet.ru>
	<BANLkTindcV0yYW=tqA-bThcj4TCsoHrm0A@mail.gmail.com>
Message-ID: <4DB06C5C.7000801@tagnet.ru>

No success. I use it with string.match function, is it right?
> Try this pattern:
> "(7?3435)?230%d%d%d
>
> -MC
>
> On Thu, Apr 21, 2011 at 12:59 AM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>     After your message I found that and my regex is invalid too :) 
>     No, I need:
>
>     73435230xxx
>     3435230xxx
>     230xxx
>
>
>
>>     So you want all these to match?
>>     73435230xxx
>>     7230xxx
>>     3435230xxx
>>     230xxx
>>
>>     Just confirming...
>>     -MC
>>
>>     On Thu, Apr 21, 2011 at 12:01 AM, Boris Kovalenko
>>     <boris at tagnet.ru <mailto:boris at tagnet.ru>> wrote:
>>
>>         Hello!
>>
>>             May somebody help me how to write regex
>>         ^7?(3435)?230\d{3} with LUA
>>         pattern? I've tried ^7?3435?230%d%d%d and this is wrong.
>>
>>         --
>>         ? ?????????,
>>           ????? ?????????
>>           ??? "??????"
>>           ???. +7 (3435) 230001
>>           ???? +7 (3435) 230005
>>
>>
>>         _______________________________________________
>>         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
>
>
>     -- 
>     ? ?????????,
>        ????? ?????????
>        ??? "??????"
>        ???. +7 (3435) 230001
>        ???? +7 (3435) 230005
>
>
>     _______________________________________________
>     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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005

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From michel at arneill-py.sacramento.ca.us  Thu Apr 21 21:49:27 2011
From: michel at arneill-py.sacramento.ca.us (Michel Py)
Date: Thu, 21 Apr 2011 10:49:27 -0700
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
	with productivity license.
In-Reply-To: <BANLkTimV6oVxz5tGQy60ODtLirL+hXP=6A@mail.gmail.com>
References: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local><BANLkTimUkXz84DHh6bac4ognmYL1WzqdiQ@mail.gmail.com><BANLkTikzB_BzKstXrsY0dYTUiY80JNWA3A@mail.gmail.com><471D76419F9EF642962323D13DF1DF69011E4F@newserver.arneill-py.local>
	<BANLkTimV6oVxz5tGQy60ODtLirL+hXP=6A@mail.gmail.com>
Message-ID: <471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local>

Yehavi,


> The buddies worked for me a long time ago.

I can make the buddy part work too, but I don't like that when the other
phone is ringing, the light goes red directly. So the operator does not
know if the extension was online for a while or if it's ringing. That's
why I want to go the attendant way, not the buddy way.


> I suggest a TCPDUMP to see whether the Polycom subscribes to the
watched extensions.

I'm not there yet. I can't even get the attendant/BLF extensions to show
up on the phone screen, so I suppose it does not even try to subscribe.
I suspect I'm missing a step in the configuration.

Michel.




From yehavi.bourvine at gmail.com  Thu Apr 21 21:55:15 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 21 Apr 2011 20:55:15 +0300
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
 with productivity license.
In-Reply-To: <471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local>
References: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local>
	<BANLkTimUkXz84DHh6bac4ognmYL1WzqdiQ@mail.gmail.com>
	<BANLkTikzB_BzKstXrsY0dYTUiY80JNWA3A@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E4F@newserver.arneill-py.local>
	<BANLkTimV6oVxz5tGQy60ODtLirL+hXP=6A@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local>
Message-ID: <BANLkTikkHt3tsV=Jcc53R_kBTTn1mqqTfg@mail.gmail.com>

Hello Michel,

  If I understand you correctly, then for attendant type you need the
productivity license. Do you have it? If so, I'll send you sample of my
config.

            __Yehavi:



2011/4/21 Michel Py <michel at arneill-py.sacramento.ca.us>

> Yehavi,
>
>
> > The buddies worked for me a long time ago.
>
> I can make the buddy part work too, but I don't like that when the other
> phone is ringing, the light goes red directly. So the operator does not
> know if the extension was online for a while or if it's ringing. That's
> why I want to go the attendant way, not the buddy way.
>
>
> > I suggest a TCPDUMP to see whether the Polycom subscribes to the
> watched extensions.
>
> I'm not there yet. I can't even get the attendant/BLF extensions to show
> up on the phone screen, so I suppose it does not even try to subscribe.
> I suspect I'm missing a step in the configuration.
>
> Michel.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 michel at arneill-py.sacramento.ca.us  Thu Apr 21 22:03:15 2011
From: michel at arneill-py.sacramento.ca.us (Michel Py)
Date: Thu, 21 Apr 2011 11:03:15 -0700
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
	with productivity license.
In-Reply-To: <BANLkTikkHt3tsV=Jcc53R_kBTTn1mqqTfg@mail.gmail.com>
References: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local><BANLkTimUkXz84DHh6bac4ognmYL1WzqdiQ@mail.gmail.com><BANLkTikzB_BzKstXrsY0dYTUiY80JNWA3A@mail.gmail.com><471D76419F9EF642962323D13DF1DF69011E4F@newserver.arneill-py.local><BANLkTimV6oVxz5tGQy60ODtLirL+hXP=6A@mail.gmail.com><471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local>
	<BANLkTikkHt3tsV=Jcc53R_kBTTn1mqqTfg@mail.gmail.com>
Message-ID: <471D76419F9EF642962323D13DF1DF69011E52@newserver.arneill-py.local>

> Yehavi Bourvine wrote:
> If I understand you correctly, then for attendant type you
> need the productivity license. Do you have it?

Yes, and I see in the logs that the phone is loading it:

from tcpdump port 69 -v
17:34:57.813155 IP (tos 0x0, ttl 64, id 1322, offset 0, flags [none],
proto: UDP (17), length: 61) 192.168.17.233.ams > 192.168.17.1.tftp: 33
RRQ "0004f203590e-license.cfg" octet

from 0004f203590e-app.log
0413174046|copy |3|01|'tftp://192.168.17.1/0004f203590e-license.cfg'
from '192.168.17.1'
0413174046|copy |3|01|Download of '0004f203590e-license.cfg' succeeded
on attempt 1 (addr 1 of 1)

> If so, I'll send you sample of my config.

That would be very useful, thanks.

Michel.



From kris at kriskinc.com  Thu Apr 21 22:06:57 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 21 Apr 2011 14:06:57 -0400
Subject: [Freeswitch-users] continue_on_fail for condition only where
 100 not received?
In-Reply-To: <BANLkTi=nuCxO2CHM0nOzpgMG8MS5i0P_tw@mail.gmail.com>
References: <1303324320582-6291782.post@n2.nabble.com>
	<245144A7-1F53-400D-A23A-C83DEA96C400@freeswitch.org>
	<BANLkTi=dDmN4s0t3SaLt2ig_cDFXo9sgkQ@mail.gmail.com>
	<BANLkTi=nuCxO2CHM0nOzpgMG8MS5i0P_tw@mail.gmail.com>
Message-ID: <BANLkTin2g_5=RJusFR2bvJMYSQumnQOztg@mail.gmail.com>

Of course it can be skipped but nothing, and I mean nothing, can fire
faster than a 100.  To my knowledge Sofia fires a 100 before the
dialplan even begins hunting, as an example (very good behavior).  In
cases where the far end has a low T1 timer you should fire a 100
before you even take the time to execute dialplan, etc.  Especially in
cases where you may encounter high call volume and varying call
setup/hunting times.

SER config 101 is to fire a 100 before you do any DB logic, etc as it
buys you time beyond the first T1 timer (500ms or less) before the far
end (that you have no control over) retries.  It's best practice and
HIGHLY, HIGHLY recommended.

This is such a strange case I'm not sure what the best recommendation
might be as the original question doesn't contain enough specific
detail.

On Thu, Apr 21, 2011 at 11:55 AM, Steven Ayre <steveayre at gmail.com> wrote:
> Yes, but it can be skipped - for instance if you're autoanswering you could
> skip the 100 and jump straight to the 200. Or similarly if you're generating
> ringback jump straight to the 183. There's no reason that 100 is needed
> first.
>
> -Steve
>

-- 
Kristian Kielhofner


From msc at freeswitch.org  Thu Apr 21 22:16:56 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Apr 2011 11:16:56 -0700
Subject: [Freeswitch-users] LUA pattern again
In-Reply-To: <4DB06C5C.7000801@tagnet.ru>
References: <4DAFD64A.4090704@tagnet.ru>
	<BANLkTimB7SXbjRp284hnDQa4Gkj0n4nYpg@mail.gmail.com>
	<4DAFE3CE.5000102@tagnet.ru>
	<BANLkTindcV0yYW=tqA-bThcj4TCsoHrm0A@mail.gmail.com>
	<4DB06C5C.7000801@tagnet.ru>
Message-ID: <BANLkTi==pCURntsL4z1Wc6cY+5uOOT+vbQ@mail.gmail.com>

On Thu, Apr 21, 2011 at 10:41 AM, Boris Kovalenko <boris at tagnet.ru> wrote:

>  No success. I use it with string.match function, is it right?
>
> Okay, this is a limitation in Lua's pattern matching. Evidently you cannot
use ? on a group like you can in PCRE. So in this case you can use an API
call to regex. I wrote a mini test script to demonstrate.

Have fun. :)
-MC

--


-- pattern.lua


--


-- Simple Lua pattern testing





api = freeswitch.API();





str_pattern = "(3435)230%d%d%d";


my_str = "73435230111"





str_regex = "^(7?3435)?230\\d\\d\\d$";


res = api:execute("regex",my_str .. "|" .. str_regex);


stream:write("string: " .. my_str .. ", regex result: " .. res .. "\n");





my_str = "3435230111";


res = api:execute("regex",my_str .. "|" .. str_regex);


stream:write("string: " .. my_str .. ", regex result: " .. res .. "\n");





my_str = "230111";


res = api:execute("regex",my_str .. "|" .. str_regex);


stream:write("string: " .. my_str .. ", regex result: " .. res .. "\n");





my_str = "987230111";


res = api:execute("regex",my_str .. "|" .. str_regex);


stream:write("string: " .. my_str .. ", regex result: " .. res .. "\n");





Output:
freeswitch at internal> lua pattern.lua
string: 73435230111, regex result: true
string: 3435230111, regex result: true
string: 230111, regex result: true
string: 987230111, regex result: false
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From yehavi.bourvine at gmail.com  Thu Apr 21 22:22:11 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Thu, 21 Apr 2011 21:22:11 +0300
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
 with productivity license.
In-Reply-To: <471D76419F9EF642962323D13DF1DF69011E52@newserver.arneill-py.local>
References: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local>
	<BANLkTimUkXz84DHh6bac4ognmYL1WzqdiQ@mail.gmail.com>
	<BANLkTikzB_BzKstXrsY0dYTUiY80JNWA3A@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E4F@newserver.arneill-py.local>
	<BANLkTimV6oVxz5tGQy60ODtLirL+hXP=6A@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local>
	<BANLkTikkHt3tsV=Jcc53R_kBTTn1mqqTfg@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E52@newserver.arneill-py.local>
Message-ID: <BANLkTim-kwOMj8_Ti1nhMkGMXaQn+mb9TA@mail.gmail.com>

Hello Michel,

  In local-phone1.cfg add:
  <attendant>
    <attendant.behaviors>
      <attendant.behaviors.display>
        <attendant.behaviors.display.spontaneousCallAppearances
attendant.behaviors.display.spontaneousCallAppearances.automata="1"
attendant.behaviors.display.remoteCallerID.normal="1"  />
      </attendant.behaviors.display>
    </attendant.behaviors>
    <attendant.resourceList attendant.resourceList.1.type="automata"
(repeat it for 2, 3, etc,),
 </attendant>
in the phone's specific configuration file add:

<attendant attendant.reg="1" attendant.ringType="1"
attendant.behaviors.display.spontaneousCallAppearances.normal="1"
attendant.behaviors.display.spontaneousCallAppearances.automata="1"
attendant.behaviors.display.remoteCallerID.normal="1"
attendant.behaviors.display.remoteCallerID.automata="1"
attendant.resourceList.1.address="8000"
attendant.resourceList.1.label="Namei"  (repeat this for the others)
</attendant>

I hipe this helps...

                        __Yehavi:



2011/4/21 Michel Py <michel at arneill-py.sacramento.ca.us>

> > Yehavi Bourvine wrote:
> > If I understand you correctly, then for attendant type you
> > need the productivity license. Do you have it?
>
> Yes, and I see in the logs that the phone is loading it:
>
> from tcpdump port 69 -v
> 17:34:57.813155 IP (tos 0x0, ttl 64, id 1322, offset 0, flags [none],
> proto: UDP (17), length: 61) 192.168.17.233.ams > 192.168.17.1.tftp: 33
> RRQ "0004f203590e-license.cfg" octet
>
> from 0004f203590e-app.log
> 0413174046|copy |3|01|'tftp://192.168.17.1/0004f203590e-license.cfg'
> from '192.168.17.1'
> 0413174046|copy |3|01|Download of '0004f203590e-license.cfg' succeeded
> on attempt 1 (addr 1 of 1)
>
> > If so, I'll send you sample of my config.
>
> That would be very useful, thanks.
>
> Michel.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Apr 21 23:21:44 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Apr 2011 12:21:44 -0700
Subject: [Freeswitch-users] Hangup a call with silence
In-Reply-To: <BANLkTikG_vsgTF8vfsxzUqDawWJ-+=qCOQ@mail.gmail.com>
References: <AANLkTi=T9hKyzb-KrdaKQZCg6YWet_TMLNc-NEvh3MY7@mail.gmail.com>
	<4D8D1911.9030303@rosengart.de>
	<BANLkTikG_vsgTF8vfsxzUqDawWJ-+=qCOQ@mail.gmail.com>
Message-ID: <BANLkTi=5mSA6D0hXh+A4uWXSFgXS5TU4Uw@mail.gmail.com>

Apologies for the late follow up. I don't see any apps that will be
triggered on silence detect. What is the problem you are trying to solve?
-MC

On Wed, Apr 13, 2011 at 8:34 AM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> The "rtp-timeout-sec" works fine when there is no RTP at all.
> I want to hangup a call when there are RTP packets but with silence (energy
> based).
> Any way to do so?
>
> Stephen
>
> On Fri, Mar 25, 2011 at 11:37 PM, Frank Rosengart <frank at rosengart.de>wrote:
>
>> On 03/25/2011 11:14 PM, Stephen Wilde wrote:
>>
>> > There is a way to auto disconnect the call where there is, for example,
>> > 30s of silence?
>>
>> What about <param name="rtp-timeout-sec" value="30"/> in your sofia
>> profile?
>>
>> Frank
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Apr 22 00:13:15 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Thu, 21 Apr 2011 22:13:15 +0200
Subject: [Freeswitch-users] Hangup a call with silence
In-Reply-To: <BANLkTi=5mSA6D0hXh+A4uWXSFgXS5TU4Uw@mail.gmail.com>
References: <AANLkTi=T9hKyzb-KrdaKQZCg6YWet_TMLNc-NEvh3MY7@mail.gmail.com>
	<4D8D1911.9030303@rosengart.de>
	<BANLkTikG_vsgTF8vfsxzUqDawWJ-+=qCOQ@mail.gmail.com>
	<BANLkTi=5mSA6D0hXh+A4uWXSFgXS5TU4Uw@mail.gmail.com>
Message-ID: <BANLkTim0MUHqAmRjcVa+LCjNR1mv_JB5eA@mail.gmail.com>

The problem is related to a provider that sometimes, when legB hangup the
call, doesn't propagate the disconnection and the call is active until legA
hangup.
If I can activate a silence detection, I can force the call dropping after x
seconds of silence.

Stephen

On Thu, Apr 21, 2011 at 9:21 PM, Michael Collins <msc at freeswitch.org> wrote:

> Apologies for the late follow up. I don't see any apps that will be
> triggered on silence detect. What is the problem you are trying to solve?
> -MC
>
>
> On Wed, Apr 13, 2011 at 8:34 AM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> The "rtp-timeout-sec" works fine when there is no RTP at all.
>> I want to hangup a call when there are RTP packets but with silence
>> (energy based).
>> Any way to do so?
>>
>> Stephen
>>
>> On Fri, Mar 25, 2011 at 11:37 PM, Frank Rosengart <frank at rosengart.de>wrote:
>>
>>> On 03/25/2011 11:14 PM, Stephen Wilde wrote:
>>>
>>> > There is a way to auto disconnect the call where there is, for example,
>>> > 30s of silence?
>>>
>>> What about <param name="rtp-timeout-sec" value="30"/> in your sofia
>>> profile?
>>>
>>> Frank
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Prometheus001 at gmx.net  Fri Apr 22 00:29:29 2011
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 21 Apr 2011 22:29:29 +0200
Subject: [Freeswitch-users] SPIT attack and how to strike back
In-Reply-To: <BANLkTi=7tKDMHzTHqXFzzRtbi=SvvZapEQ@mail.gmail.com>
References: <4DAF65A0.3070600@gmx.net>	<8906FDAD-11A2-4609-9676-EFFB1179973D@freeswitch.org>	<1303389765583-6293879.post@n2.nabble.com>	<F066B925-691E-4DBC-AEBF-4C7BA57B4D07@freeswitch.org>	<1303398205258-6294551.post@n2.nabble.com>
	<BANLkTi=7tKDMHzTHqXFzzRtbi=SvvZapEQ@mail.gmail.com>
Message-ID: <4DB093A9.4090704@gmx.net>

I started the same aproach with IPtables rules on port 5060 and 5080.
Nevertheless the Sipvisious attack generates quite significant traffic.
As - like in our case -  all traffic above certain GBytes costs us a lot
of money, the decsribed trick was done in order to stop the Sipvicisous
scanner attacking us.
And this worked quite well.

Anyway, if you want to listen on port 5080 for Sipvicisous attacks feel
free to change the code of strike_back.rb:
from

$filter = Pcap::Filter.new('udp and dst port 5060', $network.capture)
to
$filter = Pcap::Filter.new('udp and dst port 5080', $network.capture)


Best regards
Peter

Kristian Kielhofner schrieb:
> On Thu, Apr 21, 2011 at 11:03 AM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
>   
>> Brian West wrote:
>>     
>>> the little prick doesn't scan on 5080 yet as far as I know!  :P
>>>       
>> OK and that makes sense. When I telnet to both port 5060 and 5080 on my FS,
>> it responded. So, I reckon it is safe and better to include both --dport
>> 5080 and --dport 5060.
>>     
>
> Keep in mind that sipvicious typically (always?) scans using UDP.
> FreeSWITCH supports UDP, TCP and TLS (on 5061 if enabled).  Telnet is
> TCP only so it's not a valid test for exposure to UDP only scans using
> sipvicious.
>
> For effective blocking of these attacks block TCP and UDP transports
> to 5060 and 5080 if using the default config.
>
>   



From lists at telefaks.de  Fri Apr 22 00:33:13 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Thu, 21 Apr 2011 22:33:13 +0200
Subject: [Freeswitch-users] SPIT attack and how to strike back
In-Reply-To: <BANLkTi=24LxwBuW_-SigrMLc69tGk4SVeQ@mail.gmail.com>
References: <4DAF65A0.3070600@gmx.net>	<BANLkTinhK8uJu9736W+htsn1DpgcbsFQ+Q@mail.gmail.com>
	<BANLkTi=24LxwBuW_-SigrMLc69tGk4SVeQ@mail.gmail.com>
Message-ID: <4DB09489.7030604@telefaks.de>

In this case, just change the below code to

search="your custom string"

As usually your external gateways like Asterisk should connect through
port 5080 to Freeswitch, you should even be able to identify Sipvicious
with an Asterisk UserAgent string on port 5060.

Best regards
Peter


Kerem Erciyes schrieb:
> That would work until attacker defines a custom user-agent string.
> There has been reports of modified SipVicious code using Asterisk PBX
> as the agent.
>
> On Thu, Apr 21, 2011 at 4:13 AM, Denis Galvao <denisgalvao at gmail.com
> <mailto:denisgalvao at gmail.com>> wrote:
>
>     Nice! Thanks for sharing!
>
>     Denis
>
>     2011/4/20, Peter P GMX <Prometheus001 at gmx.net
>     <mailto:Prometheus001 at gmx.net>>:
>     > Hello all,
>     >
>     > I would like to share this with you as you may have also been
>     affected
>     > by this threat.
>     >
>     > Yesterday we received a SPIT attack to our Freeswitch servers.
>     We had
>     > about 50 register requests/sec. We noticed this as we saw a slight
>     > increase in the load of the Freeswitch servers. Fortunately
>     Freeswitch
>     > can handle a huge amount of register requests so we had no denial of
>     > service.
>     >
>     > You can identify this attack by finding the following in the
>     Register
>     > message:
>     >     User-Agent: friendly-scanner
>     >
>     > How to get rid of it:
>     > The attacker used Sipvicious (friendly-scanner). Sipvicious
>     itself has a
>     > nice tool "svcrash.py" wich can send a malformed packet back to the
>     > attacker which crashes their own Sipvicious tool. You can issue
>     this tool by
>     >   python svcrash.py -d <host of attacker> -p <port of attacker>
>     > You will need port 5060 on your machine to work. But there is also a
>     > workaround for that. svcrash.py will show how to overcome this
>     if your
>     > port 5060 is not available.
>     > Download it here
>     > http://sipvicious.googlecode.com/files/sipvicious-0.2.6.tar.gz and
>     > unpack it to a folder of your choice.
>     >
>     > I wrote a small Ruby script to send the packet back to a port
>     range, as
>     > our attacker used some dozens of ports to send.
>     > Here is the script (Install ruby first by "apt-get install ruby"
>     e.g. on
>     > Debian based systems). Put it into the sipvicious directory
>     > kill_ports.rb:
>     >
>     > #!/usr/bin/env ruby
>     > host=ARGV[0]
>     > start_port=ARGV[1].to_i
>     > end_port=ARGV[2].to_i
>     > start_port.upto(end_port) do |port|
>     >   cmd="python svcrash.py -d #{host} -p #{port}"
>     >   p cmd
>     >   erg=`#{cmd}`
>     >   p erg
>     > end
>     >
>     > You now can run it by
>     > ./kill_ports.rb <ip> <from_port> <to_port>
>     >
>     > By using this tool we got rid of most of the SPIT messages. But
>     after a
>     > while they started again to attack us from different ports.
>     >
>     > The next step is: Why not automate this by trying to identify
>     host and
>     > port automatically and send back the svcrash.py packet to the
>     sender's port?
>     >
>     > First install the pcap library
>     >     apt-get install libpcap-dev libpcap-ruby
>     >
>     > Then I wrote the following tool to automate this, it makes use
>     of the
>     > kill_ports.rb above:
>     > strike_back.rb:
>     >
>     > #!/usr/bin/env ruby
>     > # I used some code from http://snippets.dzone.com/posts/show/5931
>     > require 'pcaplet'
>     > require 'logger'
>     > require 'timeout'
>     > @timeout=3600 # max runtime: 1 hour
>     >
>     > @logfile='strike_back.log'
>     > class AuditLogger < Logger
>     >   def format_message(severity, timestamp, progname, msg)
>     >     puts msg
>     >     "#{msg}\n"
>     >   end
>     > end
>     >
>     > logfile = File.open(@logfile, 'a')
>     > LOGGER = AuditLogger.new(logfile)
>     > LOGGER.level = Logger::INFO
>     > search="friendly-scanner"
>     > puts"Searching for '#{ search}' in SIP packets"
>     > $network = Pcaplet.new('-s 1500')
>     > $filter = Pcap::Filter.new('udp and dst port 5060',
>     $network.capture)
>     > $network.add_filter($filter)
>     > puts "Logfile: #{@logfile}"
>     > puts "Starting capture..."
>     > begin
>     >   Timeout.timeout(@timeout) do # 3600 sec
>     >     for p in $network
>     >         header= "#{Time.now.strftime("%Y-%m-%d %H:%M:%S")}
>     > #{p.src}:#{p.sport} => #{p.dst}:#{p.dport}"
>     >         if $filter =~ p
>     >             #puts "simple search"
>     >             if p.udp_data.index(search)
>     >               LOGGER.info "Kill Friendly scanner #{p.src} with
>     Source
>     > Port #{p.sport}"
>     >               cmd="./kill_ports.rb #{p.src} #{p.sport} #{p.sport}"
>     >               erg=`#{cmd}`
>     >               p erg
>     >               LOGGER.info header
>     >               LOGGER.info p.udp_data
>     >             end
>     >         end
>     >     end
>     >   end
>     > rescue Timeout::Error
>     >   logfile.flush
>     >   puts "Timeout - finished."
>     > end
>     >
>     > There may be a better way to code this, but at least it worked.
>     After
>     > about 15min the number of attacks went to 0.
>     >
>     > Disclaimer: You can damage other systems by using these tools. So be
>     > carefull and use at your own risks. Do not use this tool for
>     attacking
>     > other systems!
>     >
>     > Best regards
>     > Peter
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>
>     --
>     Enviado do meu celular
>
>     _______________________________________________
>     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
>
>
>
>
> -- 
> Kerem Erciyes - Sistem Danismani
> http://keremerciyes.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




From me at nevian.org  Fri Apr 22 00:44:44 2011
From: me at nevian.org (Serge S. Yuriev)
Date: Fri, 22 Apr 2011 00:44:44 +0400
Subject: [Freeswitch-users] error loading mod_snmp
In-Reply-To: <1123621303311544@web68.yandex.ru>
References: <1123621303311544@web68.yandex.ru>
Message-ID: <385971303418684@web139.yandex.ru>

Hi

Sorry, noone using mod_snmp?

20.04.2011, 18:59, "Serge S. Yuriev" <me at nevian.org>;:

> ?Hi
>
> ?2011-04-20 18:14:19.923216 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_snmp.so
> ?**/usr/lib/libnssutil3.so: undefined symbol: PR_GetDirectorySeparator**
>
> ?Last GIT, centos 5.5 32/64bit
>
> ?I'm googled and found few ?links to analogous problem with mod_spidermonkey and mod_xml_cURL. The only cure back to 2008 is to rebuild libcurl myself as it broken in Fedora..
> ?Is this right or I'm doing something obvious wrong?
-- 
wbr,
Serge


From john_platts at hotmail.com  Fri Apr 22 00:50:16 2011
From: john_platts at hotmail.com (John Platts)
Date: Thu, 21 Apr 2011 15:50:16 -0500
Subject: [Freeswitch-users] Changing From header in voicemail to e-mail
	notifications
Message-ID: <SNT116-W1509E05D218F485CAC25029D920@phx.gbl>


We see FreeSWITCH mod_voicemail in the From header of the voicemail to e-mail notifications. We also want to change the E-mail address in the From header to come from a specific domain. How do we go about changing the From: header in the voicemail to e-mail notifications? We do not want FreeSWITCH mod_voicemail to appear in the From header of the e-mail messages.

 		 	   		  
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From msc at freeswitch.org  Fri Apr 22 01:01:27 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Apr 2011 14:01:27 -0700
Subject: [Freeswitch-users] Hangup a call with silence
In-Reply-To: <BANLkTim0MUHqAmRjcVa+LCjNR1mv_JB5eA@mail.gmail.com>
References: <AANLkTi=T9hKyzb-KrdaKQZCg6YWet_TMLNc-NEvh3MY7@mail.gmail.com>
	<4D8D1911.9030303@rosengart.de>
	<BANLkTikG_vsgTF8vfsxzUqDawWJ-+=qCOQ@mail.gmail.com>
	<BANLkTi=5mSA6D0hXh+A4uWXSFgXS5TU4Uw@mail.gmail.com>
	<BANLkTim0MUHqAmRjcVa+LCjNR1mv_JB5eA@mail.gmail.com>
Message-ID: <BANLkTimqupw+76=XC4RqKAZRnP_YpAG9JQ@mail.gmail.com>

On Thu, Apr 21, 2011 at 1:13 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> The problem is related to a provider that sometimes, when legB hangup the
> call, doesn't propagate the disconnection and the call is active until legA
> hangup.
> If I can activate a silence detection, I can force the call dropping after
> x seconds of silence.
>
What about the case when there is legitimately silence on the line, like
when you get put on hold with no hold music or something like that? I'm just
wondering if there's a better solution. The important question is: *why*
doesn't the provider propagate a hangup? That sounds like a serious bug on
their part and a failure to provide agreed upon services.
-MC
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From jaybinks at gmail.com  Fri Apr 22 01:04:12 2011
From: jaybinks at gmail.com (Jay Binks)
Date: Fri, 22 Apr 2011 07:04:12 +1000
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <4DB068C4.90109@xpirio.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<4DB068C4.90109@xpirio.com>
Message-ID: <6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>

I have no such problems on debian .

I use debian 5 with 2.6.18 kernel which is what Is recommended

Are you using tmpfs ??

Jay



On 22/04/2011, at 3:26 AM, Christian L?schenkohl<christian.loeschenkohl at xpirio.com> wrote:

> hi
> 
> we did use debian too and had such performance issues (sip packet delays, low cps).
> after using centos 64bit (as advised by the devs) all performance problems are gone.
> 
> br
> 
> On 2011-04-21 18:24, Antonio Teixeira wrote:
> 
>> Hello List.
>> 
>> I'm currently integrating an IVR in python together with freeswitch using mod_python and ESL and my life has been well until ...
>> The flow of calls went over 80 simultaneous calls.
>> Now freeswitch starts sending packets with huge delays ( even when establishing the call , mainly the 200 ) and firing up the IVR with tons of delay up to 20 seconds.
>> 
>> So i searched the wiki forums and mailing list:
>> 
>> Put freeswitch on a diet , trimmed modules.conf
>> Played with the ulimit stuff.
>> Played with the IVRS to reduce load to a minimum and i was able to squeeze more 5 calls of performance.
>> 
>> The problem is :
>> 
>> Top shows
>> top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92, 1.76, 1.78
>> Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0 zombie
>> Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,  0.5%si,  0.0%st
>> Mem:   8193336k total,  1639156k used,  6554180k free,   177208k buffers
>> Swap: 19534904k total,        0k used, 19534904k free,  1062272k cached
>> 
>>   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
>> 31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85 freeswitch
>> 
>> Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
>> 
>> 
>> Some basic System Info :
>> Debian 6.0 ( i heard the timming module is affected by Debian , but if the CPU % gets lower than 95% everything will be more stable)
>> Python 2.5
>> 
>> 2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
>> 8 GB of Ram
>> 
>> as you can see 94 % of the "Cpu Power" is sleeping :\
>> 
>> 
>> It appears freeswitch is only capable of using let's say "one cpu"/thread ??
>> Do you guys recommend simply starting more instances or redoing the IVR stuff.
>> 
>> 
>> Hope you guys can help me out.
>> 
>> Thanks
>> Ant?nio Teixeira
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> -- 
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
> 
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
> 
> T  +43 5 77 11 - 1000
> F  +43 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From wstephen80 at gmail.com  Fri Apr 22 01:28:14 2011
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Thu, 21 Apr 2011 23:28:14 +0200
Subject: [Freeswitch-users] Hangup a call with silence
In-Reply-To: <BANLkTimqupw+76=XC4RqKAZRnP_YpAG9JQ@mail.gmail.com>
References: <AANLkTi=T9hKyzb-KrdaKQZCg6YWet_TMLNc-NEvh3MY7@mail.gmail.com>
	<4D8D1911.9030303@rosengart.de>
	<BANLkTikG_vsgTF8vfsxzUqDawWJ-+=qCOQ@mail.gmail.com>
	<BANLkTi=5mSA6D0hXh+A4uWXSFgXS5TU4Uw@mail.gmail.com>
	<BANLkTim0MUHqAmRjcVa+LCjNR1mv_JB5eA@mail.gmail.com>
	<BANLkTimqupw+76=XC4RqKAZRnP_YpAG9JQ@mail.gmail.com>
Message-ID: <BANLkTi=aJ1Oez3wVgDRwudG5gSiBB64g5A@mail.gmail.com>

I use this provider for calls to mobile phones: the hold has a sound and I
can also accept that the call in hold is dropped.
My idea is that this provider doesn't propagate the hangup (sometimes)
simply to increase the number of seconds I pay.

Stephen

On Thu, Apr 21, 2011 at 11:01 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Thu, Apr 21, 2011 at 1:13 PM, Stephen Wilde <wstephen80 at gmail.com>wrote:
>
>> The problem is related to a provider that sometimes, when legB hangup the
>> call, doesn't propagate the disconnection and the call is active until legA
>> hangup.
>> If I can activate a silence detection, I can force the call dropping after
>> x seconds of silence.
>>
> What about the case when there is legitimately silence on the line, like
> when you get put on hold with no hold music or something like that? I'm just
> wondering if there's a better solution. The important question is: *why*
> doesn't the provider propagate a hangup? That sounds like a serious bug on
> their part and a failure to provide agreed upon services.
> -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  Fri Apr 22 02:06:45 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Apr 2011 15:06:45 -0700
Subject: [Freeswitch-users] Hangup a call with silence
In-Reply-To: <BANLkTi=aJ1Oez3wVgDRwudG5gSiBB64g5A@mail.gmail.com>
References: <AANLkTi=T9hKyzb-KrdaKQZCg6YWet_TMLNc-NEvh3MY7@mail.gmail.com>
	<4D8D1911.9030303@rosengart.de>
	<BANLkTikG_vsgTF8vfsxzUqDawWJ-+=qCOQ@mail.gmail.com>
	<BANLkTi=5mSA6D0hXh+A4uWXSFgXS5TU4Uw@mail.gmail.com>
	<BANLkTim0MUHqAmRjcVa+LCjNR1mv_JB5eA@mail.gmail.com>
	<BANLkTimqupw+76=XC4RqKAZRnP_YpAG9JQ@mail.gmail.com>
	<BANLkTi=aJ1Oez3wVgDRwudG5gSiBB64g5A@mail.gmail.com>
Message-ID: <BANLkTik-q9zuG9g3YQjfqpaOT6C7DCGgbA@mail.gmail.com>

On Thu, Apr 21, 2011 at 2:28 PM, Stephen Wilde <wstephen80 at gmail.com> wrote:

> I use this provider for calls to mobile phones: the hold has a sound and I
> can also accept that the call in hold is dropped.
> My idea is that this provider doesn't propagate the hangup (sometimes)
> simply to increase the number of seconds I pay.
>

In other words, douchebaggery is afoot in the network. :)

I don't believe we have anything for this in tree right now but I think one
of our more talented programmers could create the equivalent of tone_detect
except make it a silence_detect. We have all the pieces in the source code
now (how to detect silence, how to put a media bug on a channel, how to
execute an app when the xxx_detect conditions are met) so it's a matter of
someone taking the time to put them together and test them.

Any takers? :)

-MC
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From msc at freeswitch.org  Fri Apr 22 02:55:49 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Apr 2011 15:55:49 -0700
Subject: [Freeswitch-users] Difference Between "realm" and
 "challenge-realm" in sip_profiles
In-Reply-To: <2BF7FB90DF25EA4485949F3AF2B9D696066033601A@VA3DIAXVS351.RED001.local>
References: <Acu86SXMff5CT3lATZi95py0DET6Jw==>
	<2BF7FB90DF25EA4485949F3AF2B9D696066033601A@VA3DIAXVS351.RED001.local>
Message-ID: <BANLkTin1jw+cRsVDdRcz-Y6QGkQsC3wCgg@mail.gmail.com>

On Tue, Jan 25, 2011 at 3:39 PM, Jerry Richards
<jerry.richards at teotech.com>wrote:

> Hello All,
>
>
>
> What is the difference between the parameters "realm" and "challenge-realm"
> in the conf/sip_profiles tree?
>

I don't see a "realm" parameter for sip profiles, but there is an optional
"realm" parameter for gateways. Does that answer your question?

-MC
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From benkokakao at gmail.com  Fri Apr 22 03:37:12 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Fri, 22 Apr 2011 02:37:12 +0300
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
 with productivity license.
In-Reply-To: <471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local>
References: <471D76419F9EF642962323D13DF1DF69F1D9@newserver.arneill-py.local>
	<BANLkTimUkXz84DHh6bac4ognmYL1WzqdiQ@mail.gmail.com>
	<BANLkTikzB_BzKstXrsY0dYTUiY80JNWA3A@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E4F@newserver.arneill-py.local>
	<BANLkTimV6oVxz5tGQy60ODtLirL+hXP=6A@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local>
Message-ID: <BANLkTi=XmNDV51=xoNsViHUYaUZeyMpz_A@mail.gmail.com>

> I'm not there yet. I can't even get the attendant/BLF extensions to show
> up on the phone screen, so I suppose it does not even try to subscribe.
> I suspect I'm missing a step in the configuration.

Hi Michel!

Just a wild guess - are you actually loading the file with the
attendant-settings? Usually phone1.cfg or $MAC-phone1.cfg - there
should be an entry in the basic $MAC.cfg for this file.

Best regards
Christian


From frankie.k.yiu at gmail.com  Fri Apr 22 04:45:33 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Thu, 21 Apr 2011 17:45:33 -0700
Subject: [Freeswitch-users] setup an enviroment for implementating / testing
	IVR feature
Message-ID: <BANLkTi=Uc1uBnuV=mwhKQJ2MR2G1+18-BQ@mail.gmail.com>

Hi there,

I would like to start implementating an IVR feature, but the first thing I
need to do is to able to use a softphone to dial into my freeSWITCH so that
an audio would start playing.
Right now I have the original configuration of freeSWITCH which I am not
sure how to dial into the system using the softphone.  For example, what
extension I have to call; or if I need to change anything in dialplan,
or configuration, etc.

I have C# code that let me make outgoing call to my softphone, play audio or
collect DTMF.  But I am not sure what to do for incoming call using the
original configuration from freeSWITCH.

Thanks in advance.

Frankie
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From steveayre at gmail.com  Fri Apr 22 04:51:40 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 22 Apr 2011 01:51:40 +0100
Subject: [Freeswitch-users] error loading mod_snmp
In-Reply-To: <385971303418684@web139.yandex.ru>
References: <1123621303311544@web68.yandex.ru>
	<385971303418684@web139.yandex.ru>
Message-ID: <A1E6AFD1-27CF-44D9-9EB9-C99B09D3FFC2@gmail.com>

It's pretty new... I've played with it but haven't seen your error. (Debian 5/6 x64)

Steve on iPhone

On 21 Apr 2011, at 21:44, Serge S. Yuriev <me at nevian.org> wrote:

> Hi
> 
> Sorry, noone using mod_snmp?
> 
> 20.04.2011, 18:59, "Serge S. Yuriev" <me at nevian.org>;:
> 
>>  Hi
>> 
>>  2011-04-20 18:14:19.923216 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_snmp.so
>>  **/usr/lib/libnssutil3.so: undefined symbol: PR_GetDirectorySeparator**
>> 
>>  Last GIT, centos 5.5 32/64bit
>> 
>>  I'm googled and found few  links to analogous problem with mod_spidermonkey and mod_xml_cURL. The only cure back to 2008 is to rebuild libcurl myself as it broken in Fedora..
>>  Is this right or I'm doing something obvious wrong?
> -- 
> 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


From rhuddleston at gmail.com  Fri Apr 22 05:27:48 2011
From: rhuddleston at gmail.com (rhuddleston at gmail.com)
Date: Thu, 21 Apr 2011 21:27:48 -0400
Subject: [Freeswitch-users] setup an enviroment for implementating /
	testing IVR feature
In-Reply-To: <BANLkTi=Uc1uBnuV=mwhKQJ2MR2G1+18-BQ@mail.gmail.com>
References: <BANLkTi=Uc1uBnuV=mwhKQJ2MR2G1+18-BQ@mail.gmail.com>
Message-ID: <818FF524-234F-42DB-8439-C351D65ECC75@gmail.com>

Read manual? Ive only been working w/ freeswitch for 4 weeks and used wiki to help

On Apr 21, 2011, at 8:45 PM, Frankie Yiu <frankie.k.yiu at gmail.com> wrote:

> Hi there,
>  
> I would like to start implementating an IVR feature, but the first thing I need to do is to able to use a softphone to dial into my freeSWITCH so that an audio would start playing.
> Right now I have the original configuration of freeSWITCH which I am not sure how to dial into the system using the softphone.  For example, what extension I have to call; or if I need to change anything in dialplan, or configuration, etc.
>  
> I have C# code that let me make outgoing call to my softphone, play audio or collect DTMF.  But I am not sure what to do for incoming call using the original configuration from freeSWITCH. 
>  
> 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 boris at tagnet.ru  Fri Apr 22 05:33:02 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 22 Apr 2011 07:33:02 +0600
Subject: [Freeswitch-users] LUA pattern again
In-Reply-To: <BANLkTi==pCURntsL4z1Wc6cY+5uOOT+vbQ@mail.gmail.com>
References: <4DAFD64A.4090704@tagnet.ru>	<BANLkTimB7SXbjRp284hnDQa4Gkj0n4nYpg@mail.gmail.com>	<4DAFE3CE.5000102@tagnet.ru>	<BANLkTindcV0yYW=tqA-bThcj4TCsoHrm0A@mail.gmail.com>	<4DB06C5C.7000801@tagnet.ru>
	<BANLkTi==pCURntsL4z1Wc6cY+5uOOT+vbQ@mail.gmail.com>
Message-ID: <4DB0DACE.7030008@tagnet.ru>

Thank You, Michael! Another way is use more specific LUA patterns.
>
>
> On Thu, Apr 21, 2011 at 10:41 AM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     No success. I use it with string.match function, is it right?
>
> Okay, this is a limitation in Lua's pattern matching. Evidently you 
> cannot use ? on a group like you can in PCRE. So in this case you can 
> use an API call to regex. I wrote a mini test script to demonstrate.
>
> Have fun. :)
> -MC
>
> -- 
> -- pattern.lua
> -- 
> -- Simple Lua pattern testing
> api = freeswitch.API();
> str_pattern = "(3435)230%d%d%d";
> my_str = "73435230111"
> str_regex = "^(7?3435)?230\\d\\d\\d$";
> res = api:execute("regex",my_str .. "|" .. str_regex);
> stream:write("string: " .. my_str .. ", regex result: " .. res .. "\n");
> my_str = "3435230111";
> res = api:execute("regex",my_str .. "|" .. str_regex);
> stream:write("string: " .. my_str .. ", regex result: " .. res .. "\n");
> my_str = "230111";
> res = api:execute("regex",my_str .. "|" .. str_regex);
> stream:write("string: " .. my_str .. ", regex result: " .. res .. "\n");
> my_str = "987230111";
> res = api:execute("regex",my_str .. "|" .. str_regex);
> stream:write("string: " .. my_str .. ", regex result: " .. res .. "\n");
> Output:
> freeswitch at internal> lua pattern.lua
> string: 73435230111, regex result: true
> string: 3435230111, regex result: true
> string: 230111, regex result: true
> string: 987230111, regex result: false
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005

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From philippe at ppmt.org  Fri Apr 22 05:27:10 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Thu, 21 Apr 2011 21:27:10 -0400
Subject: [Freeswitch-users] setup an enviroment for implementating /
 testing IVR feature
In-Reply-To: <BANLkTi=Uc1uBnuV=mwhKQJ2MR2G1+18-BQ@mail.gmail.com>
References: <BANLkTi=Uc1uBnuV=mwhKQJ2MR2G1+18-BQ@mail.gmail.com>
Message-ID: <4DB0D96E.2040108@ppmt.org>

Hello,

This link is a good start to get to do things with your new Freeswitch

http://wiki.freeswitch.org/wiki/Getting_Started_Guide

to test IVR just dial 5000 from your softphone to access a demo IVR

One thing to change is the default password of the extension to make it
a bit harder to crack :)
Look in the vars.xml file

/Philippe

On 11-04-21 08:45 PM, Frankie Yiu wrote:
> Hi there,
>  
> I would like to start implementating an IVR feature, but the first
> thing I need to do is to able to use a softphone to dial into my
> freeSWITCH so that an audio would start playing.
> Right now I have the original configuration of freeSWITCH which I am
> not sure how to dial into the system using the softphone.  For
> example, what extension I have to call; or if I need to change
> anything in dialplan, or configuration, etc.
>  
> I have C# code that let me make outgoing call to my softphone, play
> audio or collect DTMF.  But I am not sure what to do for incoming call
> using the original configuration from freeSWITCH. 
>  
> 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 boris at tagnet.ru  Fri Apr 22 06:01:36 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 22 Apr 2011 08:01:36 +0600
Subject: [Freeswitch-users] variable direction
Message-ID: <4DB0E180.8020201@tagnet.ru>

Hello!

     I found that with gateways I may use <variable .... 
direction="..."/>. Is this possible with regular (directory) users too? 
If not, may somebody explain why?

-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



From david.ponzone at ipeva.fr  Fri Apr 22 06:04:21 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 22 Apr 2011 04:04:21 +0200
Subject: [Freeswitch-users] Ringback while NOT YET bridging to leg B
Message-ID: <88005444-0B2E-4BFD-B6BF-5952BE09BBBE@ipeva.fr>

Hi all,

I am trying to achieve something weird, I admit.
I would need to receive a call from A, then wait X seconds before bridging to B, but I'd like to send back a ringback to A while sleeping X seconds.
The point is to allow customers to have a nice ringback, which the beginning of HAS to be heard by callers (like an announce in a call-center).

A (ugly) way to do that would be to have 2 sound files: one to be played when answering with playback, and then another one for the ringback. I would then not use sleep at all, but the bridge would only happen when the first file is over, achieving quite the same result.
The drawback of this is that I need to ask customers for 2 sound files, and that can be a pain with most of them.

I tried all combinations I could think of with ringback, instant_ringback, with no luck.

Any ideas ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




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From yehavi.bourvine at gmail.com  Fri Apr 22 08:11:34 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Fri, 22 Apr 2011 07:11:34 +0300
Subject: [Freeswitch-users] Changing From header in voicemail to e-mail
	notifications
In-Reply-To: <SNT116-W1509E05D218F485CAC25029D920@phx.gbl>
References: <SNT116-W1509E05D218F485CAC25029D920@phx.gbl>
Message-ID: <BANLkTi=i9ycE6TYEwR38Kwc08OdtasZa4g@mail.gmail.com>

Hello John,

  You have to set the following in vars,xml:

  <X-PRE-PROCESS cmd="set" data="voicemail_account=*username*"/>
  <X-PRE-PROCESS cmd="set" data="voicemail_domain=*your-domain.com*"/>

                 Regards, __Yehavi:
 2011/4/21 John Platts <john_platts at hotmail.com>

>  We see FreeSWITCH mod_voicemail in the From header of the voicemail to
> e-mail notifications. We also want to change the E-mail address in the From
> header to come from a specific domain. How do we go about changing the From:
> header in the voicemail to e-mail notifications? We do not want FreeSWITCH
> mod_voicemail to appear in the From header of the e-mail messages.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 nasir at ictinnovations.com  Fri Apr 22 07:44:32 2011
From: nasir at ictinnovations.com (Nasir Iqbal)
Date: Fri, 22 Apr 2011 08:44:32 +0500
Subject: [Freeswitch-users] ZRTP Library
Message-ID: <BANLkTimBuiOKCZQ9AQFFtKJg23C0CGhEmQ@mail.gmail.com>

Hi,

As zfone download server is offline, is there anyone who can provide me
libzrtp SDK source?


Nasir Iqbal

ICT Innovations
http://www.ictinnovations.com/
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From msc at freeswitch.org  Fri Apr 22 09:49:26 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 21 Apr 2011 22:49:26 -0700
Subject: [Freeswitch-users] variable direction
In-Reply-To: <4DB0E180.8020201@tagnet.ru>
References: <4DB0E180.8020201@tagnet.ru>
Message-ID: <BANLkTik3qeuVzmV5Ab5uYrTzZHjKLbvtxg@mail.gmail.com>

The direction variable is set at the time of the call based upon whether or
not the user is the caller (outbound) or callee (inbound).

-MC

On Thu, Apr 21, 2011 at 7:01 PM, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     I found that with gateways I may use <variable ....
> direction="..."/>. Is this possible with regular (directory) users too?
> If not, may somebody explain why?
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435) 230001
>   ???? +7 (3435) 230005
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ktaylor91 at yahoo.com  Fri Apr 22 09:56:16 2011
From: ktaylor91 at yahoo.com (Kenneth Taylor)
Date: Thu, 21 Apr 2011 22:56:16 -0700 (PDT)
Subject: [Freeswitch-users] frame per packet
In-Reply-To: <546FE599-8C29-44AD-8D78-9711BF397544@gmail.com>
References: <630958.78492.qm@web121503.mail.ne1.yahoo.com>
	<546FE599-8C29-44AD-8D78-9711BF397544@gmail.com>
Message-ID: <934071.68468.qm@web121509.mail.ne1.yahoo.com>

Hello,
I think I found a solution.
At mod_spandsp_codecs_load function on mod_spandsp_codecs, there is a place 
where FS is loading all of its codecs.
for the gsm codec (the codec i'm using) it is loading 6 implementations of the 
codec sorted by the number of frames per network packet.
When a conversation is starting FS choose the last implementation which is for 1 
frame per packet.
I disabled all the implementations except the one with 3 frames per packet, and 
now it's working.

Just to make sure I'm not missing anything.

TNX everyone
Ken




________________________________
From: Steven Ayre <steveayre at gmail.com>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Mon, April 18, 2011 11:04:40 PM
Subject: Re: [Freeswitch-users] frame per packet


AFAIK, a rtp packet *is* a frame, so no you can't.

As the other poster said, use a higher ptime. That will make the frame store a 
longer period of time which will do the same as that you want.

Overhead will be lower, but quality will be worse if you drop packets or they 
fail to arrive in a timely manner. That's very likely on gsm, and the packet 
size will mean it takes longer to be delivered which'll mean more delay and 
probably more packets arriving too late. You'll need to experiment to find a 
good balance. And remember what works well in a city might fail to work in the 
country where coverage is poorer.

Steve on iPhone

On 18 Apr 2011, at 09:49, Kenneth Taylor <ktaylor91 at yahoo.com> wrote:


Hi,
>I'm building a voip client for android phones over gsm and consider using FS as 
>sever.
>In order to save cellular costs I want to put more than one frame per Rtp 
>packet.
>
>
>Is there any easy solution to do it in FS? 
>
>
>Or, where is the right place to add it? In mod_sofia or sofia_glue? should I go 
>to lower levels?
>I don't think I should do it in the codec, it may cause problems in the rtp 
>sequence number and probably a lot more.
>
>
>TNX,
>Ken
_______________________________________________
>FreeSWITCH-users mailing list
>FreeSWITCH-users at 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 alex8207744 at gmail.com  Fri Apr 22 10:08:45 2011
From: alex8207744 at gmail.com (Alex Zhou)
Date: Fri, 22 Apr 2011 14:08:45 +0800
Subject: [Freeswitch-users] Program Crash! What should i do ?
In-Reply-To: <BANLkTimcFC5fe+ivwdoefqMa5OX2Pt-0qA@mail.gmail.com>
References: <BANLkTik3t52A_H3PJp1L7bivxhQ74bWhUQ@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58C50EE9C0@cooper>
	<BANLkTimcFC5fe+ivwdoefqMa5OX2Pt-0qA@mail.gmail.com>
Message-ID: <BANLkTi=NSDJYh=FBqBtHWz3v=S9kC1fRqA@mail.gmail.com>

> Indeed, I think he needs to consult the "waihu dev list" :D

Yes?"waihu" is my program."waihu" means "dial out" in Chinese.

It's very strange. It has dialed out other 72000 phone number and no
crash.I think if my program "waihu" has problem ,it should crash
again.

The compiler warning are :
warning: implicit declaration of function 'strcpy'
warning: incompatible implicit declaration of built-in function 'strcpy'
warning: implicit declaration of function 'strcat'
warning: incompatible implicit declaration of built-in function 'strcat'


the codes are:
char sql_null[]="";
char sql2[]="update waihu_data set status1=";
char newsql[255];
strcpy(newsql,sql_null);
strcat(newsql,sql2);


From boris at tagnet.ru  Fri Apr 22 10:08:59 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 22 Apr 2011 12:08:59 +0600
Subject: [Freeswitch-users] variable direction
In-Reply-To: <BANLkTik3qeuVzmV5Ab5uYrTzZHjKLbvtxg@mail.gmail.com>
References: <4DB0E180.8020201@tagnet.ru>
	<BANLkTik3qeuVzmV5Ab5uYrTzZHjKLbvtxg@mail.gmail.com>
Message-ID: <4DB11B7B.2090401@tagnet.ru>

Hello!

     Yes, but this form of variable is available only to gateways? I've 
tried to set direction for user variable too without success. Variable 
is always present. My test enviroment:

<include>
<user id="73435230024">
<params>
<param name="password" value="012345"/>
<param name="vm-enabled" value="false"/>
<param name="vm-password" value="326699"/>
</params>
<variables>
<variable name="toll_allow" value="internal,local,domestic,international"/>
<variable name="v_test1" value="test1" direction="inbound"/>
<variable name="v_test2" value="test2" direction="outbound"/>
<!-- <variable name="v_tagnet_context" value="ho"/>-->
</variables>
</user>
</include>

I do a call from this user to external gateway and see both variables 
are set with application=info.

> The direction variable is set at the time of the call based upon 
> whether or not the user is the caller (outbound) or callee (inbound).
>
> -MC
>
> On Thu, Apr 21, 2011 at 7:01 PM, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         I found that with gateways I may use <variable ....
>     direction="..."/>. Is this possible with regular (directory) users
>     too?
>     If not, may somebody explain why?
>
>     --
>     ? ?????????,
>       ????? ?????????
>       ??? "??????"
>       ???. +7 (3435) 230001
>       ???? +7 (3435) 230005
>
>
>     _______________________________________________
>     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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005

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From brad at tritelcomm.com  Fri Apr 22 11:56:52 2011
From: brad at tritelcomm.com (Brad Mina)
Date: Fri, 22 Apr 2011 00:56:52 -0700
Subject: [Freeswitch-users] setup an enviroment for implementating /
 testing IVR feature
In-Reply-To: <4DB0D96E.2040108@ppmt.org>
References: <BANLkTi=Uc1uBnuV=mwhKQJ2MR2G1+18-BQ@mail.gmail.com>
	<4DB0D96E.2040108@ppmt.org>
Message-ID: <BANLkTinkrkPjNT4D+=ZGcmDgcogZ99byAA@mail.gmail.com>

Don't forget to check out mod_shout and use it to pull some on-the-fly TTS
from google translate!

http://wiki.freeswitch.org/wiki/TTS

On Thu, Apr 21, 2011 at 6:27 PM, Philippe Le Toquin <philippe at ppmt.org>wrote:

>  Hello,
>
> This link is a good start to get to do things with your new Freeswitch
>
> http://wiki.freeswitch.org/wiki/Getting_Started_Guide
>
> to test IVR just dial 5000 from your softphone to access a demo IVR
>
> One thing to change is the default password of the extension to make it a
> bit harder to crack :)
> Look in the vars.xml file
>
> /Philippe
>
>
> On 11-04-21 08:45 PM, Frankie Yiu wrote:
>
> Hi there,
>
> I would like to start implementating an IVR feature, but the first thing I
> need to do is to able to use a softphone to dial into my freeSWITCH so that
> an audio would start playing.
> Right now I have the original configuration of freeSWITCH which I am not
> sure how to dial into the system using the softphone.  For example, what
> extension I have to call; or if I need to change anything in dialplan,
> or configuration, etc.
>
> I have C# code that let me make outgoing call to my softphone, play audio
> or collect DTMF.  But I am not sure what to do for incoming call using the
> original configuration from freeSWITCH.
>
> Thanks in advance.
>
> Frankie
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From david.ponzone at ipeva.fr  Fri Apr 22 12:09:44 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 22 Apr 2011 10:09:44 +0200
Subject: [Freeswitch-users] variable direction
In-Reply-To: <4DB11B7B.2090401@tagnet.ru>
References: <4DB0E180.8020201@tagnet.ru>
	<BANLkTik3qeuVzmV5Ab5uYrTzZHjKLbvtxg@mail.gmail.com>
	<4DB11B7B.2090401@tagnet.ru>
Message-ID: <2961A5A6-C60E-498B-899E-19B5BC24535F@ipeva.fr>

Boris,

you don't set direction, it is set automatically.
If you call from the user you was talking about, the A-leg (from the user to FS) will be inbound, and the B-leg (from FS to the callee party) will be outbound.
If you call the user, it's the other way around, of course.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 22/04/2011 ? 08:08, Boris Kovalenko a ?crit :

> Hello!
> 
>     Yes, but this form of variable is available only to gateways? I've tried to set direction for user variable too without success. Variable is always present. My test enviroment:
> 
> <include>
>   <user id="73435230024">
>     <params>
>       <param name="password" value="012345"/>
>       <param name="vm-enabled" value="false"/>
>       <param name="vm-password" value="326699"/>
>     </params>
>     <variables>
>       <variable name="toll_allow" value="internal,local,domestic,international"/>
>       <variable name="v_test1" value="test1" direction="inbound"/>
>       <variable name="v_test2" value="test2" direction="outbound"/>
>       <!-- <variable name="v_tagnet_context" value="ho"/>-->
>     </variables>
>   </user>
> </include>
> 
> I do a call from this user to external gateway and see both variables are set with application=info.
> 
>> The direction variable is set at the time of the call based upon whether or not the user is the caller (outbound) or callee (inbound).
>> 
>> -MC
>> 
>> On Thu, Apr 21, 2011 at 7:01 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
>> Hello!
>> 
>>     I found that with gateways I may use <variable ....
>> direction="..."/>. Is this possible with regular (directory) users too?
>> If not, may somebody explain why?
>> 
>> --
>> ? ?????????,
>>   ????? ?????????
>>   ??? "??????"
>>   ???. +7 (3435) 230001
>>   ???? +7 (3435) 230005
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> -- 
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435) 230001
>   ???? +7 (3435) 230005
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Fri Apr 22 12:14:54 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 22 Apr 2011 10:14:54 +0200
Subject: [Freeswitch-users] frame per packet
In-Reply-To: <934071.68468.qm@web121509.mail.ne1.yahoo.com>
References: <630958.78492.qm@web121503.mail.ne1.yahoo.com>
	<546FE599-8C29-44AD-8D78-9711BF397544@gmail.com>
	<934071.68468.qm@web121509.mail.ne1.yahoo.com>
Message-ID: <6496B73C-A3EA-4E0A-889A-973613B35CC1@ipeva.fr>

Kenneth,

It seemed to me that it's the endpoint which must first decide the ptime it wants to use.
Then FreeSWITCH will relay it.
What you did will probably help FreeSWITCH to advertise only GSM with 60ms ptime to the other endpoint, but I think you may have achieved that by using:
GSM at 60i
in your codec strings.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 22/04/2011 ? 07:56, Kenneth Taylor a ?crit :

> Hello,
> I think I found a solution.
> At mod_spandsp_codecs_load function on mod_spandsp_codecs, there is a place where FS is loading all of its codecs.
> for the gsm codec (the codec i'm using) it is loading 6 implementations of the codec sorted by the number of frames per network packet.
> When a conversation is starting FS choose the last implementation which is for 1 frame per packet.
> I disabled all the implementations except the one with 3 frames per packet, and now it's working.
> 
> Just to make sure I'm not missing anything.
> 
> TNX everyone
> Ken
> 
> 
> From: Steven Ayre <steveayre at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Sent: Mon, April 18, 2011 11:04:40 PM
> Subject: Re: [Freeswitch-users] frame per packet
> 
> AFAIK, a rtp packet *is* a frame, so no you can't.
> 
> As the other poster said, use a higher ptime. That will make the frame store a longer period of time which will do the same as that you want.
> 
> Overhead will be lower, but quality will be worse if you drop packets or they fail to arrive in a timely manner. That's very likely on gsm, and the packet size will mean it takes longer to be delivered which'll mean more delay and probably more packets arriving too late. You'll need to experiment to find a good balance. And remember what works well in a city might fail to work in the country where coverage is poorer.
> 
> Steve on iPhone
> 
> On 18 Apr 2011, at 09:49, Kenneth Taylor <ktaylor91 at yahoo.com> wrote:
> 
>> Hi,
>> I'm building a voip client for android phones over gsm and consider using FS as sever.
>> In order to save cellular costs I want to put more than one frame per Rtp packet.
>> 
>> Is there any easy solution to do it in FS? 
>> 
>> Or, where is the right place to add it? In mod_sofia or sofia_glue? should I go to lower levels?
>> I don't think I should do it in the codec, it may cause problems in the rtp sequence number and probably a lot more.
>> 
>> TNX,
>> Ken
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Apr 22 12:19:24 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 22 Apr 2011 14:19:24 +0600
Subject: [Freeswitch-users] variable direction
In-Reply-To: <2961A5A6-C60E-498B-899E-19B5BC24535F@ipeva.fr>
References: <4DB0E180.8020201@tagnet.ru>	<BANLkTik3qeuVzmV5Ab5uYrTzZHjKLbvtxg@mail.gmail.com>	<4DB11B7B.2090401@tagnet.ru>
	<2961A5A6-C60E-498B-899E-19B5BC24535F@ipeva.fr>
Message-ID: <4DB13A0C.3080008@tagnet.ru>

Hello!

     Yes, so, with my configuration, when I do a call only the variable 
test1 should be set? Isn't?

> Boris,
>
> you don't set direction, it is set automatically.
> If you call from the user you was talking about, the A-leg (from the 
> user to FS) will be inbound, and the B-leg (from FS to the callee 
> party) will be outbound.
> If you call the user, it's the other way around, of course.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 22/04/2011 ? 08:08, Boris Kovalenko a ?crit :
>
>> Hello!
>>
>>     Yes, but this form of variable is available only to gateways? 
>> I've tried to set direction for user variable too without success. 
>> Variable is always present. My test enviroment:
>>
>> <include>
>> <user id="73435230024">
>> <params>
>> <param name="password" value="012345"/>
>> <param name="vm-enabled" value="false"/>
>> <param name="vm-password" value="326699"/>
>> </params>
>> <variables>
>> <variable name="toll_allow" 
>> value="internal,local,domestic,international"/>
>> <variable name="v_test1" value="test1" direction="inbound"/>
>> <variable name="v_test2" value="test2" direction="outbound"/>
>> <!-- <variable name="v_tagnet_context" value="ho"/>-->
>> </variables>
>> </user>
>> </include>
>>
>> I do a call from this user to external gateway and see both variables 
>> are set with application=info.
>>
>>> The direction variable is set at the time of the call based upon 
>>> whether or not the user is the caller (outbound) or callee (inbound).
>>>
>>> -MC
>>>
>>> On Thu, Apr 21, 2011 at 7:01 PM, Boris Kovalenko <boris at tagnet.ru 
>>> <mailto:boris at tagnet.ru>> wrote:
>>>
>>>     Hello!
>>>
>>>         I found that with gateways I may use <variable ....
>>>     direction="..."/>. Is this possible with regular (directory)
>>>     users too?
>>>     If not, may somebody explain why?
>>>
>>>     --
>>>     ? ?????????,
>>>       ????? ?????????
>>>       ??? "??????"
>>>       ???. +7 (3435) 230001
>>>       ???? +7 (3435) 230005
>>>
>>>
>>>     _______________________________________________
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> -- 
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    ???. +7 (3435) 230001
>>    ???? +7 (3435) 230005
>> _______________________________________________
>> 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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005

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From david.ponzone at ipeva.fr  Fri Apr 22 12:26:29 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 22 Apr 2011 10:26:29 +0200
Subject: [Freeswitch-users] one-way audio problem on some inbound
	gateways, but not others (and no outbound gateways)
In-Reply-To: <BANLkTi=5nROvHn=MU0J7xJX3THAvnwZ+4A@mail.gmail.com>
References: <BANLkTinuTg+DwXQ5_fU1K-mE+oqYgH7apQ@mail.gmail.com>
	<1A68D020-D97F-43E6-B83B-E3C762DAD665@freeswitch.org>
	<BANLkTi=p7=xZ_GziBTYegM=J3MkEOFRVCw@mail.gmail.com>
	<201104201942.45050.sos@sokhapkin.dyndns.org>
	<BANLkTi=5nROvHn=MU0J7xJX3THAvnwZ+4A@mail.gmail.com>
Message-ID: <DACA955A-3653-4C0E-B2B5-8F6A4753AE97@ipeva.fr>

Any possibility that the faulty gateways have changed their architecture to hide media relays behind NAT, and that you have rtp-auto-adjust disabled or not correctly configured on your SIP profile ?
You may see that in the logs.
Another way to check this is to look the IP advertised by the gateway in the SDP, and to see if your FS is trying to send traffic to that IP.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 21/04/2011 ? 03:16, Fraser Redmond a ?crit :

> But it's odd that it's only happening with some gateways, but other gateways get two-way audio fine. That's what really has me puzzled.
> 
> It's definitely not a problem at the extension side, as it happens with multiple phones, and multiple locations. (So not mic, firewall, router)
> 
> Cheers,
> Fraser
> 
> 
> 
> 
> On Wed, Apr 20, 2011 at 7:42 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:
> No outgoing audio? Usually this happens if SIP ALG is enabled in your router.
> 
> On Wednesday 20 April 2011, Fraser Redmond wrote:
> > No, I tried turning off the firewall, and as I said in the OP it works with
> > one of our other gateways.
> >
> > Mic works on that one gateway, and during the calls where the audio isn't
> > transmitted the mic-indicator goes up and down.
> >
> > Cheers,
> > Fraser
> >
> > On Wed, Apr 20, 2011 at 5:46 PM, Brian West <brian at freeswitch.org> wrote:
> > > firewall issue? mic doesn't work?  can you get a pcap of all traffic?
> > >
> > > /b
> > >
> > > On Apr 20, 2011, at 3:04 PM, Fraser Redmond wrote:
> > >
> > > Thanks Brian, I'd appreciate you looking - I don't know what to look for
> > > in the sip traces (could be worth documenting some pointers in the
> > > wiki?)
> > >
> > > The sip trace is here:
> > > http://pastebin.freeswitch.org/16136
> > >
> > > I pressed enter a few times in the console before and after it connected
> > > to the extension, so about lines 490-600 is the relevant part.
> > >
> > > I also captured a pcap, in case that is of interest - let me know and
> > > I'll email it directly.
> > >
> > > Thanks,
> > > Fraser
> > >
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Fri Apr 22 12:33:22 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 22 Apr 2011 10:33:22 +0200
Subject: [Freeswitch-users] CallsIN counter on gateway
In-Reply-To: <10717914-15BF-4D58-9A77-8FAE698661D8@scarlet-internet.nl>
References: <10717914-15BF-4D58-9A77-8FAE698661D8@scarlet-internet.nl>
Message-ID: <BD86134D-C6FC-42FF-A643-A3F8C564EB3E@ipeva.fr>

Leon,

Personally, as I know I can't always expect to get the right headers in the INVITE to do this matching, and also because I think it's only valid for gateways you register to, I prefer to split the incoming INVITEs based on source IP.
It's also the opportunity to normalize what you need to normalize (caller number, called number) and to inc counters, before transfering to the desired context.
Also, I would not want my dialplan to rely on the good behaviour of the remote gateway, which could decide suddenly to send crap, or a different crap than before, because the guys there decided to upgrade their S...S or they A..E priceless boxes over the weekend.

Sorry if I misunderstood your question.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 20/04/2011 ? 12:59, Leon de Rooij a ?crit :

> Hi all,
> 
> I have a gateway that sends me invites with the callerid in the from: header and the destination number in the to: header. I'd like these inbound invites to be matched to the proper gateway (that is used for outbound calls) so that I can use params of the gateway like context, but also so the CallsIN counter gets increased for each inbound call.
> 
> Is this at all possible ? From what I understand from sofia.c, the gateway->ib_calls is only increased when an incoming request contains a gw= param or when destination_number is prefixed by gw+, or is there another way ?
> 
> thanks & kind regards,
> 
> Leon de Rooij
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Fri Apr 22 12:40:38 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 22 Apr 2011 10:40:38 +0200
Subject: [Freeswitch-users] variable direction
In-Reply-To: <4DB13A0C.3080008@tagnet.ru>
References: <4DB0E180.8020201@tagnet.ru>	<BANLkTik3qeuVzmV5Ab5uYrTzZHjKLbvtxg@mail.gmail.com>	<4DB11B7B.2090401@tagnet.ru>
	<2961A5A6-C60E-498B-899E-19B5BC24535F@ipeva.fr>
	<4DB13A0C.3080008@tagnet.ru>
Message-ID: <9D36D2B5-CE4F-45F1-9B12-3B0056E2FD7D@ipeva.fr>

Yes.
It should :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 22/04/2011 ? 10:19, Boris Kovalenko a ?crit :

> Hello!
> 
>     Yes, so, with my configuration, when I do a call only the variable test1 should be set? Isn't?
> 
>> Boris,
>> 
>> you don't set direction, it is set automatically.
>> If you call from the user you was talking about, the A-leg (from the user to FS) will be inbound, and the B-leg (from FS to the callee party) will be outbound.
>> If you call the user, it's the other way around, of course.
>> 
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> 
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> 
>> Le 22/04/2011 ? 08:08, Boris Kovalenko a ?crit :
>> 
>>> Hello!
>>> 
>>>     Yes, but this form of variable is available only to gateways? I've tried to set direction for user variable too without success. Variable is always present. My test enviroment:
>>> 
>>> <include>
>>>   <user id="73435230024">
>>>     <params>
>>>       <param name="password" value="012345"/>
>>>       <param name="vm-enabled" value="false"/>
>>>       <param name="vm-password" value="326699"/>
>>>     </params>
>>>     <variables>
>>>       <variable name="toll_allow" value="internal,local,domestic,international"/>
>>>       <variable name="v_test1" value="test1" direction="inbound"/>
>>>       <variable name="v_test2" value="test2" direction="outbound"/>
>>>       <!-- <variable name="v_tagnet_context" value="ho"/>-->
>>>     </variables>
>>>   </user>
>>> </include>
>>> 
>>> I do a call from this user to external gateway and see both variables are set with application=info.
>>> 
>>>> The direction variable is set at the time of the call based upon whether or not the user is the caller (outbound) or callee (inbound).
>>>> 
>>>> -MC
>>>> 
>>>> On Thu, Apr 21, 2011 at 7:01 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
>>>> Hello!
>>>> 
>>>>     I found that with gateways I may use <variable ....
>>>> direction="..."/>. Is this possible with regular (directory) users too?
>>>> If not, may somebody explain why?
>>>> 
>>>> --
>>>> ? ?????????,
>>>>   ????? ?????????
>>>>   ??? "??????"
>>>>   ???. +7 (3435) 230001
>>>>   ???? +7 (3435) 230005
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>> 
>>> 
>>> -- 
>>> ? ?????????,
>>>   ????? ?????????
>>>   ??? "??????"
>>>   ???. +7 (3435) 230001
>>>   ???? +7 (3435) 230005
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> -- 
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435) 230001
>   ???? +7 (3435) 230005
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 leon at scarlet-internet.nl  Fri Apr 22 13:32:52 2011
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Fri, 22 Apr 2011 11:32:52 +0200
Subject: [Freeswitch-users] CallsIN counter on gateway
In-Reply-To: <BD86134D-C6FC-42FF-A643-A3F8C564EB3E@ipeva.fr>
References: <10717914-15BF-4D58-9A77-8FAE698661D8@scarlet-internet.nl>
	<BD86134D-C6FC-42FF-A643-A3F8C564EB3E@ipeva.fr>
Message-ID: <FE773AA5-E64D-45FE-ABC8-79B699D6F962@scarlet-internet.nl>

Hello David,

Thanks for your reply.

I think I didn't word my question properly. Just as you, I also don't want to rely on the headers in the inbound SIP INVITEs that get sent to me for distinguishing carriers, but only want to do this on the basis of the originating IP address. Right now, I have this solved by having a separate Sofia profile that sends the calls to a context where I have some conditions on network_addr to make this distinction. This works, but it would be nicer if it could be done in the gateway definition in the sip profile config itself - that way the CallsIN counter would get increased.

Perhaps I should really start using proxies in front of my FS boxes, so I can change the SIP headers to contain gw+ or gw= ? But I'd really like to do this directly in FS as well - if at all possible at the moment.

Any thoughts ?

Thanks,

Leon




On Apr 22, 2011, at 10:33 AM, David Ponzone wrote:

> Leon,
> 
> Personally, as I know I can't always expect to get the right headers in the INVITE to do this matching, and also because I think it's only valid for gateways you register to, I prefer to split the incoming INVITEs based on source IP.
> It's also the opportunity to normalize what you need to normalize (caller number, called number) and to inc counters, before transfering to the desired context.
> Also, I would not want my dialplan to rely on the good behaviour of the remote gateway, which could decide suddenly to send crap, or a different crap than before, because the guys there decided to upgrade their S...S or they A..E priceless boxes over the weekend.
> 
> Sorry if I misunderstood your question.
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 
> Le 20/04/2011 ? 12:59, Leon de Rooij a ?crit :
> 
>> Hi all,
>> 
>> I have a gateway that sends me invites with the callerid in the from: header and the destination number in the to: header. I'd like these inbound invites to be matched to the proper gateway (that is used for outbound calls) so that I can use params of the gateway like context, but also so the CallsIN counter gets increased for each inbound call.
>> 
>> Is this at all possible ? From what I understand from sofia.c, the gateway->ib_calls is only increased when an incoming request contains a gw= param or when destination_number is prefixed by gw+, or is there another way ?
>> 
>> thanks & kind regards,
>> 
>> Leon de Rooij
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Apr 22 13:34:08 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 22 Apr 2011 15:34:08 +0600
Subject: [Freeswitch-users] variable direction
In-Reply-To: <9D36D2B5-CE4F-45F1-9B12-3B0056E2FD7D@ipeva.fr>
References: <4DB0E180.8020201@tagnet.ru>	<BANLkTik3qeuVzmV5Ab5uYrTzZHjKLbvtxg@mail.gmail.com>	<4DB11B7B.2090401@tagnet.ru>	<2961A5A6-C60E-498B-899E-19B5BC24535F@ipeva.fr>	<4DB13A0C.3080008@tagnet.ru>
	<9D36D2B5-CE4F-45F1-9B12-3B0056E2FD7D@ipeva.fr>
Message-ID: <4DB14B90.1080102@tagnet.ru>

But does not :) application info shows me both variables are set :(

> Yes.
> It should :)
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service ClientIPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis 
> ? l'intention exclusive de ses destinataires. Toute utilisation ou 
> diffusion non autoris?e est interdite. Tout message ?lectronique est 
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au 
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous 
> n'?tes pas destinataire de ce message, merci de le d?truire 
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 22/04/2011 ? 10:19, Boris Kovalenko a ?crit :
>
>> Hello!
>>
>>     Yes, so, with my configuration, when I do a call only the 
>> variable test1 should be set? Isn't?
>>
>>> Boris,
>>>
>>> you don't set direction, it is set automatically.
>>> If you call from the user you was talking about, the A-leg (from the 
>>> user to FS) will be inbound, and the B-leg (from FS to the callee 
>>> party) will be outbound.
>>> If you call the user, it's the other way around, of course.
>>>
>>> David Ponzone Direction Technique
>>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>> tel:      01 74 03 18 97
>>> gsm:   06 66 98 76 34
>>>
>>> Service ClientIPeva
>>> tel:      0811 46 26 26
>>> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  - www.ipeva-studio.com 
>>> <BLOCKED::http://www.ipeva-studio.com/>
>>>
>>> /Ce message et toutes les pi?ces jointes sont confidentiels et 
>>> ?tablis ? l'intention exclusive de ses destinataires. Toute 
>>> utilisation ou diffusion non autoris?e est interdite. Tout message 
>>> ?lectronique est susceptible d'alt?ration. /*/IPeva/*/ d?cline toute 
>>> responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou 
>>> falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le 
>>> d?truire imm?diatement et d'avertir l'exp?diteur./
>>> /
>>> /
>>>
>>>
>>>
>>> Le 22/04/2011 ? 08:08, Boris Kovalenko a ?crit :
>>>
>>>> Hello!
>>>>
>>>>     Yes, but this form of variable is available only to gateways? 
>>>> I've tried to set direction for user variable too without success. 
>>>> Variable is always present. My test enviroment:
>>>>
>>>> <include>
>>>> <user id="73435230024">
>>>> <params>
>>>> <param name="password" value="012345"/>
>>>> <param name="vm-enabled" value="false"/>
>>>> <param name="vm-password" value="326699"/>
>>>> </params>
>>>> <variables>
>>>> <variable name="toll_allow" 
>>>> value="internal,local,domestic,international"/>
>>>> <variable name="v_test1" value="test1" direction="inbound"/>
>>>> <variable name="v_test2" value="test2" direction="outbound"/>
>>>> <!-- <variable name="v_tagnet_context" value="ho"/>-->
>>>> </variables>
>>>> </user>
>>>> </include>
>>>>
>>>> I do a call from this user to external gateway and see both 
>>>> variables are set with application=info.
>>>>
>>>>> The direction variable is set at the time of the call based upon 
>>>>> whether or not the user is the caller (outbound) or callee (inbound).
>>>>>
>>>>> -MC
>>>>>
>>>>> On Thu, Apr 21, 2011 at 7:01 PM, Boris Kovalenko <boris at tagnet.ru 
>>>>> <mailto:boris at tagnet.ru>> wrote:
>>>>>
>>>>>     Hello!
>>>>>
>>>>>         I found that with gateways I may use <variable ....
>>>>>     direction="..."/>. Is this possible with regular (directory)
>>>>>     users too?
>>>>>     If not, may somebody explain why?
>>>>>
>>>>>     --
>>>>>     ? ?????????,
>>>>>       ????? ?????????
>>>>>       ??? "??????"
>>>>>       ???. +7 (3435) 230001
>>>>>       ???? +7 (3435) 230005
>>>>>
>>>>>
>>>>>     _______________________________________________
>>>>>     FreeSWITCH-users mailing list
>>>>>     FreeSWITCH-users at lists.freeswitch.org
>>>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> -- 
>>>> ? ?????????,
>>>>    ????? ?????????
>>>>    ??? "??????"
>>>>    ???. +7 (3435) 230001
>>>>    ???? +7 (3435) 230005
>>>> _______________________________________________
>>>> 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
>>
>>
>> -- 
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    ???. +7 (3435) 230001
>>    ???? +7 (3435) 230005
>> _______________________________________________
>> 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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005

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From christian.loeschenkohl at xpirio.com  Fri Apr 22 14:14:36 2011
From: christian.loeschenkohl at xpirio.com (=?UTF-8?B?Q2hyaXN0aWFuIEzDtnNjaGVua29obA==?=)
Date: Fri, 22 Apr 2011 12:14:36 +0200
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>	<4DB068C4.90109@xpirio.com>
	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
Message-ID: <4DB1550C.1050607@xpirio.com>

hi

if you refer to my e-mail

yes, we do use tmpfs on both variants but
- delays occur with concurrent calls > 80-100
- cps is limited to 5-10 on debian, with centos 30 cps and more are no problem at all

also cpu load, stability and overall performace have been much better since using centos

i just found out for me that debian works not as good for me as centos does.
btw. everywhere else debian is 1st choice (desktop, lamp, db etc.)

br


On 2011-04-21 23:04, Jay Binks wrote:

> I have no such problems on debian .
>
> I use debian 5 with 2.6.18 kernel which is what Is recommended
>
> Are you using tmpfs ??
>
> Jay
>
>
>
> On 22/04/2011, at 3:26 AM, Christian L?schenkohl<christian.loeschenkohl at xpirio.com>  wrote:
>
>> hi
>>
>> we did use debian too and had such performance issues (sip packet delays, low cps).
>> after using centos 64bit (as advised by the devs) all performance problems are gone.
>>
>> br
>>
>> On 2011-04-21 18:24, Antonio Teixeira wrote:
>>
>>> Hello List.
>>>
>>> I'm currently integrating an IVR in python together with freeswitch using mod_python and ESL and my life has been well until ...
>>> The flow of calls went over 80 simultaneous calls.
>>> Now freeswitch starts sending packets with huge delays ( even when establishing the call , mainly the 200 ) and firing up the IVR with tons of delay up to 20 seconds.
>>>
>>> So i searched the wiki forums and mailing list:
>>>
>>> Put freeswitch on a diet , trimmed modules.conf
>>> Played with the ulimit stuff.
>>> Played with the IVRS to reduce load to a minimum and i was able to squeeze more 5 calls of performance.
>>>
>>> The problem is :
>>>
>>> Top shows
>>> top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92, 1.76, 1.78
>>> Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0 zombie
>>> Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,  0.5%si,  0.0%st
>>> Mem:   8193336k total,  1639156k used,  6554180k free,   177208k buffers
>>> Swap: 19534904k total,        0k used, 19534904k free,  1062272k cached
>>>
>>>    PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
>>> 31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85 freeswitch
>>>
>>> Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
>>>
>>>
>>> Some basic System Info :
>>> Debian 6.0 ( i heard the timming module is affected by Debian , but if the CPU % gets lower than 95% everything will be more stable)
>>> Python 2.5
>>>
>>> 2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
>>> 8 GB of Ram
>>>
>>> as you can see 94 % of the "Cpu Power" is sleeping :\
>>>
>>>
>>> It appears freeswitch is only capable of using let's say "one cpu"/thread ??
>>> Do you guys recommend simply starting more instances or redoing the IVR stuff.
>>>
>>>
>>> Hope you guys can help me out.
>>>
>>> Thanks
>>> Ant?nio Teixeira
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> --
>> Ing. Christian L?schenkohl
>> Technische Leitung, Forschung&  Entwicklung VoIP
>>
>> xpirio
>> Telekommunikation&  Service GmbH
>> Lakeside B04
>> 9020 Klagenfurt
>> Austria
>>
>> T  +43 5 77 11 - 1000
>> F  +43 5 77 11 - 1002
>> E  christian.loeschenkohl at xpirio.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

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


From i.ivanova at mastervoice.it  Fri Apr 22 14:34:33 2011
From: i.ivanova at mastervoice.it (Irina Ivanova)
Date: Fri, 22 Apr 2011 12:34:33 +0200
Subject: [Freeswitch-users] transfer_ringback triggered by "183 Session
 progress"
In-Reply-To: <C1AC4D08-A4CC-49A1-8331-318DD18DF7FB@freeswitch.org>
References: <4DADA951.30708@mastervoice.it>
	<C1AC4D08-A4CC-49A1-8331-318DD18DF7FB@freeswitch.org>
Message-ID: <4DB159B9.80105@mastervoice.it>

Thank you for the answer. I tried with both ignore_early_media=true and ignore_early_media=false, but unfortunately in both cases the behaviour is the same, the ringback_transfer is triggered.
More than this, I've noticed that also with normal outgoing calls when the channel is not answered before executing the bridge there is the similar behaviour. As soon as 183 is received, the ringtone generated by freeswitch can be heard (I suppose it is generated by freeswitch because looking on the tcpdump of the call no any RTP stream sent to freeswitch could be seen).

In my first test ignore_early_media is set to false.
As can be seen from the log here http://pastebin.freeswitch.org/16155 the line 168 says "entering state [early][180]" and at this moment the caller starts to hear ringing.

165. 2011-04-22 11:30:45.931314 [DEBUG] sofia.c:4659 Channel sofia/device/3382XXXXXX entering state [proceeding][183]
166. 2011-04-22 11:30:45.931314 [NOTICE] sofia.c:4737 Ring-Ready sofia/device/3382XXXXXX!
167. 2011-04-22 11:30:45.933870 [NOTICE] mod_sofia.c:2185 Ring-Ready sofia/internal/15 at 192.168.1.124!
168. 2011-04-22 11:30:45.933870 [DEBUG] sofia.c:4659 Channel sofia/internal/15 at 192.168.1.124 entering state [early][180]

The second test is made with ignore_early_media set to true.
Here is the log of the test: http://pastebin.freeswitch.org/16156. On the line 167 I can still see "entering state [early][180]" and I can still hear the ringing.
167. 2011-04-22 12:05:21.450866 [DEBUG] sofia.c:4659 Channel sofia/internal/15 at 192.168.1.124 entering state [early][180]

I suppose it is meant to be like this, I am just wondering if there is a way to avoid this short ringing sound that could disturb the callers. May be some workaround?...
I also played with uuid_broadcast and uuid_displace trying to substitute the audio stream which is played to the caller as soon as channel enters the state early, but at the end didn't get the desired result.



On 04/21/2011 04:17 PM, Brian West wrote:
> Happen to be setting ignore_early_media=true?
>
>
> On Apr 19, 2011, at 10:25 AM, Irina Ivanova wrote:
>
>> Hi,
>>
>> I've noticed that if to set transfer_ringback (to any ringback tone) for
>> already answered call and then to execute the bridge to some external
>> number through the gateway, the ringing is triggered not only by "180
>> Ringing" SIP response, but also when "183 Session progress" is received.
>> Does anybody know if there is a way to make transfer_ringback not to be
>> triggered by 183? I need it because in the case when the destination
>> number is busy and provider sends me 183 and then 486 (Busy here) the
>> caller hears one ringback tone and then the busy tone which makes an
>> impression that the called party rejected the call.
>>
>> Thanks,
>> Irina
>>
>> -- 
>> ================================================================
>>
>> Distinti saluti
>> --
>>
>> Irina Ivanova
>> Settore Sviluppo MasterVoice
>>
>> tel: +39 0522 1590100
>> fax: +39 0522 331673
>> mob: +39 334 6449290
>> e-mail: i.ivanova at mastervoice.it
>> web: www.mastertraining.it - www.registroelettronico.com
>>
>> Master Training S.r.l.
>> Sede Legale: via Timolini, 18 - Correggio (RE) - Italy
>> Sede Operativa: via Sani, 15 - Reggio Emilia - Italy
>> Sede Commerciale: via Sani, 9 - Reggio Emilia - Italy
>>
>> ================================================================
>> Le informazioni contenute in questa e-mail sono da considerarsi confidenziali e esclusivamente per uso personale dei destinatari sopra indicati. Questo messaggio pu? includere dati personali o sensibili. Qualora questo messaggio fosse da Voi ricevuto per errore vogliate cortesemente darcene notizia a mezzo e-mail e distruggere il messaggio ricevuto erroneamente. Quanto precede ai fini del rispetto del Decreto Legislativo 196/2003 sulla tutela dei dati personali e sensibili.
>> This e-mail and any file transmitted with it is intended only for the person or entity to which is addressed and may contain information that is privileged, confidential or otherwise protected from disclosure.Copying, dissemination or use of this e-mail or the information herein by anyone other than the intended recipient is prohibited. If you have received this e-mail by mistake, please notify us immediately by telephone or fax.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
================================================================

Distinti saluti
--

Irina Ivanova
Settore Sviluppo MasterVoice

tel: +39 0522 1590100
fax: +39 0522 331673
mob: +39 334 6449290
e-mail: i.ivanova at mastervoice.it
web: www.mastertraining.it - www.registroelettronico.com

Master Training S.r.l.
Sede Legale: via Timolini, 18 - Correggio (RE) - Italy
Sede Operativa: via Sani, 15 - Reggio Emilia - Italy
Sede Commerciale: via Sani, 9 - Reggio Emilia - Italy

================================================================
Le informazioni contenute in questa e-mail sono da considerarsi confidenziali e esclusivamente per uso personale dei destinatari sopra indicati. Questo messaggio pu? includere dati personali o sensibili. Qualora questo messaggio fosse da Voi ricevuto per errore vogliate cortesemente darcene notizia a mezzo e-mail e distruggere il messaggio ricevuto erroneamente. Quanto precede ai fini del rispetto del Decreto Legislativo 196/2003 sulla tutela dei dati personali e sensibili.
This e-mail and any file transmitted with it is intended only for the person or entity to which is addressed and may contain information that is privileged, confidential or otherwise protected from disclosure.Copying, dissemination or use of this e-mail or the information herein by anyone other than the intended recipient is prohibited. If you have received this e-mail by mistake, please notify us immediately by telephone or fax.



From Prometheus001 at gmx.net  Fri Apr 22 14:44:48 2011
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 22 Apr 2011 12:44:48 +0200
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
In-Reply-To: <4DAECCA8.1050203@gmx.net>
References: <4DAECCA8.1050203@gmx.net>
Message-ID: <4DB15C20.5070208@gmx.net>

Nobody has an idea?


Peter P GMX schrieb:
> Hello,
>
> I am trying to use the mod_callcenter functionality. This works nicely
> so far so thank you to everybody involved for programming this nice module!
> But I am stuck somehow with uuid-standby. Can anybody explain how
> uuid-standby works?
>
> Another question: In the Asterisk based callcenter solution named
> "Vicidial", an agent can be held permanently in a conference, waiting
> for calls who are bridged to his uuid in the conference. Can this be
> haviour be done with mod_callcenter?
>
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From david.ponzone at ipeva.fr  Fri Apr 22 15:54:38 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Fri, 22 Apr 2011 13:54:38 +0200
Subject: [Freeswitch-users] CallsIN counter on gateway
In-Reply-To: <FE773AA5-E64D-45FE-ABC8-79B699D6F962@scarlet-internet.nl>
References: <10717914-15BF-4D58-9A77-8FAE698661D8@scarlet-internet.nl>
	<BD86134D-C6FC-42FF-A643-A3F8C564EB3E@ipeva.fr>
	<FE773AA5-E64D-45FE-ABC8-79B699D6F962@scarlet-internet.nl>
Message-ID: <21612C6E-ACF9-4CC7-97B4-E08DE931D953@ipeva.fr>

As far as I know, the source IP of an inbound call is not matched against the ones configured in the gateway definitions in order to determine which is the relevant gateway, and so possibily alter some parameters like the context or else.
I'd like that too, that would simplify management and provisionning, and clarity also.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 22/04/2011 ? 11:32, Leon de Rooij a ?crit :

> Hello David,
> 
> Thanks for your reply.
> 
> I think I didn't word my question properly. Just as you, I also don't want to rely on the headers in the inbound SIP INVITEs that get sent to me for distinguishing carriers, but only want to do this on the basis of the originating IP address. Right now, I have this solved by having a separate Sofia profile that sends the calls to a context where I have some conditions on network_addr to make this distinction. This works, but it would be nicer if it could be done in the gateway definition in the sip profile config itself - that way the CallsIN counter would get increased.
> 
> Perhaps I should really start using proxies in front of my FS boxes, so I can change the SIP headers to contain gw+ or gw= ? But I'd really like to do this directly in FS as well - if at all possible at the moment.
> 
> Any thoughts ?
> 
> Thanks,
> 
> Leon
> 
> 
> 
> 
> On Apr 22, 2011, at 10:33 AM, David Ponzone wrote:
> 
>> Leon,
>> 
>> Personally, as I know I can't always expect to get the right headers in the INVITE to do this matching, and also because I think it's only valid for gateways you register to, I prefer to split the incoming INVITEs based on source IP.
>> It's also the opportunity to normalize what you need to normalize (caller number, called number) and to inc counters, before transfering to the desired context.
>> Also, I would not want my dialplan to rely on the good behaviour of the remote gateway, which could decide suddenly to send crap, or a different crap than before, because the guys there decided to upgrade their S...S or they A..E priceless boxes over the weekend.
>> 
>> Sorry if I misunderstood your question.
>> 
>> David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>> 
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>> 
>> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>> 
>> 
>> 
>> 
>> Le 20/04/2011 ? 12:59, Leon de Rooij a ?crit :
>> 
>>> Hi all,
>>> 
>>> I have a gateway that sends me invites with the callerid in the from: header and the destination number in the to: header. I'd like these inbound invites to be matched to the proper gateway (that is used for outbound calls) so that I can use params of the gateway like context, but also so the CallsIN counter gets increased for each inbound call.
>>> 
>>> Is this at all possible ? From what I understand from sofia.c, the gateway->ib_calls is only increased when an incoming request contains a gw= param or when destination_number is prefixed by gw+, or is there another way ?
>>> 
>>> thanks & kind regards,
>>> 
>>> Leon de Rooij
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Fri Apr 22 17:00:55 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Fri, 22 Apr 2011 06:00:55 -0700 (PDT)
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
In-Reply-To: <4DB15C20.5070208@gmx.net>
References: <4DAECCA8.1050203@gmx.net> <4DB15C20.5070208@gmx.net>
Message-ID: <1303477255577-6296981.post@n2.nabble.com>


Peter P GMX wrote:
> Nobody has an idea?
Yes, how about be a little patient? If you want an interactive response, try
#freeswitch IRC on freenode. There you will find hundreds of FreeSWITCH
users who can respond to your request quicker.

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-callcenter-and-uuid-standby-tp6290465p6296981.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jcasale at activenetwerx.com  Fri Apr 22 19:04:09 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Fri, 22 Apr 2011 15:04:09 +0000
Subject: [Freeswitch-users] Changing From header in voicemail to
	e-mail	notifications
In-Reply-To: <BANLkTi=i9ycE6TYEwR38Kwc08OdtasZa4g@mail.gmail.com>
References: <SNT116-W1509E05D218F485CAC25029D920@phx.gbl>
	<BANLkTi=i9ycE6TYEwR38Kwc08OdtasZa4g@mail.gmail.com>
Message-ID: <assp.0093417dcc.92097A6A775D5147B1078E3F15430B923EF3A7@prato.activenetwerx.local>

>? You have to set the following in vars,xml:
>?
>? <X-PRE-PROCESS cmd="set" data="voicemail_account=username"/>
>? <X-PRE-PROCESS cmd="set" data="voicemail_domain=your-domain.com"/>

In one of my installs, (1.0.6) that never worked. Is that only valid in trunk?


From zetruger at gmail.com  Fri Apr 22 12:17:24 2011
From: zetruger at gmail.com (=?KOI8-R?B?6dfBziD+ydPU0cvP1w==?=)
Date: Fri, 22 Apr 2011 12:17:24 +0400
Subject: [Freeswitch-users] HOLD - signals from phones? How to play MoH?
Message-ID: <BANLkTikc5sR-7saccTXnVGiut2R7kwEY-w@mail.gmail.com>

In a bridge mode, when i push a button hold, FreeSWITCH don't playing MoH.
But if I set hold through FS console it's works.


From boris at tagnet.ru  Fri Apr 22 20:49:26 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 22 Apr 2011 22:49:26 +0600
Subject: [Freeswitch-users] Need a little bit clarification about gateways
Message-ID: <4DB1B196.8030802@tagnet.ru>

Hello!

     Am I right, if I created gateway into sofia profile and set 
register=false I also should permit its IP in the ACL to receive a calls 
from it? Or there is some param to gateway for automatic acl creation? 
Like cidr for a user?

-- 
With respect,
Boris




From curriegrad2004 at gmail.com  Fri Apr 22 21:03:44 2011
From: curriegrad2004 at gmail.com (curriegrad2004)
Date: Fri, 22 Apr 2011 10:03:44 -0700
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <4DB1550C.1050607@xpirio.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<4DB068C4.90109@xpirio.com>
	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
	<4DB1550C.1050607@xpirio.com>
Message-ID: <BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>

Freeswitch is targeted for CentOS 5.3, which in my opinion quite short
sighted for the developers to do this. However with the limited size
of developers and testers, I'm afraid there's not much platforms we
can throughly test and actually say "okay, FS will run flawlessly on X
distro"

However you can always try messing with the CFLAG's mtune option and
see what it produces for you...

2011/4/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
> hi
>
> if you refer to my e-mail
>
> yes, we do use tmpfs on both variants but
> - delays occur with concurrent calls > 80-100
> - cps is limited to 5-10 on debian, with centos 30 cps and more are no problem at all
>
> also cpu load, stability and overall performace have been much better since using centos
>
> i just found out for me that debian works not as good for me as centos does.
> btw. everywhere else debian is 1st choice (desktop, lamp, db etc.)
>
> br
>
>
> On 2011-04-21 23:04, Jay Binks wrote:
>
>> I have no such problems on debian .
>>
>> I use debian 5 with 2.6.18 kernel which is what Is recommended
>>
>> Are you using tmpfs ??
>>
>> Jay
>>
>>
>>
>> On 22/04/2011, at 3:26 AM, Christian L?schenkohl<christian.loeschenkohl at xpirio.com> ?wrote:
>>
>>> hi
>>>
>>> we did use debian too and had such performance issues (sip packet delays, low cps).
>>> after using centos 64bit (as advised by the devs) all performance problems are gone.
>>>
>>> br
>>>
>>> On 2011-04-21 18:24, Antonio Teixeira wrote:
>>>
>>>> Hello List.
>>>>
>>>> I'm currently integrating an IVR in python together with freeswitch using mod_python and ESL and my life has been well until ...
>>>> The flow of calls went over 80 simultaneous calls.
>>>> Now freeswitch starts sending packets with huge delays ( even when establishing the call , mainly the 200 ) and firing up the IVR with tons of delay up to 20 seconds.
>>>>
>>>> So i searched the wiki forums and mailing list:
>>>>
>>>> Put freeswitch on a diet , trimmed modules.conf
>>>> Played with the ulimit stuff.
>>>> Played with the IVRS to reduce load to a minimum and i was able to squeeze more 5 calls of performance.
>>>>
>>>> The problem is :
>>>>
>>>> Top shows
>>>> top - 16:14:33 up 35 days, ?8:15, ?3 users, ?load average: 1.92, 1.76, 1.78
>>>> Tasks: 133 total, ? 1 running, 132 sleeping, ? 0 stopped, ? 0 zombie
>>>> Cpu(s): ?1.4%us, ?3.3%sy, ?0.0%ni, 94.6%id, ?0.0%wa, ?0.3%hi, ?0.5%si, ?0.0%st
>>>> Mem: ? 8193336k total, ?1639156k used, ?6554180k free, ? 177208k buffers
>>>> Swap: 19534904k total, ? ? ? ?0k used, 19534904k free, ?1062272k cached
>>>>
>>>> ? ?PID USER ? ? ?PR ?NI ?VIRT ?RES ?SHR S %CPU %MEM ? ?TIME+ ?COMMAND
>>>> 31361 yadayada ? ? ?20 ? 0 ?716m 164m 9628 S ? 73 ?2.1 155:17.85 freeswitch
>>>>
>>>> Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
>>>>
>>>>
>>>> Some basic System Info :
>>>> Debian 6.0 ( i heard the timming module is affected by Debian , but if the CPU % gets lower than 95% everything will be more stable)
>>>> Python 2.5
>>>>
>>>> 2 x Intel(R) Xeon(R) CPU ? ? ? ? ? E5506 ?@ 2.13GHz
>>>> 8 GB of Ram
>>>>
>>>> as you can see 94 % of the "Cpu Power" is sleeping :\
>>>>
>>>>
>>>> It appears freeswitch is only capable of using let's say "one cpu"/thread ??
>>>> Do you guys recommend simply starting more instances or redoing the IVR stuff.
>>>>
>>>>
>>>> Hope you guys can help me out.
>>>>
>>>> Thanks
>>>> Ant?nio Teixeira
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> Ing. Christian L?schenkohl
>>> Technische Leitung, Forschung& ?Entwicklung VoIP
>>>
>>> xpirio
>>> Telekommunikation& ?Service GmbH
>>> Lakeside B04
>>> 9020 Klagenfurt
>>> Austria
>>>
>>> T ?+43 5 77 11 - 1000
>>> F ?+43 5 77 11 - 1002
>>> E ?christian.loeschenkohl at xpirio.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T ?+43 5 77 11 - 1000
> F ?+43 5 77 11 - 1002
> E ?christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From eric at loopfx.com  Fri Apr 22 22:02:40 2011
From: eric at loopfx.com (Eric Beard)
Date: Fri, 22 Apr 2011 14:02:40 -0400
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<4DB068C4.90109@xpirio.com>	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
	<4DB1550C.1050607@xpirio.com>
	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DF4D19B3@doctorbob.loop.local>

FreeSwitch is working quite well for me on openSuse 11.4.  I'm actually running it on a default desktop install of openSuse, with no changes to the OS at all other than installing FS and its dependencies.  I'm sustaining 165 concurrent calls with no problems.

-----------------------
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 curriegrad2004
Sent: Friday, April 22, 2011 1:04 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Tuning Up Freeswitch

Freeswitch is targeted for CentOS 5.3, which in my opinion quite short
sighted for the developers to do this. However with the limited size
of developers and testers, I'm afraid there's not much platforms we
can throughly test and actually say "okay, FS will run flawlessly on X
distro"

However you can always try messing with the CFLAG's mtune option and
see what it produces for you...

2011/4/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
> hi
>
> if you refer to my e-mail
>
> yes, we do use tmpfs on both variants but
> - delays occur with concurrent calls > 80-100
> - cps is limited to 5-10 on debian, with centos 30 cps and more are no problem at all
>
> also cpu load, stability and overall performace have been much better since using centos
>
> i just found out for me that debian works not as good for me as centos does.
> btw. everywhere else debian is 1st choice (desktop, lamp, db etc.)
>
> br
>
>
> On 2011-04-21 23:04, Jay Binks wrote:
>
>> I have no such problems on debian .
>>
>> I use debian 5 with 2.6.18 kernel which is what Is recommended
>>
>> Are you using tmpfs ??
>>
>> Jay
>>
>>
>>
>> On 22/04/2011, at 3:26 AM, Christian L?schenkohl<christian.loeschenkohl at xpirio.com> ?wrote:
>>
>>> hi
>>>
>>> we did use debian too and had such performance issues (sip packet delays, low cps).
>>> after using centos 64bit (as advised by the devs) all performance problems are gone.
>>>
>>> br
>>>
>>> On 2011-04-21 18:24, Antonio Teixeira wrote:
>>>
>>>> Hello List.
>>>>
>>>> I'm currently integrating an IVR in python together with freeswitch using mod_python and ESL and my life has been well until ...
>>>> The flow of calls went over 80 simultaneous calls.
>>>> Now freeswitch starts sending packets with huge delays ( even when establishing the call , mainly the 200 ) and firing up the IVR with tons of delay up to 20 seconds.
>>>>
>>>> So i searched the wiki forums and mailing list:
>>>>
>>>> Put freeswitch on a diet , trimmed modules.conf
>>>> Played with the ulimit stuff.
>>>> Played with the IVRS to reduce load to a minimum and i was able to squeeze more 5 calls of performance.
>>>>
>>>> The problem is :
>>>>
>>>> Top shows
>>>> top - 16:14:33 up 35 days, ?8:15, ?3 users, ?load average: 1.92, 1.76, 1.78
>>>> Tasks: 133 total, ? 1 running, 132 sleeping, ? 0 stopped, ? 0 zombie
>>>> Cpu(s): ?1.4%us, ?3.3%sy, ?0.0%ni, 94.6%id, ?0.0%wa, ?0.3%hi, ?0.5%si, ?0.0%st
>>>> Mem: ? 8193336k total, ?1639156k used, ?6554180k free, ? 177208k buffers
>>>> Swap: 19534904k total, ? ? ? ?0k used, 19534904k free, ?1062272k cached
>>>>
>>>> ? ?PID USER ? ? ?PR ?NI ?VIRT ?RES ?SHR S %CPU %MEM ? ?TIME+ ?COMMAND
>>>> 31361 yadayada ? ? ?20 ? 0 ?716m 164m 9628 S ? 73 ?2.1 155:17.85 freeswitch
>>>>
>>>> Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
>>>>
>>>>
>>>> Some basic System Info :
>>>> Debian 6.0 ( i heard the timming module is affected by Debian , but if the CPU % gets lower than 95% everything will be more stable)
>>>> Python 2.5
>>>>
>>>> 2 x Intel(R) Xeon(R) CPU ? ? ? ? ? E5506 ?@ 2.13GHz
>>>> 8 GB of Ram
>>>>
>>>> as you can see 94 % of the "Cpu Power" is sleeping :\
>>>>
>>>>
>>>> It appears freeswitch is only capable of using let's say "one cpu"/thread ??
>>>> Do you guys recommend simply starting more instances or redoing the IVR stuff.
>>>>
>>>>
>>>> Hope you guys can help me out.
>>>>
>>>> Thanks
>>>> Ant?nio Teixeira
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> --
>>> Ing. Christian L?schenkohl
>>> Technische Leitung, Forschung& ?Entwicklung VoIP
>>>
>>> xpirio
>>> Telekommunikation& ?Service GmbH
>>> Lakeside B04
>>> 9020 Klagenfurt
>>> Austria
>>>
>>> T ?+43 5 77 11 - 1000
>>> F ?+43 5 77 11 - 1002
>>> E ?christian.loeschenkohl at xpirio.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T ?+43 5 77 11 - 1000
> F ?+43 5 77 11 - 1002
> E ?christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

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

From brian at freeswitch.org  Fri Apr 22 22:17:59 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 22 Apr 2011 13:17:59 -0500
Subject: [Freeswitch-users] HOLD - signals from phones? How to play MoH?
In-Reply-To: <BANLkTikc5sR-7saccTXnVGiut2R7kwEY-w@mail.gmail.com>
References: <BANLkTikc5sR-7saccTXnVGiut2R7kwEY-w@mail.gmail.com>
Message-ID: <EFCB5275-7FF4-460A-9381-145B45E73B06@freeswitch.org>

What phones are involved?

/b

On Apr 22, 2011, at 3:17 AM, ???? ???????? wrote:

> In a bridge mode, when i push a button hold, FreeSWITCH don't playing MoH.
> But if I set hold through FS console it's works.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Fri Apr 22 22:18:24 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 22 Apr 2011 13:18:24 -0500
Subject: [Freeswitch-users] Need a little bit clarification about
	gateways
In-Reply-To: <4DB1B196.8030802@tagnet.ru>
References: <4DB1B196.8030802@tagnet.ru>
Message-ID: <9BD70C31-770C-428F-8BAE-D56040C5B257@freeswitch.org>

you should only add the ACL if you have auth-calls=true.

/b

On Apr 22, 2011, at 11:49 AM, Boris Kovalenko wrote:

> Hello!
> 
>     Am I right, if I created gateway into sofia profile and set 
> register=false I also should permit its IP in the ACL to receive a calls 
> from it? Or there is some param to gateway for automatic acl creation? 
> Like cidr for a user?
> 
> -- 
> With respect,
> Boris



From msc at freeswitch.org  Fri Apr 22 22:34:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Apr 2011 11:34:29 -0700
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
In-Reply-To: <4DB15C20.5070208@gmx.net>
References: <4DAECCA8.1050203@gmx.net>
	<4DB15C20.5070208@gmx.net>
Message-ID: <BANLkTinP0zsE4PWniZWZPkyBzf1hbU-WSw@mail.gmail.com>

On Fri, Apr 22, 2011 at 3:44 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Nobody has an idea?
>
> You need to get a hold of Moc for this one...

-MC


>
> Peter P GMX schrieb:
> > Hello,
> >
> > I am trying to use the mod_callcenter functionality. This works nicely
> > so far so thank you to everybody involved for programming this nice
> module!
> > But I am stuck somehow with uuid-standby. Can anybody explain how
> > uuid-standby works?
> >
> > Another question: In the Asterisk based callcenter solution named
> > "Vicidial", an agent can be held permanently in a conference, waiting
> > for calls who are bridged to his uuid in the conference. Can this be
> > haviour be done with mod_callcenter?
> >
> >
> > Best regards
> > Peter
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vibha_dear6 at yahoo.co.in  Fri Apr 22 22:38:24 2011
From: vibha_dear6 at yahoo.co.in (vibha dear)
Date: Sat, 23 Apr 2011 00:08:24 +0530 (IST)
Subject: [Freeswitch-users] freeswitch xml cdr and csv
Message-ID: <307668.83671.qm@web137305.mail.in.yahoo.com>

hi all..!
freeswitch generated cdr in both formats xml an multiline csv format, will there be any parsing overhead in XML parsing while fetching information for reporting and billing? how they will effect the system performance? which module is better to generate CDRs?
thanks in advance..!!kasturi ?:)
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From msc at freeswitch.org  Fri Apr 22 22:38:29 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Apr 2011 11:38:29 -0700
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<4DB068C4.90109@xpirio.com>
	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
	<4DB1550C.1050607@xpirio.com>
	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
Message-ID: <BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>

FreeSWITCH runs well on many platforms. However, the devs are painfully
aware that bleeding edge distros have bleeding edge gcc compilers and other
interesting issues. That being said, CentOS 5.x is "stable" in that it's old
and boring, therefore it has the least amount of drama. OTOH, some of our
users have been having great success with Arch Linux (IRC:bougeyman) and FS,
even though Arch uses very recent kernels.

Bottom line: if you know what you're doing then you can probably run FS
anywhere. If you don't know what you're doing then stick with CentOS 5.x or
Debian Lenny until you do. (I run then both with zero issues, compiling
latest git each day.)

-MC

On Fri, Apr 22, 2011 at 10:03 AM, curriegrad2004
<curriegrad2004 at gmail.com>wrote:

> Freeswitch is targeted for CentOS 5.3, which in my opinion quite short
> sighted for the developers to do this. However with the limited size
> of developers and testers, I'm afraid there's not much platforms we
> can throughly test and actually say "okay, FS will run flawlessly on X
> distro"
>
> However you can always try messing with the CFLAG's mtune option and
> see what it produces for you...
>
> 2011/4/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
> > hi
> >
> > if you refer to my e-mail
> >
> > yes, we do use tmpfs on both variants but
> > - delays occur with concurrent calls > 80-100
> > - cps is limited to 5-10 on debian, with centos 30 cps and more are no
> problem at all
> >
> > also cpu load, stability and overall performace have been much better
> since using centos
> >
> > i just found out for me that debian works not as good for me as centos
> does.
> > btw. everywhere else debian is 1st choice (desktop, lamp, db etc.)
> >
> > br
> >
> >
> > On 2011-04-21 23:04, Jay Binks wrote:
> >
> >> I have no such problems on debian .
> >>
> >> I use debian 5 with 2.6.18 kernel which is what Is recommended
> >>
> >> Are you using tmpfs ??
> >>
> >> Jay
> >>
> >>
> >>
> >> On 22/04/2011, at 3:26 AM, Christian L?schenkohl<
> christian.loeschenkohl at xpirio.com>  wrote:
> >>
> >>> hi
> >>>
> >>> we did use debian too and had such performance issues (sip packet
> delays, low cps).
> >>> after using centos 64bit (as advised by the devs) all performance
> problems are gone.
> >>>
> >>> br
> >>>
> >>> On 2011-04-21 18:24, Antonio Teixeira wrote:
> >>>
> >>>> Hello List.
> >>>>
> >>>> I'm currently integrating an IVR in python together with freeswitch
> using mod_python and ESL and my life has been well until ...
> >>>> The flow of calls went over 80 simultaneous calls.
> >>>> Now freeswitch starts sending packets with huge delays ( even when
> establishing the call , mainly the 200 ) and firing up the IVR with tons of
> delay up to 20 seconds.
> >>>>
> >>>> So i searched the wiki forums and mailing list:
> >>>>
> >>>> Put freeswitch on a diet , trimmed modules.conf
> >>>> Played with the ulimit stuff.
> >>>> Played with the IVRS to reduce load to a minimum and i was able to
> squeeze more 5 calls of performance.
> >>>>
> >>>> The problem is :
> >>>>
> >>>> Top shows
> >>>> top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92, 1.76,
> 1.78
> >>>> Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0 zombie
> >>>> Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,  0.5%si,
>  0.0%st
> >>>> Mem:   8193336k total,  1639156k used,  6554180k free,   177208k
> buffers
> >>>> Swap: 19534904k total,        0k used, 19534904k free,  1062272k
> cached
> >>>>
> >>>>    PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
> >>>> 31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85
> freeswitch
> >>>>
> >>>> Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
> >>>>
> >>>>
> >>>> Some basic System Info :
> >>>> Debian 6.0 ( i heard the timming module is affected by Debian , but if
> the CPU % gets lower than 95% everything will be more stable)
> >>>> Python 2.5
> >>>>
> >>>> 2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
> >>>> 8 GB of Ram
> >>>>
> >>>> as you can see 94 % of the "Cpu Power" is sleeping :\
> >>>>
> >>>>
> >>>> It appears freeswitch is only capable of using let's say "one
> cpu"/thread ??
> >>>> Do you guys recommend simply starting more instances or redoing the
> IVR stuff.
> >>>>
> >>>>
> >>>> Hope you guys can help me out.
> >>>>
> >>>> Thanks
> >>>> Ant?nio Teixeira
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>> --
> >>> Ing. Christian L?schenkohl
> >>> Technische Leitung, Forschung&  Entwicklung VoIP
> >>>
> >>> xpirio
> >>> Telekommunikation&  Service GmbH
> >>> Lakeside B04
> >>> 9020 Klagenfurt
> >>> Austria
> >>>
> >>> T  +43 5 77 11 - 1000
> >>> F  +43 5 77 11 - 1002
> >>> E  christian.loeschenkohl at xpirio.com
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > --
> > Ing. Christian L?schenkohl
> > Technische Leitung, Forschung & Entwicklung VoIP
> >
> > xpirio
> > Telekommunikation & Service GmbH
> > Lakeside B04
> > 9020 Klagenfurt
> > Austria
> >
> > T  +43 5 77 11 - 1000
> > F  +43 5 77 11 - 1002
> > E  christian.loeschenkohl at xpirio.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Apr 22 22:44:10 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Apr 2011 11:44:10 -0700
Subject: [Freeswitch-users] ZRTP Library
In-Reply-To: <BANLkTimBuiOKCZQ9AQFFtKJg23C0CGhEmQ@mail.gmail.com>
References: <BANLkTimBuiOKCZQ9AQFFtKJg23C0CGhEmQ@mail.gmail.com>
Message-ID: <BANLkTimggX_sBqxWnZENNfPK+hQ+Rj0afQ@mail.gmail.com>

I don't know if it's okay to distribute. I've pinged PRZ on this issue and
have not heard back from him. Sorry...

-MC

On Thu, Apr 21, 2011 at 8:44 PM, Nasir Iqbal <nasir at ictinnovations.com>wrote:

> Hi,
>
> As zfone download server is offline, is there anyone who can provide me
> libzrtp SDK source?
>
>
> Nasir Iqbal
>
> ICT Innovations
> http://www.ictinnovations.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 22 22:46:00 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Apr 2011 11:46:00 -0700
Subject: [Freeswitch-users] freeswitch xml cdr and csv
In-Reply-To: <307668.83671.qm@web137305.mail.in.yahoo.com>
References: <307668.83671.qm@web137305.mail.in.yahoo.com>
Message-ID: <BANLkTi=CmcK5HiMTpwdb96ysMB0ZdNd5ew@mail.gmail.com>

XML CDR is quite robust. It will post to a web server and/or store on disk.
We've got people doing hundreds of thousands of calls per day using XML CDR.
However, they aren't doing the parsing on their FS server, they offload that
to a separate server, which is the wise thing to do if you have lots of
traffic.

-MC

On Fri, Apr 22, 2011 at 11:38 AM, vibha dear <vibha_dear6 at yahoo.co.in>wrote:

> hi all..!
>
> freeswitch generated cdr in both formats xml an multiline csv format, will
> there be any parsing overhead in XML parsing while fetching information for
> reporting and billing? how they will effect the system performance? which
> module is better to generate CDRs?
>
> thanks in advance..!!
> kasturi  :)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 eagle.antonio at gmail.com  Fri Apr 22 23:39:12 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Fri, 22 Apr 2011 20:39:12 +0100
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<4DB068C4.90109@xpirio.com>
	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
	<4DB1550C.1050607@xpirio.com>
	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
	<BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>
Message-ID: <BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com>

Well i totally agree with the Dev team decision , it would be impossible for
a DEV team to "get it right" in all the distros that's why i started this
post.

But we also need to take into consideration that we are talking about IVR
Processing ,  not auto-*attendants* or simply call pass trough.
This means heavy use of TTS / ASR , Database Logic and Scripts , GetDigits
and the works.

I use python alot , but i think the mod_python is not the best tool for this
job i admit that , that's why im currently looking for other solutions.
I'm thinking in using mod_socket , but that scares me ( let? say bad
experiences with Asterisk AGI) Or Mod Curl

The main problem is that some IVR are extremely complex , like
questionnaires , etc.

It would be great if we could mod_event_zmq to control the calls [?]

Just to Sum it UP so far , so i can later add it to the wiki.

Use Cent OS 64.
Use tmpfs for all the databases.

Thank you all for helping and Happy Eastern.
Ant?nio Teixeira




2011/4/22 Michael Collins <msc at freeswitch.org>

> FreeSWITCH runs well on many platforms. However, the devs are painfully
> aware that bleeding edge distros have bleeding edge gcc compilers and other
> interesting issues. That being said, CentOS 5.x is "stable" in that it's old
> and boring, therefore it has the least amount of drama. OTOH, some of our
> users have been having great success with Arch Linux (IRC:bougeyman) and FS,
> even though Arch uses very recent kernels.
>
> Bottom line: if you know what you're doing then you can probably run FS
> anywhere. If you don't know what you're doing then stick with CentOS 5.x or
> Debian Lenny until you do. (I run then both with zero issues, compiling
> latest git each day.)
>
> -MC
>
>
> On Fri, Apr 22, 2011 at 10:03 AM, curriegrad2004 <curriegrad2004 at gmail.com
> > wrote:
>
>> Freeswitch is targeted for CentOS 5.3, which in my opinion quite short
>> sighted for the developers to do this. However with the limited size
>> of developers and testers, I'm afraid there's not much platforms we
>> can throughly test and actually say "okay, FS will run flawlessly on X
>> distro"
>>
>> However you can always try messing with the CFLAG's mtune option and
>> see what it produces for you...
>>
>> 2011/4/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
>> > hi
>> >
>> > if you refer to my e-mail
>> >
>> > yes, we do use tmpfs on both variants but
>> > - delays occur with concurrent calls > 80-100
>> > - cps is limited to 5-10 on debian, with centos 30 cps and more are no
>> problem at all
>> >
>> > also cpu load, stability and overall performace have been much better
>> since using centos
>> >
>> > i just found out for me that debian works not as good for me as centos
>> does.
>> > btw. everywhere else debian is 1st choice (desktop, lamp, db etc.)
>> >
>> > br
>> >
>> >
>> > On 2011-04-21 23:04, Jay Binks wrote:
>> >
>> >> I have no such problems on debian .
>> >>
>> >> I use debian 5 with 2.6.18 kernel which is what Is recommended
>> >>
>> >> Are you using tmpfs ??
>> >>
>> >> Jay
>> >>
>> >>
>> >>
>> >> On 22/04/2011, at 3:26 AM, Christian L?schenkohl<
>> christian.loeschenkohl at xpirio.com>  wrote:
>> >>
>> >>> hi
>> >>>
>> >>> we did use debian too and had such performance issues (sip packet
>> delays, low cps).
>> >>> after using centos 64bit (as advised by the devs) all performance
>> problems are gone.
>> >>>
>> >>> br
>> >>>
>> >>> On 2011-04-21 18:24, Antonio Teixeira wrote:
>> >>>
>> >>>> Hello List.
>> >>>>
>> >>>> I'm currently integrating an IVR in python together with freeswitch
>> using mod_python and ESL and my life has been well until ...
>> >>>> The flow of calls went over 80 simultaneous calls.
>> >>>> Now freeswitch starts sending packets with huge delays ( even when
>> establishing the call , mainly the 200 ) and firing up the IVR with tons of
>> delay up to 20 seconds.
>> >>>>
>> >>>> So i searched the wiki forums and mailing list:
>> >>>>
>> >>>> Put freeswitch on a diet , trimmed modules.conf
>> >>>> Played with the ulimit stuff.
>> >>>> Played with the IVRS to reduce load to a minimum and i was able to
>> squeeze more 5 calls of performance.
>> >>>>
>> >>>> The problem is :
>> >>>>
>> >>>> Top shows
>> >>>> top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92,
>> 1.76, 1.78
>> >>>> Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0 zombie
>> >>>> Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,
>>  0.5%si,  0.0%st
>> >>>> Mem:   8193336k total,  1639156k used,  6554180k free,   177208k
>> buffers
>> >>>> Swap: 19534904k total,        0k used, 19534904k free,  1062272k
>> cached
>> >>>>
>> >>>>    PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
>> >>>> 31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85
>> freeswitch
>> >>>>
>> >>>> Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
>> >>>>
>> >>>>
>> >>>> Some basic System Info :
>> >>>> Debian 6.0 ( i heard the timming module is affected by Debian , but
>> if the CPU % gets lower than 95% everything will be more stable)
>> >>>> Python 2.5
>> >>>>
>> >>>> 2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
>> >>>> 8 GB of Ram
>> >>>>
>> >>>> as you can see 94 % of the "Cpu Power" is sleeping :\
>> >>>>
>> >>>>
>> >>>> It appears freeswitch is only capable of using let's say "one
>> cpu"/thread ??
>> >>>> Do you guys recommend simply starting more instances or redoing the
>> IVR stuff.
>> >>>>
>> >>>>
>> >>>> Hope you guys can help me out.
>> >>>>
>> >>>> Thanks
>> >>>> Ant?nio Teixeira
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>>
>> >>>> _______________________________________________
>> >>>> FreeSWITCH-users mailing list
>> >>>> FreeSWITCH-users at lists.freeswitch.org
>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>>> http://www.freeswitch.org
>> >>>
>> >>>
>> >>> --
>> >>> Ing. Christian L?schenkohl
>> >>> Technische Leitung, Forschung&  Entwicklung VoIP
>> >>>
>> >>> xpirio
>> >>> Telekommunikation&  Service GmbH
>> >>> Lakeside B04
>> >>> 9020 Klagenfurt
>> >>> Austria
>> >>>
>> >>> T  +43 5 77 11 - 1000
>> >>> F  +43 5 77 11 - 1002
>> >>> E  christian.loeschenkohl at xpirio.com
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > --
>> > Ing. Christian L?schenkohl
>> > Technische Leitung, Forschung & Entwicklung VoIP
>> >
>> > xpirio
>> > Telekommunikation & Service GmbH
>> > Lakeside B04
>> > 9020 Klagenfurt
>> > Austria
>> >
>> > T  +43 5 77 11 - 1000
>> > F  +43 5 77 11 - 1002
>> > E  christian.loeschenkohl at xpirio.com
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 23 00:22:45 2011
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 22 Apr 2011 15:22:45 -0500
Subject: [Freeswitch-users] freeswitch xml cdr and csv
In-Reply-To: <BANLkTi=CmcK5HiMTpwdb96ysMB0ZdNd5ew@mail.gmail.com>
Message-ID: <C9D74DC5.144E4%krice@freeswitch.org>

Not true... We parse on the FS server and handle over a million CDRs/day
using a PHP parsing script no posting to the webserver...

xml_cdr drops the xml file for us in log/xml_cdr where we have a php script
similar to below that picks them up and trudges thru them at a rate of about
30 per second... 

<? $files = glob(xml_cdr/dir/*.xml);
    foreach($files as $file){
        $results = parse($file);
        $status = db_handler($results);
    }
?>

This handles rating of the calls and archival of the xml_cdrs for later
incase of a DB failure or other administrative reasons we might need to go
back to them... 

Our biggest bottle neck in all this is Disk IO which there are many ways to
work around/with

K




On 4/22/11 1:46 PM, "Michael Collins" <msc at freeswitch.org> wrote:

> XML CDR is quite robust. It will post to a web server and/or store on disk.
> We've got people doing hundreds of thousands of calls per day using XML CDR.
> However, they aren't doing the parsing on their FS server, they offload that
> to a separate server, which is the wise thing to do if you have lots of
> traffic.
> 
> -MC
> 
> On Fri, Apr 22, 2011 at 11:38 AM, vibha dear <vibha_dear6 at yahoo.co.in> wrote:
>> hi all..!
>> freeswitch generated cdr in both formats xml an multiline csv format, will
>> there be any parsing overhead in XML parsing while fetching information for
>> reporting and billing? how they will effect the system performance? which
>> module is better to generate CDRs?
>> 
>> thanks in advance..!!
>> kasturi ?:)
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Sat Apr 23 00:26:55 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Apr 2011 13:26:55 -0700
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<4DB068C4.90109@xpirio.com>
	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
	<4DB1550C.1050607@xpirio.com>
	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
	<BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>
	<BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com>
Message-ID: <BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com>

You can definitely use the event socket. Heck, you can even use Python if
you want to. The dev team wrote ESL specifically for cases like these -
where you want to control FS externally. ESL beats the pants off AGI
scripts. It gives you complete control over the system.

That all being said, you can do all sorts of stuff with Lua dialplan
scripts. Check out chapter 7 of the FreeSWITCH book for some nice examples.
Lua is lightweight and easy to learn. It's a good alternative for many
cases. Check it out...

-MC

On Fri, Apr 22, 2011 at 12:39 PM, Antonio Teixeira
<eagle.antonio at gmail.com>wrote:

> Well i totally agree with the Dev team decision , it would be impossible
> for a DEV team to "get it right" in all the distros that's why i started
> this post.
>
> But we also need to take into consideration that we are talking about IVR
> Processing ,  not auto-*attendants* or simply call pass trough.
> This means heavy use of TTS / ASR , Database Logic and Scripts , GetDigits
> and the works.
>
> I use python alot , but i think the mod_python is not the best tool for
> this job i admit that , that's why im currently looking for other solutions.
> I'm thinking in using mod_socket , but that scares me ( let? say bad
> experiences with Asterisk AGI) Or Mod Curl
>
> The main problem is that some IVR are extremely complex , like
> questionnaires , etc.
>
> It would be great if we could mod_event_zmq to control the calls [?]
>
> Just to Sum it UP so far , so i can later add it to the wiki.
>
> Use Cent OS 64.
> Use tmpfs for all the databases.
>
> Thank you all for helping and Happy Eastern.
> Ant?nio Teixeira
>
>
>
>
>
> 2011/4/22 Michael Collins <msc at freeswitch.org>
>
>> FreeSWITCH runs well on many platforms. However, the devs are painfully
>> aware that bleeding edge distros have bleeding edge gcc compilers and other
>> interesting issues. That being said, CentOS 5.x is "stable" in that it's old
>> and boring, therefore it has the least amount of drama. OTOH, some of our
>> users have been having great success with Arch Linux (IRC:bougeyman) and FS,
>> even though Arch uses very recent kernels.
>>
>> Bottom line: if you know what you're doing then you can probably run FS
>> anywhere. If you don't know what you're doing then stick with CentOS 5.x or
>> Debian Lenny until you do. (I run then both with zero issues, compiling
>> latest git each day.)
>>
>> -MC
>>
>>
>> On Fri, Apr 22, 2011 at 10:03 AM, curriegrad2004 <
>> curriegrad2004 at gmail.com> wrote:
>>
>>> Freeswitch is targeted for CentOS 5.3, which in my opinion quite short
>>> sighted for the developers to do this. However with the limited size
>>> of developers and testers, I'm afraid there's not much platforms we
>>> can throughly test and actually say "okay, FS will run flawlessly on X
>>> distro"
>>>
>>> However you can always try messing with the CFLAG's mtune option and
>>> see what it produces for you...
>>>
>>> 2011/4/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
>>> > hi
>>> >
>>> > if you refer to my e-mail
>>> >
>>> > yes, we do use tmpfs on both variants but
>>> > - delays occur with concurrent calls > 80-100
>>> > - cps is limited to 5-10 on debian, with centos 30 cps and more are no
>>> problem at all
>>> >
>>> > also cpu load, stability and overall performace have been much better
>>> since using centos
>>> >
>>> > i just found out for me that debian works not as good for me as centos
>>> does.
>>> > btw. everywhere else debian is 1st choice (desktop, lamp, db etc.)
>>> >
>>> > br
>>> >
>>> >
>>> > On 2011-04-21 23:04, Jay Binks wrote:
>>> >
>>> >> I have no such problems on debian .
>>> >>
>>> >> I use debian 5 with 2.6.18 kernel which is what Is recommended
>>> >>
>>> >> Are you using tmpfs ??
>>> >>
>>> >> Jay
>>> >>
>>> >>
>>> >>
>>> >> On 22/04/2011, at 3:26 AM, Christian L?schenkohl<
>>> christian.loeschenkohl at xpirio.com>  wrote:
>>> >>
>>> >>> hi
>>> >>>
>>> >>> we did use debian too and had such performance issues (sip packet
>>> delays, low cps).
>>> >>> after using centos 64bit (as advised by the devs) all performance
>>> problems are gone.
>>> >>>
>>> >>> br
>>> >>>
>>> >>> On 2011-04-21 18:24, Antonio Teixeira wrote:
>>> >>>
>>> >>>> Hello List.
>>> >>>>
>>> >>>> I'm currently integrating an IVR in python together with freeswitch
>>> using mod_python and ESL and my life has been well until ...
>>> >>>> The flow of calls went over 80 simultaneous calls.
>>> >>>> Now freeswitch starts sending packets with huge delays ( even when
>>> establishing the call , mainly the 200 ) and firing up the IVR with tons of
>>> delay up to 20 seconds.
>>> >>>>
>>> >>>> So i searched the wiki forums and mailing list:
>>> >>>>
>>> >>>> Put freeswitch on a diet , trimmed modules.conf
>>> >>>> Played with the ulimit stuff.
>>> >>>> Played with the IVRS to reduce load to a minimum and i was able to
>>> squeeze more 5 calls of performance.
>>> >>>>
>>> >>>> The problem is :
>>> >>>>
>>> >>>> Top shows
>>> >>>> top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92,
>>> 1.76, 1.78
>>> >>>> Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0 zombie
>>> >>>> Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,
>>>  0.5%si,  0.0%st
>>> >>>> Mem:   8193336k total,  1639156k used,  6554180k free,   177208k
>>> buffers
>>> >>>> Swap: 19534904k total,        0k used, 19534904k free,  1062272k
>>> cached
>>> >>>>
>>> >>>>    PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
>>>  COMMAND
>>> >>>> 31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85
>>> freeswitch
>>> >>>>
>>> >>>> Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
>>> >>>>
>>> >>>>
>>> >>>> Some basic System Info :
>>> >>>> Debian 6.0 ( i heard the timming module is affected by Debian , but
>>> if the CPU % gets lower than 95% everything will be more stable)
>>> >>>> Python 2.5
>>> >>>>
>>> >>>> 2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
>>> >>>> 8 GB of Ram
>>> >>>>
>>> >>>> as you can see 94 % of the "Cpu Power" is sleeping :\
>>> >>>>
>>> >>>>
>>> >>>> It appears freeswitch is only capable of using let's say "one
>>> cpu"/thread ??
>>> >>>> Do you guys recommend simply starting more instances or redoing the
>>> IVR stuff.
>>> >>>>
>>> >>>>
>>> >>>> Hope you guys can help me out.
>>> >>>>
>>> >>>> Thanks
>>> >>>> Ant?nio Teixeira
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>>
>>> >>>> _______________________________________________
>>> >>>> FreeSWITCH-users mailing list
>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>>> http://www.freeswitch.org
>>> >>>
>>> >>>
>>> >>> --
>>> >>> Ing. Christian L?schenkohl
>>> >>> Technische Leitung, Forschung&  Entwicklung VoIP
>>> >>>
>>> >>> xpirio
>>> >>> Telekommunikation&  Service GmbH
>>> >>> Lakeside B04
>>> >>> 9020 Klagenfurt
>>> >>> Austria
>>> >>>
>>> >>> T  +43 5 77 11 - 1000
>>> >>> F  +43 5 77 11 - 1002
>>> >>> E  christian.loeschenkohl at xpirio.com
>>> >>>
>>> >>> _______________________________________________
>>> >>> FreeSWITCH-users mailing list
>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >>> http://www.freeswitch.org
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > --
>>> > Ing. Christian L?schenkohl
>>> > Technische Leitung, Forschung & Entwicklung VoIP
>>> >
>>> > xpirio
>>> > Telekommunikation & Service GmbH
>>> > Lakeside B04
>>> > 9020 Klagenfurt
>>> > Austria
>>> >
>>> > T  +43 5 77 11 - 1000
>>> > F  +43 5 77 11 - 1002
>>> > E  christian.loeschenkohl at xpirio.com
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 grsingh750 at gmail.com  Sat Apr 23 01:09:56 2011
From: grsingh750 at gmail.com (guru singh)
Date: Sat, 23 Apr 2011 02:39:56 +0530
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
In-Reply-To: <4DAECCA8.1050203@gmx.net>
References: <4DAECCA8.1050203@gmx.net>
Message-ID: <BANLkTikH9QMuhvKXqfBu78fOzqxoME3X8A@mail.gmail.com>

Hi Peter,

I've been using mod_callcenter for a while and must say it works really well.
I just tried the uuid-standby strategy and basically it's exactly what
you say the asterisk thing does.
See the dialplan example.
http://wiki.freeswitch.org/wiki/Mod_callcenter#uuid-standby
Agent is dialing 4099 and listening to MOH. When a call arrives, it's
bridged directly to the agent.

Regards
guru

On Wed, Apr 20, 2011 at 5:38 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
> Hello,
>
> I am trying to use the mod_callcenter functionality. This works nicely
> so far so thank you to everybody involved for programming this nice module!
> But I am stuck somehow with uuid-standby. Can anybody explain how
> uuid-standby works?
>
> Another question: In the Asterisk based callcenter solution named
> "Vicidial", an agent can be held permanently in a conference, waiting
> for calls who are bridged to his uuid in the conference. Can this be
> haviour be done with mod_callcenter?
>
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Sat Apr 23 02:09:38 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 22 Apr 2011 17:09:38 -0500
Subject: [Freeswitch-users] ZRTP Library
In-Reply-To: <BANLkTimggX_sBqxWnZENNfPK+hQ+Rj0afQ@mail.gmail.com>
References: <BANLkTimBuiOKCZQ9AQFFtKJg23C0CGhEmQ@mail.gmail.com>
	<BANLkTimggX_sBqxWnZENNfPK+hQ+Rj0afQ@mail.gmail.com>
Message-ID: <01DD1957-7DA6-48D8-BF63-7F133F5417CC@freeswitch.org>

And nobody can provide it without violating the crypto Export laws of the US.

/b

On Apr 22, 2011, at 1:44 PM, Michael Collins wrote:

> I don't know if it's okay to distribute. I've pinged PRZ on this issue and
> have not heard back from him. Sorry...
> 
> -MC



From sam.oredoyin at gmail.com  Sat Apr 23 01:15:31 2011
From: sam.oredoyin at gmail.com (Sam Oredoyin)
Date: Fri, 22 Apr 2011 22:15:31 +0100
Subject: [Freeswitch-users] Reading Channel Variables on B-Leg - To display
	Callee name
In-Reply-To: <BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>	<4DB068C4.90109@xpirio.com>	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>	<4DB1550C.1050607@xpirio.com>	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>	<BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>	<BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com>
	<BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com>
Message-ID: <4db1f0cf.4536e30a.6c38.ffffa8d2@mx.google.com>

Hi All,

 

I need to read SIP header details about the B-Leg of a call, I am looking to read the Remote-Party-ID so I can display the Callee Name on the caller?s phone.

 

I know the callee name can be set to static values in the dialplan buy I need to use the actual remote party caller name.

 

Regards

Sam

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From hasnain110 at gmail.com  Sat Apr 23 02:12:00 2011
From: hasnain110 at gmail.com (Hasnain)
Date: Fri, 22 Apr 2011 22:12:00 +0000
Subject: [Freeswitch-users] Reading Channel Variables on B-Leg - To
	displayCallee name
In-Reply-To: <4db1f0cf.4536e30a.6c38.ffffa8d2@mx.google.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>	<4DB068C4.90109@xpirio.com>	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>	<4DB1550C.1050607@xpirio.com>	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>	<BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>	<BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com><BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com><4db1f0cf.4536e30a.6c38.ffffa8d2@mx.google.com>
Message-ID: <1064150385-1303510332-cardhu_decombobulator_blackberry.rim.net-1810404828-@b14.c15.bise7.blackberry>

Use wireshark to trace the call and then read header
Sent from my BlackBerry? wireless handheld

-----Original Message-----
From: "Sam Oredoyin" <sam.oredoyin at gmail.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Fri, 22 Apr 2011 22:15:31 
To: 'FreeSWITCH Users Help'<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Reading Channel Variables on B-Leg - To display
	Callee name

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


From brian at freeswitch.org  Sat Apr 23 02:19:26 2011
From: brian at freeswitch.org (Brian West)
Date: Fri, 22 Apr 2011 17:19:26 -0500
Subject: [Freeswitch-users] Reading Channel Variables on B-Leg - To
	display Callee name
In-Reply-To: <4db1f0cf.4536e30a.6c38.ffffa8d2@mx.google.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>	<4DB068C4.90109@xpirio.com>	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>	<4DB1550C.1050607@xpirio.com>	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>	<BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>	<BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com>
	<BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com>
	<4db1f0cf.4536e30a.6c38.ffffa8d2@mx.google.com>
Message-ID: <FE4AE45B-52EB-45B1-AF0F-0BB2B1E450C2@freeswitch.org>

Please don't hijack threads.  Compose your own new messages to the list and don't click reply and change the subject.

/b

On Apr 22, 2011, at 4:15 PM, Sam Oredoyin wrote:

> Hi All,
> 
> 
> 
> I need to read SIP header details about the B-Leg of a call, I am looking to read the Remote-Party-ID so I can display the Callee Name on the caller?s phone.
> 
> 
> 
> I know the callee name can be set to static values in the dialplan buy I need to use the actual remote party caller name.
> 
> 
> 
> 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



From sam.oredoyin at gmail.com  Sat Apr 23 02:22:41 2011
From: sam.oredoyin at gmail.com (Sam Oredoyin)
Date: Fri, 22 Apr 2011 23:22:41 +0100
Subject: [Freeswitch-users] Reading Channel Variables on B-Leg -
	To	displayCallee name
In-Reply-To: <1064150385-1303510332-cardhu_decombobulator_blackberry.rim.net-1810404828-@b14.c15.bise7.blackberry>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>	<4DB068C4.90109@xpirio.com>	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>	<4DB1550C.1050607@xpirio.com>	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>	<BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>	<BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com><BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com><4db1f0cf.4536e30a.6c38.ffffa8d2@mx.google.com>
	<1064150385-1303510332-cardhu_decombobulator_blackberry.rim.net-1810404828-@b14.c15.bise7.blackberry>
Message-ID: <4db2000a.4b1be30a.69f0.ffffa9b8@mx.google.com>

Thanks Hasnain,

I need to read it in the dialplan, so dialplan can insert the B-Leg
Remote-Party-ID into the caller display.






-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hasnain
Sent: 22 April 2011 23:12
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Reading Channel Variables on B-Leg - To
displayCallee name

Use wireshark to trace the call and then read header
Sent from my BlackBerryR wireless handheld

-----Original Message-----
From: "Sam Oredoyin" <sam.oredoyin at gmail.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Fri, 22 Apr 2011 22:15:31 
To: 'FreeSWITCH Users Help'<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Reading Channel Variables on B-Leg - To display
	Callee name

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

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.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  Sat Apr 23 03:31:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 22 Apr 2011 16:31:11 -0700
Subject: [Freeswitch-users] Reading Channel Variables on B-Leg - To
 displayCallee name
In-Reply-To: <4db2000a.4b1be30a.69f0.ffffa9b8@mx.google.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<4DB068C4.90109@xpirio.com>
	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
	<4DB1550C.1050607@xpirio.com>
	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
	<BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>
	<BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com>
	<BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com>
	<4db1f0cf.4536e30a.6c38.ffffa8d2@mx.google.com>
	<1064150385-1303510332-cardhu_decombobulator_blackberry.rim.net-1810404828-@b14.c15.bise7.blackberry>
	<4db2000a.4b1be30a.69f0.ffffa9b8@mx.google.com>
Message-ID: <BANLkTinw2Xy2rQm-B+oHheBcyzKxjHkWUg@mail.gmail.com>

On Fri, Apr 22, 2011 at 3:22 PM, Sam Oredoyin <sam.oredoyin at gmail.com>wrote:

> Thanks Hasnain,
>
> I need to read it in the dialplan, so dialplan can insert the B-Leg
> Remote-Party-ID into the caller display.
>

Send the inbound call to the "info" dialplan app and look at the console
output. There are many sip_xxx variables you can choose from. Find the one
you want and use it.
-MC
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From frankie.k.yiu at gmail.com  Sat Apr 23 04:21:20 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Fri, 22 Apr 2011 17:21:20 -0700
Subject: [Freeswitch-users] setup an enviroment for implementating /
	testing IVR
Message-ID: <BANLkTin3zrNYJtrhdQXaA3r80b8qCvdo5A@mail.gmail.com>

Thanks for the input.  But dialing 5000 would be testing the demo IVR, but
my question is that how do I make an incoming call (Do I have to dial a
extension number?) so that I can hear my own custom audio?

Sorry for the confusing/dumb question.

Thank you.

Frankie




>
> ---------- Forwarded message ----------
> From: Philippe Le Toquin <philippe at ppmt.org>
> To: freeswitch-users at lists.freeswitch.org
> Date: Thu, 21 Apr 2011 21:27:10 -0400
> Subject: Re: [Freeswitch-users] setup an enviroment for implementating /
> testing IVR feature
>  Hello,
>
> This link is a good start to get to do things with your new Freeswitch
>
> http://wiki.freeswitch.org/wiki/Getting_Started_Guide
>
> to test IVR just dial 5000 from your softphone to access a demo IVR
>
> One thing to change is the default password of the extension to make it a
> bit harder to crack :)
> Look in the vars.xml file
>
> /Philippe
>
> On 11-04-21 08:45 PM, Frankie Yiu wrote:
>
> Hi there,
>
> I would like to start implementating an IVR feature, but the first thing I
> need to do is to able to use a softphone to dial into my freeSWITCH so that
> an audio would start playing.
> Right now I have the original configuration of freeSWITCH which I am not
> sure how to dial into the system using the softphone.  For example, what
> extension I have to call; or if I need to change anything in dialplan,
> or configuration, etc.
>
> I have C# code that let me make outgoing call to my softphone, play audio
> or collect DTMF.  But I am not sure what to do for incoming call using the
> original configuration from freeSWITCH.
>
> Thanks in advance.
>
> Frankie
>
>
> _______________________________________________
> 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
>
>
>
> ---------- Forwarded message ----------
> From: Boris Kovalenko <boris at tagnet.ru>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 22 Apr 2011 08:01:36 +0600
> Subject: [Freeswitch-users] variable direction
> Hello!
>
>    I found that with gateways I may use <variable .... direction="..."/>.
> Is this possible with regular (directory) users too? If not, may somebody
> explain why?
>
> --
> ? ?????????,
>  ????? ?????????
>  ??? "??????"
>  ???. +7 (3435) 230001
>  ???? +7 (3435) 230005
>
>
>
>
>
> ---------- Forwarded message ----------
> From: David Ponzone <david.ponzone at ipeva.fr>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 22 Apr 2011 04:04:21 +0200
> Subject: [Freeswitch-users] Ringback while NOT YET bridging to leg B
> Hi all,
>
> I am trying to achieve something weird, I admit.
> I would need to receive a call from A, then wait X seconds before bridging
> to B, but I'd like to send back a ringback to A while sleeping X seconds.
> The point is to allow customers to have a nice ringback, which the
> beginning of HAS to be heard by callers (like an announce in a call-center).
>
> A (ugly) way to do that would be to have 2 sound files: one to be played
> when answering with playback, and then another one for the ringback. I would
> then not use sleep at all, but the bridge would only happen when the first
> file is over, achieving quite the same result.
> The drawback of this is that I need to ask customers for 2 sound files, and
> that can be a pain with most of them.
>
> I tried all combinations I could think of with ringback, instant_ringback,
> with no luck.
>
> Any ideas ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Yehavi Bourvine <yehavi.bourvine at gmail.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Fri, 22 Apr 2011 07:11:34 +0300
> Subject: Re: [Freeswitch-users] Changing From header in voicemail to e-mail
> notifications
> Hello John,
>
>   You have to set the following in vars,xml:
>
>   <X-PRE-PROCESS cmd="set" data="voicemail_account=*username*"/>
>   <X-PRE-PROCESS cmd="set" data="voicemail_domain=*your-domain.com*"/>
>
>                  Regards, __Yehavi:
>  2011/4/21 John Platts <john_platts at hotmail.com>
>
>>  We see FreeSWITCH mod_voicemail in the From header of the voicemail to
>> e-mail notifications. We also want to change the E-mail address in the From
>> header to come from a specific domain. How do we go about changing the From:
>> header in the voicemail to e-mail notifications? We do not want FreeSWITCH
>> mod_voicemail to appear in the From header of the e-mail messages.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> ---------- Forwarded message ----------
> From: Nasir Iqbal <nasir at ictinnovations.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Fri, 22 Apr 2011 08:44:32 +0500
> Subject: [Freeswitch-users] ZRTP Library
> Hi,
>
> As zfone download server is offline, is there anyone who can provide me
> libzrtp SDK source?
>
>
> Nasir Iqbal
>
> ICT Innovations
> http://www.ictinnovations.com/
>
>
>
> ---------- Forwarded message ----------
> From: Michael Collins <msc at freeswitch.org>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Thu, 21 Apr 2011 22:49:26 -0700
> Subject: Re: [Freeswitch-users] variable direction
> The direction variable is set at the time of the call based upon whether or
> not the user is the caller (outbound) or callee (inbound).
>
> -MC
>
> On Thu, Apr 21, 2011 at 7:01 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
>
>> Hello!
>>
>>     I found that with gateways I may use <variable ....
>> direction="..."/>. Is this possible with regular (directory) users too?
>> If not, may somebody explain why?
>>
>> --
>> ? ?????????,
>>   ????? ?????????
>>   ??? "??????"
>>   ???. +7 (3435) 230001
>>   ???? +7 (3435) 230005
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 pieter_eduard at biznetnetworks.com  Sat Apr 23 04:41:36 2011
From: pieter_eduard at biznetnetworks.com (Pieter Eduard)
Date: Sat, 23 Apr 2011 00:41:36 +0000
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com><4DB068C4.90109@xpirio.com><6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com><4DB1550C.1050607@xpirio.com><BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com><BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com><BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com><BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com>
Message-ID: <1867782920-1303519297-cardhu_decombobulator_blackberry.rim.net-1866367412-@b27.c4.bise3.blackberry>

M? 
Pieter Eduard

-----Original Message-----
From: Michael Collins <msc at freeswitch.org>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Fri, 22 Apr 2011 13:26:55 
To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Tuning Up Freeswitch

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


From david.ponzone at ipeva.fr  Sat Apr 23 04:58:12 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sat, 23 Apr 2011 02:58:12 +0200
Subject: [Freeswitch-users] Ringback while NOT YET bridging to leg B
In-Reply-To: <88005444-0B2E-4BFD-B6BF-5952BE09BBBE@ipeva.fr>
References: <88005444-0B2E-4BFD-B6BF-5952BE09BBBE@ipeva.fr>
Message-ID: <165C97F1-5958-4171-B750-939ACE001EAE@ipeva.fr>

Well, I am not really surprised noone replied yet :)
I am not even sure anyone understood what I asked for!

Anyway, I did some more digging, and I found a solution, a ugly one I would say.
Again, the idea is to receive a call, wait for X seconds while FS starts sending the ringback to the caller, and then bridge the call while continuing sending the ringback (without playing it again from the start). The idea is to force the caller to hear a part of the ringback by no bridging the call (so not taking any chance for it to be answered), and then, we bridge the call as usual.
The same thing could be achieved by using 2 sound files: playback with the first one, and ringback with the second one.

Here it is: 
  
  <extension name="123456789">
      <condition field="destination_number" expression="^(123456789)$">
        <action application="set" data="instant_ringback=true"/>
        <action application="set" data="ringback=$${sound_prefix}/ringback.wav"/>
        <action application="bridge" data="loopback/wait$1"/>
      </condition>
    </extension>

    <extension name="wait123456789">
      <condition field="destination_number" expression="^wait(123456789)$">
        <action application="sleep" data="5000"/>
        <action application="bridge" data="sofia/gateway/mygw/$1"/>
      </condition>
    </extension>

I find it ugly, especially when the final bridge is not successfull. I then have duplicate CDRs for the bridge to wait.

I think I will prefer to deal with the 2 sound files, but I wanted to share that in case it helps someone.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 22/04/2011 ? 04:04, David Ponzone a ?crit :

> Hi all,
> 
> I am trying to achieve something weird, I admit.
> I would need to receive a call from A, then wait X seconds before bridging to B, but I'd like to send back a ringback to A while sleeping X seconds.
> The point is to allow customers to have a nice ringback, which the beginning of HAS to be heard by callers (like an announce in a call-center).
> 
> A (ugly) way to do that would be to have 2 sound files: one to be played when answering with playback, and then another one for the ringback. I would then not use sleep at all, but the bridge would only happen when the first file is over, achieving quite the same result.
> The drawback of this is that I need to ask customers for 2 sound files, and that can be a pain with most of them.
> 
> I tried all combinations I could think of with ringback, instant_ringback, with no luck.
> 
> Any ideas ?
> 
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
> 
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
> 
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
> 
> 
> 
> 

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From anthony.minessale at gmail.com  Sat Apr 23 05:41:36 2011
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 22 Apr 2011 20:41:36 -0500
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<4DB068C4.90109@xpirio.com>
	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
	<4DB1550C.1050607@xpirio.com>
	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
Message-ID: <BANLkTik6Wx66Mkzt-bfe0sbOQsMVLqT41w@mail.gmail.com>

I would hardly call directing users to a stable distro "quite short sighted"
considering we run on mac windows bsd solaris and most any linux, I fail to
see any short sightedness.

In fact we have a guy who maintains ubuntu packages and another on arch.  In
fact I just got a new bleeding arch box and got very good results.  The
latest fs works well on both old and new distros.  I would venture to say
that perhaps you are the short sighted individual, reading too specifically
into one aspect of our development.   I will continue to recommend centos to
new users until they gain experience and have the skill to use the distro or
os they want.
On Apr 22, 2011 12:04 PM, "curriegrad2004" <curriegrad2004 at gmail.com> wrote:
> Freeswitch is targeted for CentOS 5.3, which in my opinion quite short
> sighted for the developers to do this. However with the limited size
> of developers and testers, I'm afraid there's not much platforms we
> can throughly test and actually say "okay, FS will run flawlessly on X
> distro"
>
> However you can always try messing with the CFLAG's mtune option and
> see what it produces for you...
>
> 2011/4/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
>> hi
>>
>> if you refer to my e-mail
>>
>> yes, we do use tmpfs on both variants but
>> - delays occur with concurrent calls > 80-100
>> - cps is limited to 5-10 on debian, with centos 30 cps and more are no
problem at all
>>
>> also cpu load, stability and overall performace have been much better
since using centos
>>
>> i just found out for me that debian works not as good for me as centos
does.
>> btw. everywhere else debian is 1st choice (desktop, lamp, db etc.)
>>
>> br
>>
>>
>> On 2011-04-21 23:04, Jay Binks wrote:
>>
>>> I have no such problems on debian .
>>>
>>> I use debian 5 with 2.6.18 kernel which is what Is recommended
>>>
>>> Are you using tmpfs ??
>>>
>>> Jay
>>>
>>>
>>>
>>> On 22/04/2011, at 3:26 AM, Christian L?schenkohl<
christian.loeschenkohl at xpirio.com>  wrote:
>>>
>>>> hi
>>>>
>>>> we did use debian too and had such performance issues (sip packet
delays, low cps).
>>>> after using centos 64bit (as advised by the devs) all performance
problems are gone.
>>>>
>>>> br
>>>>
>>>> On 2011-04-21 18:24, Antonio Teixeira wrote:
>>>>
>>>>> Hello List.
>>>>>
>>>>> I'm currently integrating an IVR in python together with freeswitch
using mod_python and ESL and my life has been well until ...
>>>>> The flow of calls went over 80 simultaneous calls.
>>>>> Now freeswitch starts sending packets with huge delays ( even when
establishing the call , mainly the 200 ) and firing up the IVR with tons of
delay up to 20 seconds.
>>>>>
>>>>> So i searched the wiki forums and mailing list:
>>>>>
>>>>> Put freeswitch on a diet , trimmed modules.conf
>>>>> Played with the ulimit stuff.
>>>>> Played with the IVRS to reduce load to a minimum and i was able to
squeeze more 5 calls of performance.
>>>>>
>>>>> The problem is :
>>>>>
>>>>> Top shows
>>>>> top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92, 1.76,
1.78
>>>>> Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0 zombie
>>>>> Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,  0.5%si,
 0.0%st
>>>>> Mem:   8193336k total,  1639156k used,  6554180k free,   177208k
buffers
>>>>> Swap: 19534904k total,        0k used, 19534904k free,  1062272k
cached
>>>>>
>>>>>    PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
>>>>> 31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85
freeswitch
>>>>>
>>>>> Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
>>>>>
>>>>>
>>>>> Some basic System Info :
>>>>> Debian 6.0 ( i heard the timming module is affected by Debian , but if
the CPU % gets lower than 95% everything will be more stable)
>>>>> Python 2.5
>>>>>
>>>>> 2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
>>>>> 8 GB of Ram
>>>>>
>>>>> as you can see 94 % of the "Cpu Power" is sleeping :\
>>>>>
>>>>>
>>>>> It appears freeswitch is only capable of using let's say "one
cpu"/thread ??
>>>>> Do you guys recommend simply starting more instances or redoing the
IVR stuff.
>>>>>
>>>>>
>>>>> Hope you guys can help me out.
>>>>>
>>>>> Thanks
>>>>> Ant?nio Teixeira
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> --
>>>> Ing. Christian L?schenkohl
>>>> Technische Leitung, Forschung&  Entwicklung VoIP
>>>>
>>>> xpirio
>>>> Telekommunikation&  Service GmbH
>>>> Lakeside B04
>>>> 9020 Klagenfurt
>>>> Austria
>>>>
>>>> T  +43 5 77 11 - 1000
>>>> F  +43 5 77 11 - 1002
>>>> E  christian.loeschenkohl at xpirio.com
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> --
>> Ing. Christian L?schenkohl
>> Technische Leitung, Forschung & Entwicklung VoIP
>>
>> xpirio
>> Telekommunikation & Service GmbH
>> Lakeside B04
>> 9020 Klagenfurt
>> Austria
>>
>> T  +43 5 77 11 - 1000
>> F  +43 5 77 11 - 1002
>> E  christian.loeschenkohl at xpirio.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From boris at tagnet.ru  Sat Apr 23 07:07:16 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 23 Apr 2011 09:07:16 +0600
Subject: [Freeswitch-users] Need a little bit clarification
	about	gateways
In-Reply-To: <9BD70C31-770C-428F-8BAE-D56040C5B257@freeswitch.org>
References: <4DB1B196.8030802@tagnet.ru>
	<9BD70C31-770C-428F-8BAE-D56040C5B257@freeswitch.org>
Message-ID: <4DB24264.5010606@tagnet.ru>

Hello!

Thank You!. And a little question about ACL if I use form:
<param name="apply-inbound-acl" value="domains,gateways"/> the user will 
be authenticated if it is in domains OR gateways ACL.
And if i use:
<param name="apply-inbound-acl" value="domains"/>
<param name="apply-inbound-acl" value="gateway"/> the user will be 
authenticated if it is in domains AND gateways ACL. Right?

> you should only add the ACL if you have auth-calls=true.
>
> /b
>
> On Apr 22, 2011, at 11:49 AM, Boris Kovalenko wrote:
>
>> Hello!
>>
>>      Am I right, if I created gateway into sofia profile and set
>> register=false I also should permit its IP in the ACL to receive a calls
>> from it? Or there is some param to gateway for automatic acl creation?
>> Like cidr for a user?
>>
>> -- 
>> With respect,
>> 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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



From fieldpeak at gmail.com  Sat Apr 23 07:50:38 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sat, 23 Apr 2011 11:50:38 +0800
Subject: [Freeswitch-users] config ramdisk on CentOS 5.5
Message-ID: <BANLkTinTkAR64kGsZbah5=Z3Hyxf8MwQiw@mail.gmail.com>

i'm trying tuning the FS to max performance on centos 5.5, and referred to
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#FreeSWITCH.27s_core.db_I.2FO_bottleneck

#1,  i configure the DB of FS to ramdisk ,

when i run "tmpfs /opt/freeswitch/db tmpfs defaults 0 0", it output: "-bash:
tmpfs: command not found"

#2,  i run "ethtool -g eth0", the output is below, what value i should
config for RX and TX for max performance...
Ring parameters for eth0:
Pre-set maximums:
RX:             4096
RX Mini:        0
RX Jumbo:       0
TX:             4096
Current hardware settings:
RX:             256
RX Mini:        0
RX Jumbo:       0
TX:             256

Apprecited if anyone help how to configure it... thanks!

Regards,
Charles
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From krice at freeswitch.org  Sat Apr 23 08:02:12 2011
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 22 Apr 2011 23:02:12 -0500
Subject: [Freeswitch-users] config ramdisk on CentOS 5.5
In-Reply-To: <BANLkTinTkAR64kGsZbah5=Z3Hyxf8MwQiw@mail.gmail.com>
Message-ID: <C9D7B974.14501%krice@freeswitch.org>

Tmpfs is not a program... Read that page a little closer... That?s
particular line is for your fstab...

If you want to mount it from the command line its
mount ?o tmpfs tmpfs /usr/local/freeswitch/db

K


On 4/22/11 10:50 PM, "fieldpeak" <fieldpeak at gmail.com> wrote:

> i'm trying tuning the FS to max performance on centos 5.5, and referred to
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#FreeSWI
> TCH.27s_core.db_I.2FO_bottleneck
> 
> #1,? i configure the DB of FS to ramdisk ,
> 
> when i run "tmpfs /opt/freeswitch/db tmpfs defaults 0 0", it output: "-bash:
> tmpfs: command not found"
> 
> #2,? i run "ethtool -g eth0", the output is below, what value i should config
> for RX and TX for max performance...
> Ring parameters for eth0:
> Pre-set maximums:
> RX:???????????? 4096
> RX Mini:??????? 0
> RX Jumbo:?????? 0
> TX:???????????? 4096
> Current hardware settings:
> RX:???????????? 256
> RX Mini:??????? 0
> RX Jumbo:?????? 0
> TX:???????????? 256
> 
> Apprecited if anyone help how to configure it... 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

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From michel at arneill-py.sacramento.ca.us  Sat Apr 23 08:47:11 2011
From: michel at arneill-py.sacramento.ca.us (Michel Py)
Date: Fri, 22 Apr 2011 21:47:11 -0700
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
	with productivity license.
In-Reply-To: <BANLkTi=XmNDV51=xoNsViHUYaUZeyMpz_A@mail.gmail.com>
References: <471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local>
	<BANLkTi=XmNDV51=xoNsViHUYaUZeyMpz_A@mail.gmail.com>
Message-ID: <471D76419F9EF642962323D13DF1DF69011E58@newserver.arneill-py.local>

Hi Christian and Yehavi,

Thanks for the time looking into this!


> Christian Benke wrote:
> Just a wild guess - are you actually loading the file with the
> attendant-settings? Usually phone1.cfg or $MAC-phone1.cfg -
> there should be an entry in the basic $MAC.cfg for this file.

Thanks for suggesting this; it did not solve my problem, but it made me
change / clean my test setting for the better. Here is my new test
procedure:

- On the phone, I regularly do erase local config, factory settings, and
format file system. Just to be sure.

- I completely cleaned the /tftpboot directory, except for 2 files:
0004f2xxxxxx-phone.cfg
0004f2xxxxxx-license.cfg

- Extracted the files from the Polycom zip into /tftpboot, tried that
both with 3.1.7 and 3.1.0 rev C. I can positively confirm that even
after a format file system, the phone will successfully load both the
3.x SIP version of the moment AND the 0004f2xxxxxx-phone.cfg.

- I don't bother to have a 0004f2xxxxxx.cfg file anymore. I don't need
it as long as I keep the 0000000000000.cfg file into /tftproot. It finds
the phone1_31x.cfg and the sip_31x.cfg files just fine.

- About the reading of the 0004f2xxxxxx-phone.cfg: I am absolutely
positive that it reads it for 2 reasons:

1. I can see it in tcpdump port 69 -v
2. Changes made to 0004f2xxxxxx-phone.cfg will show up on the phone
after reboot.

I pasted the current version of the file below.
It reads the file, parses it, and some parts work and some don't.

What does work:
- The reg.1.stuff. The line shows up, registers, can make calls, etc.
- The SNTP timezone stuff. Changes to the gmtOffset value will show up
at next phone reboot.
- The presence feature. Changing it to "1" will indeed enable the
feature, and if I add buddy entries in the directory (from the phone
itself) they actually work, with their known limitations. If enabled,
the "MyStat" and "Buddies" softkey do appear on the main screen, and do
go out if deactivated.
- The corporate directory feature. I have not actually tested it, but
the feature does show in the menus when activated and not when
deactivated.

What does not work:
- The attendant thing (duh).
- Trying to disable the "MyStat" and "Buddies" softkeys when presence is
enabled.

In short:
- The phone DOES read the 0004f2xxxxxx-phone.cfg file.
- When it parses it, some functions work and some don't.

- In my tests, it does not matter in which file you put things. Let's
say the TZ commands or the feature commands: they work regardless if I
put them in 0004f2xxxxxx-phone.cfg or in {phone1_31x.cfg|sip_31x.cfg}. I
know, I'm not supposed to modify these but I did try, and it does work
as expected. Same applies to the attendant commands: regardless of which
file you put them in, they don't work.

- In case it was not clear: I can make the "buddies" thing work any way
I want, and I can't make the "attendant" way work at all, regardless of
what I do. I have tried about 25 different variants, no go.

What bugs the ##^%@# out of me is this: Whether it actually works or
does anything at all, just the following 2 lines should display
something on the phone's screen: 
  attendant.resourceList.1.address="8000"
  attendant.resourceList.1.label="Name1"

So the LED may not change, the button may not work, but it should show.
It does not. So my problem is not that the phone does or does not
register with the server, which may or may not work... It's that the
phone fails to parse the config file for these items.

Feedback more than welcome!

Thanks,
Michel.


<?xml version="1.0" encoding="utf-8" standalone="yes"?>
<userinfo>
 <reg
  reg.1.displayName="Test 2499"
  reg.1.address="2499"
  reg.1.type="private"
  reg.1.auth.userId="2499"
  reg.1.auth.password="xxxxxxx"
  reg.1.server.1.register="1"
  reg.1.server.1.address="192.168.17.1"
  reg.1.server.1.port="5060"
 />
</userinfo>

<attendant
  attendant.reg="1"
  attendant.ringType="1"
  attendant.behaviors.display.spontaneousCallAppearances.normal="1"
  attendant.behaviors.display.spontaneousCallAppearances.automata="1"
  attendant.behaviors.display.remoteCallerID.normal="1"
  attendant.behaviors.display.remoteCallerID.automata="1"
  attendant.resourceList.1.address="8000"
  attendant.resourceList.1.label="Name1"
  attendant.resourceList.2.address="8001"
  attendant.resourceList.2.label="Name2"
/>

<softkey softkey.feature.mystatus="0" softkey.feature.buddies="0" />
<SNTP tcpIpApp.sntp.gmtOffset="-25200" />
<feature
 feature.1.name="presence" feature.1.enabled="0"
 feature feature.19.name="corporate-directory" feature.19.enabled="0"
/>



From yehavi.bourvine at gmail.com  Sat Apr 23 09:20:17 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Sat, 23 Apr 2011 08:20:17 +0300
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
 with productivity license.
In-Reply-To: <471D76419F9EF642962323D13DF1DF69011E58@newserver.arneill-py.local>
References: <471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local>
	<BANLkTi=XmNDV51=xoNsViHUYaUZeyMpz_A@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E58@newserver.arneill-py.local>
Message-ID: <BANLkTi=TJCn2Ut+1ED5c24EYKRqH=nLKQw@mail.gmail.com>

Hello Michel,

 Please  start TCPDUMP of your phone for port 5060 only, and reboot it.
After it is up do a few operations on the watched phones (i.e. extensions
8000, 8001) like oroginating and receiving calls, and send me the resultant
dump.

             Regards, __Yehavi:

2011/4/23 Michel Py <michel at arneill-py.sacramento.ca.us>

> Hi Christian and Yehavi,
>
> Thanks for the time looking into this!
>
>
> > Christian Benke wrote:
> > Just a wild guess - are you actually loading the file with the
> > attendant-settings? Usually phone1.cfg or $MAC-phone1.cfg -
> > there should be an entry in the basic $MAC.cfg for this file.
>
> Thanks for suggesting this; it did not solve my problem, but it made me
> change / clean my test setting for the better. Here is my new test
> procedure:
>
> - On the phone, I regularly do erase local config, factory settings, and
> format file system. Just to be sure.
>
> - I completely cleaned the /tftpboot directory, except for 2 files:
> 0004f2xxxxxx-phone.cfg
> 0004f2xxxxxx-license.cfg
>
> - Extracted the files from the Polycom zip into /tftpboot, tried that
> both with 3.1.7 and 3.1.0 rev C. I can positively confirm that even
> after a format file system, the phone will successfully load both the
> 3.x SIP version of the moment AND the 0004f2xxxxxx-phone.cfg.
>
> - I don't bother to have a 0004f2xxxxxx.cfg file anymore. I don't need
> it as long as I keep the 0000000000000.cfg file into /tftproot. It finds
> the phone1_31x.cfg and the sip_31x.cfg files just fine.
>
> - About the reading of the 0004f2xxxxxx-phone.cfg: I am absolutely
> positive that it reads it for 2 reasons:
>
> 1. I can see it in tcpdump port 69 -v
> 2. Changes made to 0004f2xxxxxx-phone.cfg will show up on the phone
> after reboot.
>
> I pasted the current version of the file below.
> It reads the file, parses it, and some parts work and some don't.
>
> What does work:
> - The reg.1.stuff. The line shows up, registers, can make calls, etc.
> - The SNTP timezone stuff. Changes to the gmtOffset value will show up
> at next phone reboot.
> - The presence feature. Changing it to "1" will indeed enable the
> feature, and if I add buddy entries in the directory (from the phone
> itself) they actually work, with their known limitations. If enabled,
> the "MyStat" and "Buddies" softkey do appear on the main screen, and do
> go out if deactivated.
> - The corporate directory feature. I have not actually tested it, but
> the feature does show in the menus when activated and not when
> deactivated.
>
> What does not work:
> - The attendant thing (duh).
> - Trying to disable the "MyStat" and "Buddies" softkeys when presence is
> enabled.
>
> In short:
> - The phone DOES read the 0004f2xxxxxx-phone.cfg file.
> - When it parses it, some functions work and some don't.
>
> - In my tests, it does not matter in which file you put things. Let's
> say the TZ commands or the feature commands: they work regardless if I
> put them in 0004f2xxxxxx-phone.cfg or in {phone1_31x.cfg|sip_31x.cfg}. I
> know, I'm not supposed to modify these but I did try, and it does work
> as expected. Same applies to the attendant commands: regardless of which
> file you put them in, they don't work.
>
> - In case it was not clear: I can make the "buddies" thing work any way
> I want, and I can't make the "attendant" way work at all, regardless of
> what I do. I have tried about 25 different variants, no go.
>
> What bugs the ##^%@# out of me is this: Whether it actually works or
> does anything at all, just the following 2 lines should display
> something on the phone's screen:
>  attendant.resourceList.1.address="8000"
>  attendant.resourceList.1.label="Name1"
>
> So the LED may not change, the button may not work, but it should show.
> It does not. So my problem is not that the phone does or does not
> register with the server, which may or may not work... It's that the
> phone fails to parse the config file for these items.
>
> Feedback more than welcome!
>
> Thanks,
> Michel.
>
>
> <?xml version="1.0" encoding="utf-8" standalone="yes"?>
> <userinfo>
>  <reg
>  reg.1.displayName="Test 2499"
>  reg.1.address="2499"
>  reg.1.type="private"
>  reg.1.auth.userId="2499"
>  reg.1.auth.password="xxxxxxx"
>  reg.1.server.1.register="1"
>  reg.1.server.1.address="192.168.17.1"
>  reg.1.server.1.port="5060"
>  />
> </userinfo>
>
> <attendant
>  attendant.reg="1"
>  attendant.ringType="1"
>  attendant.behaviors.display.spontaneousCallAppearances.normal="1"
>  attendant.behaviors.display.spontaneousCallAppearances.automata="1"
>  attendant.behaviors.display.remoteCallerID.normal="1"
>  attendant.behaviors.display.remoteCallerID.automata="1"
>  attendant.resourceList.1.address="8000"
>  attendant.resourceList.1.label="Name1"
>  attendant.resourceList.2.address="8001"
>  attendant.resourceList.2.label="Name2"
> />
>
> <softkey softkey.feature.mystatus="0" softkey.feature.buddies="0" />
> <SNTP tcpIpApp.sntp.gmtOffset="-25200" />
> <feature
>  feature.1.name="presence" feature.1.enabled="0"
>  feature feature.19.name="corporate-directory" feature.19.enabled="0"
> />
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From eagle.antonio at gmail.com  Sat Apr 23 11:51:27 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Sat, 23 Apr 2011 08:51:27 +0100
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<4DB068C4.90109@xpirio.com>
	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
	<4DB1550C.1050607@xpirio.com>
	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
	<BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>
	<BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com>
	<BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com>
Message-ID: <BANLkTin6c0q_66bNk-W+1WjVKkf+0w27FA@mail.gmail.com>

Hello Michael & The rest.

What i'm trying to find is the solution with the highest performance for IVR
Applications. And squeezing the last bit of performance out of freeswitch is
a needed path specially on a industry that does second by second billing :P
i'm going to do some performance testing with Lua , ESL And Mod Curl and of
course python and measure some performance data and will post it later.

A/T


2011/4/22 Michael Collins <msc at freeswitch.org>

> You can definitely use the event socket. Heck, you can even use Python if
> you want to. The dev team wrote ESL specifically for cases like these -
> where you want to control FS externally. ESL beats the pants off AGI
> scripts. It gives you complete control over the system.
>
> That all being said, you can do all sorts of stuff with Lua dialplan
> scripts. Check out chapter 7 of the FreeSWITCH book for some nice examples.
> Lua is lightweight and easy to learn. It's a good alternative for many
> cases. Check it out...
>
> -MC
>
>
> On Fri, Apr 22, 2011 at 12:39 PM, Antonio Teixeira <
> eagle.antonio at gmail.com> wrote:
>
>> Well i totally agree with the Dev team decision , it would be impossible
>> for a DEV team to "get it right" in all the distros that's why i started
>> this post.
>>
>> But we also need to take into consideration that we are talking about IVR
>> Processing ,  not auto-*attendants* or simply call pass trough.
>> This means heavy use of TTS / ASR , Database Logic and Scripts , GetDigits
>> and the works.
>>
>> I use python alot , but i think the mod_python is not the best tool for
>> this job i admit that , that's why im currently looking for other solutions.
>> I'm thinking in using mod_socket , but that scares me ( let? say bad
>> experiences with Asterisk AGI) Or Mod Curl
>>
>> The main problem is that some IVR are extremely complex , like
>> questionnaires , etc.
>>
>> It would be great if we could mod_event_zmq to control the calls [?]
>>
>> Just to Sum it UP so far , so i can later add it to the wiki.
>>
>> Use Cent OS 64.
>> Use tmpfs for all the databases.
>>
>> Thank you all for helping and Happy Eastern.
>> Ant?nio Teixeira
>>
>>
>>
>>
>>
>> 2011/4/22 Michael Collins <msc at freeswitch.org>
>>
>>> FreeSWITCH runs well on many platforms. However, the devs are painfully
>>> aware that bleeding edge distros have bleeding edge gcc compilers and other
>>> interesting issues. That being said, CentOS 5.x is "stable" in that it's old
>>> and boring, therefore it has the least amount of drama. OTOH, some of our
>>> users have been having great success with Arch Linux (IRC:bougeyman) and FS,
>>> even though Arch uses very recent kernels.
>>>
>>> Bottom line: if you know what you're doing then you can probably run FS
>>> anywhere. If you don't know what you're doing then stick with CentOS 5.x or
>>> Debian Lenny until you do. (I run then both with zero issues, compiling
>>> latest git each day.)
>>>
>>> -MC
>>>
>>>
>>> On Fri, Apr 22, 2011 at 10:03 AM, curriegrad2004 <
>>> curriegrad2004 at gmail.com> wrote:
>>>
>>>> Freeswitch is targeted for CentOS 5.3, which in my opinion quite short
>>>> sighted for the developers to do this. However with the limited size
>>>> of developers and testers, I'm afraid there's not much platforms we
>>>> can throughly test and actually say "okay, FS will run flawlessly on X
>>>> distro"
>>>>
>>>> However you can always try messing with the CFLAG's mtune option and
>>>> see what it produces for you...
>>>>
>>>> 2011/4/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
>>>> > hi
>>>> >
>>>> > if you refer to my e-mail
>>>> >
>>>> > yes, we do use tmpfs on both variants but
>>>> > - delays occur with concurrent calls > 80-100
>>>> > - cps is limited to 5-10 on debian, with centos 30 cps and more are no
>>>> problem at all
>>>> >
>>>> > also cpu load, stability and overall performace have been much better
>>>> since using centos
>>>> >
>>>> > i just found out for me that debian works not as good for me as centos
>>>> does.
>>>> > btw. everywhere else debian is 1st choice (desktop, lamp, db etc.)
>>>> >
>>>> > br
>>>> >
>>>> >
>>>> > On 2011-04-21 23:04, Jay Binks wrote:
>>>> >
>>>> >> I have no such problems on debian .
>>>> >>
>>>> >> I use debian 5 with 2.6.18 kernel which is what Is recommended
>>>> >>
>>>> >> Are you using tmpfs ??
>>>> >>
>>>> >> Jay
>>>> >>
>>>> >>
>>>> >>
>>>> >> On 22/04/2011, at 3:26 AM, Christian L?schenkohl<
>>>> christian.loeschenkohl at xpirio.com>  wrote:
>>>> >>
>>>> >>> hi
>>>> >>>
>>>> >>> we did use debian too and had such performance issues (sip packet
>>>> delays, low cps).
>>>> >>> after using centos 64bit (as advised by the devs) all performance
>>>> problems are gone.
>>>> >>>
>>>> >>> br
>>>> >>>
>>>> >>> On 2011-04-21 18:24, Antonio Teixeira wrote:
>>>> >>>
>>>> >>>> Hello List.
>>>> >>>>
>>>> >>>> I'm currently integrating an IVR in python together with freeswitch
>>>> using mod_python and ESL and my life has been well until ...
>>>> >>>> The flow of calls went over 80 simultaneous calls.
>>>> >>>> Now freeswitch starts sending packets with huge delays ( even when
>>>> establishing the call , mainly the 200 ) and firing up the IVR with tons of
>>>> delay up to 20 seconds.
>>>> >>>>
>>>> >>>> So i searched the wiki forums and mailing list:
>>>> >>>>
>>>> >>>> Put freeswitch on a diet , trimmed modules.conf
>>>> >>>> Played with the ulimit stuff.
>>>> >>>> Played with the IVRS to reduce load to a minimum and i was able to
>>>> squeeze more 5 calls of performance.
>>>> >>>>
>>>> >>>> The problem is :
>>>> >>>>
>>>> >>>> Top shows
>>>> >>>> top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92,
>>>> 1.76, 1.78
>>>> >>>> Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0
>>>> zombie
>>>> >>>> Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,
>>>>  0.5%si,  0.0%st
>>>> >>>> Mem:   8193336k total,  1639156k used,  6554180k free,   177208k
>>>> buffers
>>>> >>>> Swap: 19534904k total,        0k used, 19534904k free,  1062272k
>>>> cached
>>>> >>>>
>>>> >>>>    PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
>>>>  COMMAND
>>>> >>>> 31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85
>>>> freeswitch
>>>> >>>>
>>>> >>>> Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
>>>> >>>>
>>>> >>>>
>>>> >>>> Some basic System Info :
>>>> >>>> Debian 6.0 ( i heard the timming module is affected by Debian , but
>>>> if the CPU % gets lower than 95% everything will be more stable)
>>>> >>>> Python 2.5
>>>> >>>>
>>>> >>>> 2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
>>>> >>>> 8 GB of Ram
>>>> >>>>
>>>> >>>> as you can see 94 % of the "Cpu Power" is sleeping :\
>>>> >>>>
>>>> >>>>
>>>> >>>> It appears freeswitch is only capable of using let's say "one
>>>> cpu"/thread ??
>>>> >>>> Do you guys recommend simply starting more instances or redoing the
>>>> IVR stuff.
>>>> >>>>
>>>> >>>>
>>>> >>>> Hope you guys can help me out.
>>>> >>>>
>>>> >>>> Thanks
>>>> >>>> Ant?nio Teixeira
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> >>>>
>>>> >>>> _______________________________________________
>>>> >>>> FreeSWITCH-users mailing list
>>>> >>>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>>> http://www.freeswitch.org
>>>> >>>
>>>> >>>
>>>> >>> --
>>>> >>> Ing. Christian L?schenkohl
>>>> >>> Technische Leitung, Forschung&  Entwicklung VoIP
>>>> >>>
>>>> >>> xpirio
>>>> >>> Telekommunikation&  Service GmbH
>>>> >>> Lakeside B04
>>>> >>> 9020 Klagenfurt
>>>> >>> Austria
>>>> >>>
>>>> >>> T  +43 5 77 11 - 1000
>>>> >>> F  +43 5 77 11 - 1002
>>>> >>> E  christian.loeschenkohl at xpirio.com
>>>> >>>
>>>> >>> _______________________________________________
>>>> >>> FreeSWITCH-users mailing list
>>>> >>> FreeSWITCH-users at lists.freeswitch.org
>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >>> http://www.freeswitch.org
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> > --
>>>> > Ing. Christian L?schenkohl
>>>> > Technische Leitung, Forschung & Entwicklung VoIP
>>>> >
>>>> > xpirio
>>>> > Telekommunikation & Service GmbH
>>>> > Lakeside B04
>>>> > 9020 Klagenfurt
>>>> > Austria
>>>> >
>>>> > T  +43 5 77 11 - 1000
>>>> > F  +43 5 77 11 - 1002
>>>> > E  christian.loeschenkohl at xpirio.com
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 vibha_dear6 at yahoo.co.in  Sat Apr 23 14:21:00 2011
From: vibha_dear6 at yahoo.co.in (vibha dear)
Date: Sat, 23 Apr 2011 15:51:00 +0530 (IST)
Subject: [Freeswitch-users] Contents of FreeSWITCH-users digest...
In-Reply-To: <mailman.8014.1303503801.24555.freeswitch-users@lists.freeswitch.org>
Message-ID: <93236.10448.qm@web137311.mail.in.yahoo.com>


thanks Micheal..!!?
--- On Sat, 23/4/11, freeswitch-users-request at lists.freeswitch.org <freeswitch-users-request at lists.freeswitch.org> wrote:

From: freeswitch-users-request at lists.freeswitch.org <freeswitch-users-request at lists.freeswitch.org>
Subject: FreeSWITCH-users Digest, Vol 58, Issue 157
To: freeswitch-users at lists.freeswitch.org
Date: Saturday, 23 April, 2011, 1:53 AM

Send FreeSWITCH-users mailing list submissions to
??? freeswitch-users at lists.freeswitch.org

To subscribe or unsubscribe via the World Wide Web, visit
??? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
or, via email, send a message with subject or body 'help' to
??? freeswitch-users-request at lists.freeswitch.org

You can reach the person managing the list at
??? freeswitch-users-owner at lists.freeswitch.org

When replying, please edit your Subject line so it is more specific
than "Re: Contents of FreeSWITCH-users digest..."
Today's Topics:

???1. Re: freeswitch xml cdr and csv (Michael Collins)
???2. Re: Tuning Up Freeswitch (Antonio Teixeira)
???3. Re: freeswitch xml cdr and csv (Ken Rice)
_______________________________________________
FreeSWITCH-users mailing 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 steveayre at gmail.com  Sat Apr 23 17:46:09 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 23 Apr 2011 14:46:09 +0100
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTin6c0q_66bNk-W+1WjVKkf+0w27FA@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<4DB068C4.90109@xpirio.com>
	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
	<4DB1550C.1050607@xpirio.com>
	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
	<BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>
	<BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com>
	<BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com>
	<BANLkTin6c0q_66bNk-W+1WjVKkf+0w27FA@mail.gmail.com>
Message-ID: <B40BD7F2-EEB9-4C1E-AE1F-6474FFFBAE71@gmail.com>

ESL would be best as you can offload the processing to another server - potentially a load balanced cluster.

Steve on iPhone

On 23 Apr 2011, at 08:51, Antonio Teixeira <eagle.antonio at gmail.com> wrote:

> Hello Michael & The rest.
> 
> What i'm trying to find is the solution with the highest performance for IVR Applications. And squeezing the last bit of performance out of freeswitch is a needed path specially on a industry that does second by second billing :P
> i'm going to do some performance testing with Lua , ESL And Mod Curl and of course python and measure some performance data and will post it later.
> 
> A/T
> 
> 
> 2011/4/22 Michael Collins <msc at freeswitch.org>
> You can definitely use the event socket. Heck, you can even use Python if you want to. The dev team wrote ESL specifically for cases like these - where you want to control FS externally. ESL beats the pants off AGI scripts. It gives you complete control over the system.
> 
> That all being said, you can do all sorts of stuff with Lua dialplan scripts. Check out chapter 7 of the FreeSWITCH book for some nice examples. Lua is lightweight and easy to learn. It's a good alternative for many cases. Check it out...
> 
> -MC
> 
> 
> On Fri, Apr 22, 2011 at 12:39 PM, Antonio Teixeira <eagle.antonio at gmail.com> wrote:
> Well i totally agree with the Dev team decision , it would be impossible for a DEV team to "get it right" in all the distros that's why i started this post.
> 
> But we also need to take into consideration that we are talking about IVR Processing ,  not auto-attendants or simply call pass trough. 
> This means heavy use of TTS / ASR , Database Logic and Scripts , GetDigits and the works.
> 
> I use python alot , but i think the mod_python is not the best tool for this job i admit that , that's why im currently looking for other solutions.
> I'm thinking in using mod_socket , but that scares me ( let? say bad experiences with Asterisk AGI) Or Mod Curl
> 
> The main problem is that some IVR are extremely complex , like questionnaires , etc.
> 
> It would be great if we could mod_event_zmq to control the calls <330.gif>
> 
> Just to Sum it UP so far , so i can later add it to the wiki.
> 
> Use Cent OS 64.
> Use tmpfs for all the databases.
> 
> Thank you all for helping and Happy Eastern.
> Ant?nio Teixeira
> 
> 
> 
> 
> 
> 2011/4/22 Michael Collins <msc at freeswitch.org>
> FreeSWITCH runs well on many platforms. However, the devs are painfully aware that bleeding edge distros have bleeding edge gcc compilers and other interesting issues. That being said, CentOS 5.x is "stable" in that it's old and boring, therefore it has the least amount of drama. OTOH, some of our users have been having great success with Arch Linux (IRC:bougeyman) and FS, even though Arch uses very recent kernels.
> 
> Bottom line: if you know what you're doing then you can probably run FS anywhere. If you don't know what you're doing then stick with CentOS 5.x or Debian Lenny until you do. (I run then both with zero issues, compiling latest git each day.)
> 
> -MC
> 
> 
> On Fri, Apr 22, 2011 at 10:03 AM, curriegrad2004 <curriegrad2004 at gmail.com> wrote:
> Freeswitch is targeted for CentOS 5.3, which in my opinion quite short
> sighted for the developers to do this. However with the limited size
> of developers and testers, I'm afraid there's not much platforms we
> can throughly test and actually say "okay, FS will run flawlessly on X
> distro"
> 
> However you can always try messing with the CFLAG's mtune option and
> see what it produces for you...
> 
> 2011/4/22 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>:
> > hi
> >
> > if you refer to my e-mail
> >
> > yes, we do use tmpfs on both variants but
> > - delays occur with concurrent calls > 80-100
> > - cps is limited to 5-10 on debian, with centos 30 cps and more are no problem at all
> >
> > also cpu load, stability and overall performace have been much better since using centos
> >
> > i just found out for me that debian works not as good for me as centos does.
> > btw. everywhere else debian is 1st choice (desktop, lamp, db etc.)
> >
> > br
> >
> >
> > On 2011-04-21 23:04, Jay Binks wrote:
> >
> >> I have no such problems on debian .
> >>
> >> I use debian 5 with 2.6.18 kernel which is what Is recommended
> >>
> >> Are you using tmpfs ??
> >>
> >> Jay
> >>
> >>
> >>
> >> On 22/04/2011, at 3:26 AM, Christian L?schenkohl<christian.loeschenkohl at xpirio.com>  wrote:
> >>
> >>> hi
> >>>
> >>> we did use debian too and had such performance issues (sip packet delays, low cps).
> >>> after using centos 64bit (as advised by the devs) all performance problems are gone.
> >>>
> >>> br
> >>>
> >>> On 2011-04-21 18:24, Antonio Teixeira wrote:
> >>>
> >>>> Hello List.
> >>>>
> >>>> I'm currently integrating an IVR in python together with freeswitch using mod_python and ESL and my life has been well until ...
> >>>> The flow of calls went over 80 simultaneous calls.
> >>>> Now freeswitch starts sending packets with huge delays ( even when establishing the call , mainly the 200 ) and firing up the IVR with tons of delay up to 20 seconds.
> >>>>
> >>>> So i searched the wiki forums and mailing list:
> >>>>
> >>>> Put freeswitch on a diet , trimmed modules.conf
> >>>> Played with the ulimit stuff.
> >>>> Played with the IVRS to reduce load to a minimum and i was able to squeeze more 5 calls of performance.
> >>>>
> >>>> The problem is :
> >>>>
> >>>> Top shows
> >>>> top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92, 1.76, 1.78
> >>>> Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0 zombie
> >>>> Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,  0.5%si,  0.0%st
> >>>> Mem:   8193336k total,  1639156k used,  6554180k free,   177208k buffers
> >>>> Swap: 19534904k total,        0k used, 19534904k free,  1062272k cached
> >>>>
> >>>>    PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
> >>>> 31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85 freeswitch
> >>>>
> >>>> Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
> >>>>
> >>>>
> >>>> Some basic System Info :
> >>>> Debian 6.0 ( i heard the timming module is affected by Debian , but if the CPU % gets lower than 95% everything will be more stable)
> >>>> Python 2.5
> >>>>
> >>>> 2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
> >>>> 8 GB of Ram
> >>>>
> >>>> as you can see 94 % of the "Cpu Power" is sleeping :\
> >>>>
> >>>>
> >>>> It appears freeswitch is only capable of using let's say "one cpu"/thread ??
> >>>> Do you guys recommend simply starting more instances or redoing the IVR stuff.
> >>>>
> >>>>
> >>>> Hope you guys can help me out.
> >>>>
> >>>> Thanks
> >>>> Ant?nio Teixeira
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>> --
> >>> Ing. Christian L?schenkohl
> >>> Technische Leitung, Forschung&  Entwicklung VoIP
> >>>
> >>> xpirio
> >>> Telekommunikation&  Service GmbH
> >>> Lakeside B04
> >>> 9020 Klagenfurt
> >>> Austria
> >>>
> >>> T  +43 5 77 11 - 1000
> >>> F  +43 5 77 11 - 1002
> >>> E  christian.loeschenkohl at xpirio.com
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > --
> > Ing. Christian L?schenkohl
> > Technische Leitung, Forschung & Entwicklung VoIP
> >
> > xpirio
> > Telekommunikation & Service GmbH
> > Lakeside B04
> > 9020 Klagenfurt
> > Austria
> >
> > T  +43 5 77 11 - 1000
> > F  +43 5 77 11 - 1002
> > E  christian.loeschenkohl at xpirio.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 23 20:56:19 2011
From: infos at madovsky.org (Madovsky)
Date: Sat, 23 Apr 2011 12:56:19 -0400
Subject: [Freeswitch-users] sounds_dir
Message-ID: <A9368C863649467F86467B0C68030C43@e1705>

I have (another stupid) question :

where the variable $${sounds_dir} is set ?
I see the variable used in vars.xml but not set everywhere in conf directory

Thanks
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From nasir at ictinnovations.com  Sat Apr 23 22:07:05 2011
From: nasir at ictinnovations.com (Nasir Iqbal)
Date: Sat, 23 Apr 2011 23:07:05 +0500
Subject: [Freeswitch-users] ZRTP Library
In-Reply-To: <01DD1957-7DA6-48D8-BF63-7F133F5417CC@freeswitch.org>
References: <BANLkTimBuiOKCZQ9AQFFtKJg23C0CGhEmQ@mail.gmail.com>
	<BANLkTimggX_sBqxWnZENNfPK+hQ+Rj0afQ@mail.gmail.com>
	<01DD1957-7DA6-48D8-BF63-7F133F5417CC@freeswitch.org>
Message-ID: <BANLkTimF_G-YFH6fiQ1JdK0NM1jQRgXJbg@mail.gmail.com>

At last I got it.

Thanks Everybody

Nasir Iqbal

ICT Innovations
http://www.ictinnovations.com/



On Sat, Apr 23, 2011 at 3:09 AM, Brian West <brian at freeswitch.org> wrote:

> And nobody can provide it without violating the crypto Export laws of the
> US.
>
> /b
>
> On Apr 22, 2011, at 1:44 PM, Michael Collins wrote:
>
> > I don't know if it's okay to distribute. I've pinged PRZ on this issue
> and
> > have not heard back from him. Sorry...
> >
> > -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 jcasale at activenetwerx.com  Sat Apr 23 23:24:50 2011
From: jcasale at activenetwerx.com (Joseph L. Casale)
Date: Sat, 23 Apr 2011 19:24:50 +0000
Subject: [Freeswitch-users] Changing From header in voicemail
	to	e-mail	notifications
In-Reply-To: <92097A6A775D5147B1078E3F15430B923EF3A7@prato.activenetwerx.local>
References: <SNT116-W1509E05D218F485CAC25029D920@phx.gbl>
	<BANLkTi=i9ycE6TYEwR38Kwc08OdtasZa4g@mail.gmail.com>
	<92097A6A775D5147B1078E3F15430B923EF3A7@prato.activenetwerx.local>
Message-ID: <assp.0094c03637.92097A6A775D5147B1078E3F15430B924117EE@prato.activenetwerx.local>

>>? You have to set the following in vars,xml:
>>?
>>? <X-PRE-PROCESS cmd="set" data="voicemail_account=username"/>
>>? <X-PRE-PROCESS cmd="set" data="voicemail_domain=your-domain.com"/>
>
>In one of my installs, (1.0.6) that never worked. Is that only valid in trunk?

Although the notify-voicemail.tpl & voicemail.tpl utilize ${voicemail_domain}
its ignored on my OpenWRT 1.0.6 setup (haven't tested on my CentOS switch.

A 'global_getvar' enumerates it as I wanted it...

I just manually setup the templates as I needed them...

HTH,
jlc


From lists at telefaks.de  Sat Apr 23 23:29:51 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Sat, 23 Apr 2011 21:29:51 +0200
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
In-Reply-To: <BANLkTikH9QMuhvKXqfBu78fOzqxoME3X8A@mail.gmail.com>
References: <4DAECCA8.1050203@gmx.net>
	<BANLkTikH9QMuhvKXqfBu78fOzqxoME3X8A@mail.gmail.com>
Message-ID: <4DB328AF.8090107@telefaks.de>

Thank you guru,

I tried the example in the wiki. This worked.
I wanted the agent also to wait again with MOH after the caller hung up.
This did not work in my environment (Freeswitch git April 2011). The
agent was always hungup after the caller hung up and was not transferred
to the same dialplan extension again.
Also hangup after bridge =false did not work.

Does this work in your environment?

Best regards
Peter


guru singh schrieb:
> Hi Peter,
>
> I've been using mod_callcenter for a while and must say it works really well.
> I just tried the uuid-standby strategy and basically it's exactly what
> you say the asterisk thing does.
> See the dialplan example.
> http://wiki.freeswitch.org/wiki/Mod_callcenter#uuid-standby
> Agent is dialing 4099 and listening to MOH. When a call arrives, it's
> bridged directly to the agent.
>
> Regards
> guru
>
> On Wed, Apr 20, 2011 at 5:38 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>   
>> Hello,
>>
>> I am trying to use the mod_callcenter functionality. This works nicely
>> so far so thank you to everybody involved for programming this nice module!
>> But I am stuck somehow with uuid-standby. Can anybody explain how
>> uuid-standby works?
>>
>> Another question: In the Asterisk based callcenter solution named
>> "Vicidial", an agent can be held permanently in a conference, waiting
>> for calls who are bridged to his uuid in the conference. Can this be
>> haviour be done with mod_callcenter?
>>
>>
>> Best regards
>> Peter
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


-- 
With kind regards
Peter Steinbach 

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




From paul at cupis.co.uk  Sat Apr 23 23:55:42 2011
From: paul at cupis.co.uk (Paul Cupis)
Date: Sat, 23 Apr 2011 20:55:42 +0100
Subject: [Freeswitch-users] sounds_dir
In-Reply-To: <A9368C863649467F86467B0C68030C43@e1705>
References: <A9368C863649467F86467B0C68030C43@e1705>
Message-ID: <4DB32EBE.2010100@cupis.co.uk>

On 23/04/11 17:56, Madovsky wrote:
> where the variable $${sounds_dir} is set ?
> I see the variable used in vars.xml but not set everywhere in conf directory

It is set at compile time to ${prefix}/sounds - see ./configure.in and
./src/switch_core.c

Regards,


From benkokakao at gmail.com  Sun Apr 24 00:13:01 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Sat, 23 Apr 2011 23:13:01 +0300
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
 with productivity license.
In-Reply-To: <471D76419F9EF642962323D13DF1DF69011E58@newserver.arneill-py.local>
References: <471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local>
	<BANLkTi=XmNDV51=xoNsViHUYaUZeyMpz_A@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E58@newserver.arneill-py.local>
Message-ID: <BANLkTi=o1nUa70X_PHU51M_TTGTfnSSdKg@mail.gmail.com>

> <attendant
> ?attendant.reg="1"
> ?attendant.ringType="1"
> ?attendant.behaviors.display.spontaneousCallAppearances.normal="1"
> ?attendant.behaviors.display.spontaneousCallAppearances.automata="1"
> ?attendant.behaviors.display.remoteCallerID.normal="1"
> ?attendant.behaviors.display.remoteCallerID.automata="1"
> ?attendant.resourceList.1.address="8000"
> ?attendant.resourceList.1.label="Name1"
> ?attendant.resourceList.2.address="8001"
> ?attendant.resourceList.2.label="Name2"
> />

I don't think this is the cause of the problem - but what about
attendant.resourceList.1.type?

 attendant.resourceList.1.type="automata"

None of these attendant-parameters are listed in the 3.1.7 sip admin
guide though(In the 3.3.0-guide they are).
I didn't look deeply into the docs - is this actually supposed to work in 3.1.7?

Regards
Christian


From benkokakao at gmail.com  Sun Apr 24 00:29:07 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Sat, 23 Apr 2011 23:29:07 +0300
Subject: [Freeswitch-users] OpenVox B100P BRI PTP on FS(libpri/dahdi)
Message-ID: <BANLkTikO9vZRZnhgNtVTXFqs0pt17_N93A@mail.gmail.com>

Hi!

Has anyone gotten a OpenVox B100P or other cheap HFC-S based ISDN-card
to work with freeswitch with BRI point-to-point?

I get signalling errors and line-status is DOWN("ERR]
ftmod_libpri.c:128 Don't know how to SABME on a type 0 node"). I'm
using  libpri-1.4.12-beta3 and dahdi 2.3.0.1(The last version
supported by the OpenVox patch).

I've opened a ticket with OpenVox(see
http://bbs.openvox.cn/viewthread.php?tid=1543 for more details) and
they've been responsive, but my feeling is that this is not going
anywhere soon.

Regards
Christian


From benkokakao at gmail.com  Sun Apr 24 00:32:41 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Sat, 23 Apr 2011 23:32:41 +0300
Subject: [Freeswitch-users] OpenVox B100P BRI PTP on FS(libpri/dahdi)
In-Reply-To: <BANLkTikO9vZRZnhgNtVTXFqs0pt17_N93A@mail.gmail.com>
References: <BANLkTikO9vZRZnhgNtVTXFqs0pt17_N93A@mail.gmail.com>
Message-ID: <BANLkTin06iV66utNekNq+r8y7LyRTOFn6w@mail.gmail.com>

On 23 April 2011 23:29, Christian Benke <benkokakao at gmail.com> wrote:
> Hi!
>
> Has anyone gotten a OpenVox B100P or other cheap HFC-S based ISDN-card
> to work with freeswitch with BRI point-to-point?

Here's a full debug/config of the setup btw:

http://pastebin.com/Kc4BPL9E


From grsingh750 at gmail.com  Sun Apr 24 01:51:27 2011
From: grsingh750 at gmail.com (guru singh)
Date: Sun, 24 Apr 2011 03:21:27 +0530
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
In-Reply-To: <4DB328AF.8090107@telefaks.de>
References: <4DAECCA8.1050203@gmx.net>
	<BANLkTikH9QMuhvKXqfBu78fOzqxoME3X8A@mail.gmail.com>
	<4DB328AF.8090107@telefaks.de>
Message-ID: <BANLkTikz_PSaE6T99jnrJSgEFJSWBpyxvw@mail.gmail.com>

Hi Peter

Try setting the status as 'Available' instead of 'Available (On Demand)'
In case of 'Available (On Demand)' after the call ends, the agent's
status is set to 'idle', so therefore no calls are given to the
specific agent. I'm not too sure if this is the only change required
to get the behavior you expect. I've not tried it on my box yet. I can
only do it Monday and let you know.

Regards,
guru

On Sun, Apr 24, 2011 at 12:59 AM, Peter Steinbach <lists at telefaks.de> wrote:
> Thank you guru,
>
> I tried the example in the wiki. This worked.
> I wanted the agent also to wait again with MOH after the caller hung up.
> This did not work in my environment (Freeswitch git April 2011). The
> agent was always hungup after the caller hung up and was not transferred
> to the same dialplan extension again.
> Also hangup after bridge =false did not work.
>
> Does this work in your environment?
>
> Best regards
> Peter
>
>
> guru singh schrieb:
>> Hi Peter,
>>
>> I've been using mod_callcenter for a while and must say it works really well.
>> I just tried the uuid-standby strategy and basically it's exactly what
>> you say the asterisk thing does.
>> See the dialplan example.
>> http://wiki.freeswitch.org/wiki/Mod_callcenter#uuid-standby
>> Agent is dialing 4099 and listening to MOH. When a call arrives, it's
>> bridged directly to the agent.
>>
>> Regards
>> guru
>>
>> On Wed, Apr 20, 2011 at 5:38 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>> Hello,
>>>
>>> I am trying to use the mod_callcenter functionality. This works nicely
>>> so far so thank you to everybody involved for programming this nice module!
>>> But I am stuck somehow with uuid-standby. Can anybody explain how
>>> uuid-standby works?
>>>
>>> Another question: In the Asterisk based callcenter solution named
>>> "Vicidial", an agent can be held permanently in a conference, waiting
>>> for calls who are bridged to his uuid in the conference. Can this be
>>> haviour be done with mod_callcenter?
>>>
>>>
>>> Best regards
>>> Peter
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From fraserredmond at gmail.com  Sun Apr 24 03:24:55 2011
From: fraserredmond at gmail.com (Fraser Redmond)
Date: Sat, 23 Apr 2011 19:24:55 -0400
Subject: [Freeswitch-users] one-way audio problem on some inbound
 gateways, but not others (and no outbound gateways)
In-Reply-To: <DACA955A-3653-4C0E-B2B5-8F6A4753AE97@ipeva.fr>
References: <BANLkTinuTg+DwXQ5_fU1K-mE+oqYgH7apQ@mail.gmail.com>
	<1A68D020-D97F-43E6-B83B-E3C762DAD665@freeswitch.org>
	<BANLkTi=p7=xZ_GziBTYegM=J3MkEOFRVCw@mail.gmail.com>
	<201104201942.45050.sos@sokhapkin.dyndns.org>
	<BANLkTi=5nROvHn=MU0J7xJX3THAvnwZ+4A@mail.gmail.com>
	<DACA955A-3653-4C0E-B2B5-8F6A4753AE97@ipeva.fr>
Message-ID: <BANLkTikxYKYC+TEGcOBFpuysa10LX1mc=Q@mail.gmail.com>

Thanks David - it's more than one gateway though (different companies,
different countries), so unlikely that both changed things at around the
same time.

Because it was different gateways (and not all gateways), and different
softphones (in different locations), the only common denominator was
freeswitch.

Only one other thing is consistent: the gateway that works is geographically
close to our server (both in California), but the 2 gateways that don't work
are both overseas. Otherwise the only common link is Freeswitch.

Cheers,
Fraser




On Fri, Apr 22, 2011 at 4:26 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:

> Any possibility that the faulty gateways have changed their architecture to
> hide media relays behind NAT, and that you have rtp-auto-adjust disabled or
> not correctly configured on your SIP profile ?
> You may see that in the logs.
> Another way to check this is to look the IP advertised by the gateway in
> the SDP, and to see if your FS is trying to send traffic to that IP.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 21/04/2011 ? 03:16, Fraser Redmond a ?crit :
>
> But it's odd that it's only happening with some gateways, but other
> gateways get two-way audio fine. That's what really has me puzzled.
>
> It's definitely not a problem at the extension side, as it happens with
> multiple phones, and multiple locations. (So not mic, firewall, router)
>
> Cheers,
> Fraser
>
>
>
>
> On Wed, Apr 20, 2011 at 7:42 PM, Sergey Okhapkin <sos at sokhapkin.dyndns.org
> > wrote:
>
>> No outgoing audio? Usually this happens if SIP ALG is enabled in your
>> router.
>>
>> On Wednesday 20 April 2011, Fraser Redmond wrote:
>> > No, I tried turning off the firewall, and as I said in the OP it works
>> with
>> > one of our other gateways.
>> >
>> > Mic works on that one gateway, and during the calls where the audio
>> isn't
>> > transmitted the mic-indicator goes up and down.
>> >
>> > Cheers,
>> > Fraser
>> >
>> > On Wed, Apr 20, 2011 at 5:46 PM, Brian West <brian at freeswitch.org>
>> wrote:
>> > > firewall issue? mic doesn't work?  can you get a pcap of all traffic?
>> > >
>> > > /b
>> > >
>> > > On Apr 20, 2011, at 3:04 PM, Fraser Redmond wrote:
>> > >
>> > > Thanks Brian, I'd appreciate you looking - I don't know what to look
>> for
>> > > in the sip traces (could be worth documenting some pointers in the
>> > > wiki?)
>> > >
>> > > The sip trace is here:
>> > > http://pastebin.freeswitch.org/16136
>> > >
>> > > I pressed enter a few times in the console before and after it
>> connected
>> > > to the extension, so about lines 490-600 is the relevant part.
>> > >
>> > > I also captured a pcap, in case that is of interest - let me know and
>> > > I'll email it directly.
>> > >
>> > > Thanks,
>> > > Fraser
>> > >
>> > >
>> > >
>> > > _______________________________________________
>> > > FreeSWITCH-users mailing list
>> > > FreeSWITCH-users at lists.freeswitch.org
>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Sun Apr 24 12:33:57 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Sun, 24 Apr 2011 10:33:57 +0200
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
In-Reply-To: <BANLkTikz_PSaE6T99jnrJSgEFJSWBpyxvw@mail.gmail.com>
References: <4DAECCA8.1050203@gmx.net>	<BANLkTikH9QMuhvKXqfBu78fOzqxoME3X8A@mail.gmail.com>	<4DB328AF.8090107@telefaks.de>
	<BANLkTikz_PSaE6T99jnrJSgEFJSWBpyxvw@mail.gmail.com>
Message-ID: <4DB3E075.1050202@telefaks.de>

Hello Guru,

thanks for your hint, however this did not help.
The point is that according to the dialplan the agent should be
transferred to the same extension again an wait. In my case, when the
call hanges up, there is no attempt to continue with the dialplan.
So I expect it does not have to do with the agent's setings.

Best regards
Peter


guru singh schrieb:
> Hi Peter
>
> Try setting the status as 'Available' instead of 'Available (On Demand)'
> In case of 'Available (On Demand)' after the call ends, the agent's
> status is set to 'idle', so therefore no calls are given to the
> specific agent. I'm not too sure if this is the only change required
> to get the behavior you expect. I've not tried it on my box yet. I can
> only do it Monday and let you know.
>
> Regards,
> guru
>
> On Sun, Apr 24, 2011 at 12:59 AM, Peter Steinbach <lists at telefaks.de> wrote:
>   
>> Thank you guru,
>>
>> I tried the example in the wiki. This worked.
>> I wanted the agent also to wait again with MOH after the caller hung up.
>> This did not work in my environment (Freeswitch git April 2011). The
>> agent was always hungup after the caller hung up and was not transferred
>> to the same dialplan extension again.
>> Also hangup after bridge =false did not work.
>>
>> Does this work in your environment?
>>
>> Best regards
>> Peter
>>
>>
>> guru singh schrieb:
>>     
>>> Hi Peter,
>>>
>>> I've been using mod_callcenter for a while and must say it works really well.
>>> I just tried the uuid-standby strategy and basically it's exactly what
>>> you say the asterisk thing does.
>>> See the dialplan example.
>>> http://wiki.freeswitch.org/wiki/Mod_callcenter#uuid-standby
>>> Agent is dialing 4099 and listening to MOH. When a call arrives, it's
>>> bridged directly to the agent.
>>>
>>> Regards
>>> guru
>>>
>>> On Wed, Apr 20, 2011 at 5:38 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>       
>>>> Hello,
>>>>
>>>> I am trying to use the mod_callcenter functionality. This works nicely
>>>> so far so thank you to everybody involved for programming this nice module!
>>>> But I am stuck somehow with uuid-standby. Can anybody explain how
>>>> uuid-standby works?
>>>>
>>>> Another question: In the Asterisk based callcenter solution named
>>>> "Vicidial", an agent can be held permanently in a conference, waiting
>>>> for calls who are bridged to his uuid in the conference. Can this be
>>>> haviour be done with mod_callcenter?
>>>>
>>>>
>>>> Best regards
>>>> Peter
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>>         
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>       
>> --
>> With kind regards
>> Peter Steinbach
>>
>> Telefaks Services GmbH
>> mailto:lists (att) telefaks.de
>> Internet: www.telefaks.de
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 grsingh750 at gmail.com  Sun Apr 24 16:39:31 2011
From: grsingh750 at gmail.com (guru singh)
Date: Sun, 24 Apr 2011 18:09:31 +0530
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
In-Reply-To: <4DB3E075.1050202@telefaks.de>
References: <4DAECCA8.1050203@gmx.net>
	<BANLkTikH9QMuhvKXqfBu78fOzqxoME3X8A@mail.gmail.com>
	<4DB328AF.8090107@telefaks.de>
	<BANLkTikz_PSaE6T99jnrJSgEFJSWBpyxvw@mail.gmail.com>
	<4DB3E075.1050202@telefaks.de>
Message-ID: <BANLkTi=_nMDp_kjCrDmUkfLM-tMXt+nZ1g@mail.gmail.com>

Hi Peter,

You're right. Please ignore my previous message, status 'Available (On
Demand)' should also be fine.
I've tried it and I see the same behavior as you. Reading the logs, I
can see that nothing after playback gets executed once the call is
hungup. It's not just transfer, any other application also is not
getting executing. Something is amiss.
Maybe moc or somebody else will point it out.

Regards,
guru

On Sun, Apr 24, 2011 at 2:03 PM, Peter Steinbach <lists at telefaks.de> wrote:
> Hello Guru,
>
> thanks for your hint, however this did not help.
> The point is that according to the dialplan the agent should be transferred
> to the same extension again an wait. In my case, when the call hanges up,
> there is no attempt to continue with the dialplan.
> So I expect it does not have to do with the agent's setings.
>
> Best regards
> Peter
>
>
> guru singh schrieb:
>
> Hi Peter
>
> Try setting the status as 'Available' instead of 'Available (On Demand)'
> In case of 'Available (On Demand)' after the call ends, the agent's
> status is set to 'idle', so therefore no calls are given to the
> specific agent. I'm not too sure if this is the only change required
> to get the behavior you expect. I've not tried it on my box yet. I can
> only do it Monday and let you know.
>
> Regards,
> guru
>
> On Sun, Apr 24, 2011 at 12:59 AM, Peter Steinbach <lists at telefaks.de> wrote:
>
>
> Thank you guru,
>
> I tried the example in the wiki. This worked.
> I wanted the agent also to wait again with MOH after the caller hung up.
> This did not work in my environment (Freeswitch git April 2011). The
> agent was always hungup after the caller hung up and was not transferred
> to the same dialplan extension again.
> Also hangup after bridge =false did not work.
>
> Does this work in your environment?
>
> Best regards
> Peter
>
>
> guru singh schrieb:
>
>
> Hi Peter,
>
> I've been using mod_callcenter for a while and must say it works really
> well.
> I just tried the uuid-standby strategy and basically it's exactly what
> you say the asterisk thing does.
> See the dialplan example.
> http://wiki.freeswitch.org/wiki/Mod_callcenter#uuid-standby
> Agent is dialing 4099 and listening to MOH. When a call arrives, it's
> bridged directly to the agent.
>
> Regards
> guru
>
> On Wed, Apr 20, 2011 at 5:38 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>
>
>
> Hello,
>
> I am trying to use the mod_callcenter functionality. This works nicely
> so far so thank you to everybody involved for programming this nice module!
> But I am stuck somehow with uuid-standby. Can anybody explain how
> uuid-standby works?
>
> Another question: In the Asterisk based callcenter solution named
> "Vicidial", an agent can be held permanently in a conference, waiting
> for calls who are bridged to his uuid in the conference. Can this be
> haviour be done with mod_callcenter?
>
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From philippe at ppmt.org  Sun Apr 24 23:18:27 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Sun, 24 Apr 2011 15:18:27 -0400
Subject: [Freeswitch-users] call not connecting sometime
Message-ID: <4DB47783.5090507@ppmt.org>

Hello,

I had the problem a few weeks ago but after a reinstall (unrelated to
that) the problem had gone so I put it
done as my messing up my system at the time!

But today I had the same issue again. When I call sometime the call is
not going through completely

The symptom on my side are that nothing happens (no ring tone) on the
other they say that the phones
rings but when they pick up the phone they can't hear anything.

Below is a siptrace where I changed the number and IP so I hope I didn't
mess it up too much.

http://pastebin.freeswitch.org/16165

Can someone let me know if they see something wrong? I tried to
understand but it is beyond me :(

Regards

/Philippe
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From vetali100 at gmail.com  Sun Apr 24 23:49:26 2011
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Sun, 24 Apr 2011 22:49:26 +0300
Subject: [Freeswitch-users] one-way audio problem on some inbound
 gateways, but not others (and no outbound gateways)
In-Reply-To: <BANLkTikxYKYC+TEGcOBFpuysa10LX1mc=Q@mail.gmail.com>
References: <BANLkTinuTg+DwXQ5_fU1K-mE+oqYgH7apQ@mail.gmail.com>
	<1A68D020-D97F-43E6-B83B-E3C762DAD665@freeswitch.org>
	<BANLkTi=p7=xZ_GziBTYegM=J3MkEOFRVCw@mail.gmail.com>
	<201104201942.45050.sos@sokhapkin.dyndns.org>
	<BANLkTi=5nROvHn=MU0J7xJX3THAvnwZ+4A@mail.gmail.com>
	<DACA955A-3653-4C0E-B2B5-8F6A4753AE97@ipeva.fr>
	<BANLkTikxYKYC+TEGcOBFpuysa10LX1mc=Q@mail.gmail.com>
Message-ID: <BANLkTik4n=AtxPewNrzNWNTU+UiE9tKEbQ@mail.gmail.com>

If you have access to the clients machine, you can install a network
monitoring tool there (Wireshark will be fine) and check OUTGOING RTP stream
during the call.

Make sure that RTP is sent to the port described in FreeSWITCH's OK reply
(for example 17252, from pcap file you provided, line #10).

Also check on the FS server, if it receives packets to this stream - use
"ngrep port xxxxx" command.

If data arrives, then problem is in FreeSWITCH (almost unlikely). If not,
then you will check further at the routers in between.

Maybe some router is doing some manipulations with the port number, so data
is lost before arriving to FS.

Vitalie



2011/4/24 Fraser Redmond <fraserredmond at gmail.com>

> Thanks David - it's more than one gateway though (different companies,
> different countries), so unlikely that both changed things at around the
> same time.
>
> Because it was different gateways (and not all gateways), and different
> softphones (in different locations), the only common denominator was
> freeswitch.
>
> Only one other thing is consistent: the gateway that works is
> geographically close to our server (both in California), but the 2 gateways
> that don't work are both overseas. Otherwise the only common link is
> Freeswitch.
>
> Cheers,
> Fraser
>
>
>
>
>
> On Fri, Apr 22, 2011 at 4:26 AM, David Ponzone <david.ponzone at ipeva.fr>wrote:
>
>> Any possibility that the faulty gateways have changed their architecture
>> to hide media relays behind NAT, and that you have rtp-auto-adjust disabled
>> or not correctly configured on your SIP profile ?
>> You may see that in the logs.
>> Another way to check this is to look the IP advertised by the gateway in
>> the SDP, and to see if your FS is trying to send traffic to that IP.
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 21/04/2011 ? 03:16, Fraser Redmond a ?crit :
>>
>> But it's odd that it's only happening with some gateways, but other
>> gateways get two-way audio fine. That's what really has me puzzled.
>>
>> It's definitely not a problem at the extension side, as it happens with
>> multiple phones, and multiple locations. (So not mic, firewall, router)
>>
>> Cheers,
>> Fraser
>>
>>
>>
>>
>> On Wed, Apr 20, 2011 at 7:42 PM, Sergey Okhapkin <
>> sos at sokhapkin.dyndns.org> wrote:
>>
>>> No outgoing audio? Usually this happens if SIP ALG is enabled in your
>>> router.
>>>
>>> On Wednesday 20 April 2011, Fraser Redmond wrote:
>>> > No, I tried turning off the firewall, and as I said in the OP it works
>>> with
>>> > one of our other gateways.
>>> >
>>> > Mic works on that one gateway, and during the calls where the audio
>>> isn't
>>> > transmitted the mic-indicator goes up and down.
>>> >
>>> > Cheers,
>>> > Fraser
>>> >
>>> > On Wed, Apr 20, 2011 at 5:46 PM, Brian West <brian at freeswitch.org>
>>> wrote:
>>> > > firewall issue? mic doesn't work?  can you get a pcap of all traffic?
>>> > >
>>> > > /b
>>> > >
>>> > > On Apr 20, 2011, at 3:04 PM, Fraser Redmond wrote:
>>> > >
>>> > > Thanks Brian, I'd appreciate you looking - I don't know what to look
>>> for
>>> > > in the sip traces (could be worth documenting some pointers in the
>>> > > wiki?)
>>> > >
>>> > > The sip trace is here:
>>> > > http://pastebin.freeswitch.org/16136
>>> > >
>>> > > I pressed enter a few times in the console before and after it
>>> connected
>>> > > to the extension, so about lines 490-600 is the relevant part.
>>> > >
>>> > > I also captured a pcap, in case that is of interest - let me know and
>>> > > I'll email it directly.
>>> > >
>>> > > Thanks,
>>> > > Fraser
>>> > >
>>> > >
>>> > >
>>> > > _______________________________________________
>>> > > FreeSWITCH-users mailing list
>>> > > FreeSWITCH-users at lists.freeswitch.org
>>> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > > http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 vetali100 at gmail.com  Mon Apr 25 00:18:27 2011
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Sun, 24 Apr 2011 23:18:27 +0300
Subject: [Freeswitch-users] call not connecting sometime
In-Reply-To: <4DB47783.5090507@ppmt.org>
References: <4DB47783.5090507@ppmt.org>
Message-ID: <BANLkTik=FgKtQOhm7gJkABXnkYfUeoC_6g@mail.gmail.com>

Not sure, but maybe it is related to this:


   1. 2011-04-24 11:02:33.597127 [DEBUG] switch_core_session.c:1939 Application
   start_dtmf_generate Requires media! pre_answering channel sofia/internal/
   1234 at 172.20.0.20
   2. 2011-04-24 11:02:33.597127 [INFO] switch_core_session.c:1941 Sending
   early media
   3. 2011-04-24 11:02:33.597127 [DEBUG] sofia_glue.c:3033 AUDIO RTP [
   sofia/internal/1234 at 172.20.0.20] 172.20.0.20 port 26668 -> 172.20.0.50
    port 16442 codec: 0 ms: 20


Maybe try to disable temporarily the "start_dtmf_generate"
and see if it helps.

Vitalie


2011/4/24 Philippe Le Toquin <philippe at ppmt.org>

>  Hello,
>
> I had the problem a few weeks ago but after a reinstall (unrelated to that)
> the problem had gone so I put it
> done as my messing up my system at the time!
>
> But today I had the same issue again. When I call sometime the call is not
> going through completely
>
> The symptom on my side are that nothing happens (no ring tone) on the other
> they say that the phones
> rings but when they pick up the phone they can't hear anything.
>
> Below is a siptrace where I changed the number and IP so I hope I didn't
> mess it up too much.
>
> http://pastebin.freeswitch.org/16165
>
> Can someone let me know if they see something wrong? I tried to understand
> but it is beyond me :(
>
> Regards
>
> /Philippe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 philippe at ppmt.org  Mon Apr 25 00:29:57 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Sun, 24 Apr 2011 16:29:57 -0400
Subject: [Freeswitch-users] call not connecting sometime
In-Reply-To: <BANLkTik=FgKtQOhm7gJkABXnkYfUeoC_6g@mail.gmail.com>
References: <4DB47783.5090507@ppmt.org>
	<BANLkTik=FgKtQOhm7gJkABXnkYfUeoC_6g@mail.gmail.com>
Message-ID: <4DB48845.3000403@ppmt.org>

Hello,

Thanks for the pointer. I checked another trace where it works and right
after the start_dtmf_generate message
I can see this message

2011-04-24 11:05:50.415651 [INFO] switch_core_session.c:1941 Sending
early media

I will try to deactivate it.

On 11-04-24 04:18 PM, Vitalie Colosov wrote:
> Not sure, but maybe it is related to this: 
>
>   1.
>       2011-04-24 11:02:33.597127 [DEBUG] switch_core_session.c:1939 Application
>       start_dtmf_generate Requires media! pre_answering channel
>       sofia/internal/1234 at 172.20.0.20
>   2.
>       2011-04-24 11:02:33.597127 [INFO] switch_core_session.c:1941 Sending
>       early media
>   3.
>       2011-04-24 11:02:33.597127 [DEBUG] sofia_glue.c:3033 AUDIO
>       RTP [sofia/internal/1234 at 172.20.0.20] 172.20.0.20 port 26668 -> 172.20.0.50 port 16442 codec: 0 ms: 20
>
>
> Maybe try to disable temporarily the "start_dtmf_generate"
> and see if it helps.
>
> Vitalie
>
>
> 2011/4/24 Philippe Le Toquin <philippe at ppmt.org
> <mailto:philippe at ppmt.org>>
>
>     Hello,
>
>     I had the problem a few weeks ago but after a reinstall (unrelated
>     to that) the problem had gone so I put it
>     done as my messing up my system at the time!
>
>     But today I had the same issue again. When I call sometime the
>     call is not going through completely
>
>     The symptom on my side are that nothing happens (no ring tone) on
>     the other they say that the phones
>     rings but when they pick up the phone they can't hear anything.
>
>     Below is a siptrace where I changed the number and IP so I hope I
>     didn't mess it up too much.
>
>     http://pastebin.freeswitch.org/16165
>
>     Can someone let me know if they see something wrong? I tried to
>     understand but it is beyond me :(
>
>     Regards
>
>     /Philippe
>
>     _______________________________________________
>     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 ckmonkey158 at yahoo.com  Sat Apr 23 06:38:52 2011
From: ckmonkey158 at yahoo.com (Chris Monkey)
Date: Fri, 22 Apr 2011 19:38:52 -0700 (PDT)
Subject: [Freeswitch-users] conference_set_auto_outcall No Audio
Message-ID: <397737.87013.qm@web59401.mail.ac4.yahoo.com>

Hi,

I am am relatively new to FS and have setup a home PBX with 
about 5 Snom 320s with TLS. Everything works fine, except when I try to 
use the mad_boss example, or any auto_outcall conference, the phones 
ring and will auto answer, but there is no audio. However, if I dial in 
to the conference, everything works fine.

I've tried adding 
<action application="set" data="sip_secure_media=true"/> to the 
conference extension, but to no avail. I don't see any errors in the 
log, and I just did a make clean this afternoon. What am I missing?

Thanks
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From msc at freeswitch.org  Mon Apr 25 02:32:20 2011
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 24 Apr 2011 15:32:20 -0700
Subject: [Freeswitch-users] setup an enviroment for implementating /
 testing IVR
In-Reply-To: <BANLkTin3zrNYJtrhdQXaA3r80b8qCvdo5A@mail.gmail.com>
References: <BANLkTin3zrNYJtrhdQXaA3r80b8qCvdo5A@mail.gmail.com>
Message-ID: <BANLkTikAe09zHoy3R45FBVwcxXF9W980wQ@mail.gmail.com>

Do you have a sound file recorded already? If so you can do a few different
things:

#1 - Emulate x5000 (demo IVR) by creating your own XML IVR file and
specifying your sound file.

#2 - Create a simple extension with play_and_get_digits. An example is found
here:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_play_and_get_digits

And yes, you *will* need to create an extension of some sort to route the
call. If the call is "external" - from say a VoIP provider - and coming in
via a DID then you will need to make an entry in the public dialplan
context. Check this page for an example of doing that:
http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_9:_Routing_DID_to_an_extension

-MC

On Fri, Apr 22, 2011 at 5:21 PM, Frankie Yiu <frankie.k.yiu at gmail.com>wrote:

> Thanks for the input.  But dialing 5000 would be testing the demo IVR, but
> my question is that how do I make an incoming call (Do I have to dial a
> extension number?) so that I can hear my own custom audio?
>
> Sorry for the confusing/dumb question.
>
> Thank you.
>
> Frankie
>
>
>
>
>>
>> ---------- Forwarded message ----------
>> From: Philippe Le Toquin <philippe at ppmt.org>
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Thu, 21 Apr 2011 21:27:10 -0400
>> Subject: Re: [Freeswitch-users] setup an enviroment for implementating /
>> testing IVR feature
>>  Hello,
>>
>> This link is a good start to get to do things with your new Freeswitch
>>
>> http://wiki.freeswitch.org/wiki/Getting_Started_Guide
>>
>> to test IVR just dial 5000 from your softphone to access a demo IVR
>>
>> One thing to change is the default password of the extension to make it a
>> bit harder to crack :)
>> Look in the vars.xml file
>>
>> /Philippe
>>
>> On 11-04-21 08:45 PM, Frankie Yiu wrote:
>>
>> Hi there,
>>
>> I would like to start implementating an IVR feature, but the first thing I
>> need to do is to able to use a softphone to dial into my freeSWITCH so that
>> an audio would start playing.
>> Right now I have the original configuration of freeSWITCH which I am not
>> sure how to dial into the system using the softphone.  For example, what
>> extension I have to call; or if I need to change anything in dialplan,
>> or configuration, etc.
>>
>> I have C# code that let me make outgoing call to my softphone, play audio
>> or collect DTMF.  But I am not sure what to do for incoming call using the
>> original configuration from freeSWITCH.
>>
>> Thanks in advance.
>>
>> Frankie
>>
>>
>> _______________________________________________
>> 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
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Boris Kovalenko <boris at tagnet.ru>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date: Fri, 22 Apr 2011 08:01:36 +0600
>> Subject: [Freeswitch-users] variable direction
>> Hello!
>>
>>    I found that with gateways I may use <variable .... direction="..."/>.
>> Is this possible with regular (directory) users too? If not, may somebody
>> explain why?
>>
>> --
>> ? ?????????,
>>  ????? ?????????
>>  ??? "??????"
>>  ???. +7 (3435) 230001
>>  ???? +7 (3435) 230005
>>
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: David Ponzone <david.ponzone at ipeva.fr>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date: Fri, 22 Apr 2011 04:04:21 +0200
>> Subject: [Freeswitch-users] Ringback while NOT YET bridging to leg B
>> Hi all,
>>
>> I am trying to achieve something weird, I admit.
>> I would need to receive a call from A, then wait X seconds before bridging
>> to B, but I'd like to send back a ringback to A while sleeping X seconds.
>> The point is to allow customers to have a nice ringback, which the
>> beginning of HAS to be heard by callers (like an announce in a call-center).
>>
>> A (ugly) way to do that would be to have 2 sound files: one to be played
>> when answering with playback, and then another one for the ringback. I would
>> then not use sleep at all, but the bridge would only happen when the first
>> file is over, achieving quite the same result.
>> The drawback of this is that I need to ask customers for 2 sound files,
>> and that can be a pain with most of them.
>>
>> I tried all combinations I could think of with ringback, instant_ringback,
>> with no luck.
>>
>> Any ideas ?
>>
>>  David Ponzone  Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel:      01 74 03 18 97
>> gsm:   06 66 98 76 34
>>
>> Service Client IPeva
>> tel:      0811 46 26 26
>> www.ipeva.fr  -   www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Yehavi Bourvine <yehavi.bourvine at gmail.com>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date: Fri, 22 Apr 2011 07:11:34 +0300
>> Subject: Re: [Freeswitch-users] Changing From header in voicemail to
>> e-mail notifications
>> Hello John,
>>
>>   You have to set the following in vars,xml:
>>
>>   <X-PRE-PROCESS cmd="set" data="voicemail_account=*username*"/>
>>   <X-PRE-PROCESS cmd="set" data="voicemail_domain=*your-domain.com*"/>
>>
>>                  Regards, __Yehavi:
>>  2011/4/21 John Platts <john_platts at hotmail.com>
>>
>>>  We see FreeSWITCH mod_voicemail in the From header of the voicemail to
>>> e-mail notifications. We also want to change the E-mail address in the From
>>> header to come from a specific domain. How do we go about changing the From:
>>> header in the voicemail to e-mail notifications? We do not want FreeSWITCH
>>> mod_voicemail to appear in the From header of the e-mail messages.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Nasir Iqbal <nasir at ictinnovations.com>
>> To: freeswitch-users at lists.freeswitch.org
>> Date: Fri, 22 Apr 2011 08:44:32 +0500
>> Subject: [Freeswitch-users] ZRTP Library
>> Hi,
>>
>> As zfone download server is offline, is there anyone who can provide me
>> libzrtp SDK source?
>>
>>
>> Nasir Iqbal
>>
>> ICT Innovations
>> http://www.ictinnovations.com/
>>
>>
>>
>> ---------- Forwarded message ----------
>> From: Michael Collins <msc at freeswitch.org>
>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> Date: Thu, 21 Apr 2011 22:49:26 -0700
>> Subject: Re: [Freeswitch-users] variable direction
>> The direction variable is set at the time of the call based upon whether
>> or not the user is the caller (outbound) or callee (inbound).
>>
>> -MC
>>
>> On Thu, Apr 21, 2011 at 7:01 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
>>
>>> Hello!
>>>
>>>     I found that with gateways I may use <variable ....
>>> direction="..."/>. Is this possible with regular (directory) users too?
>>> If not, may somebody explain why?
>>>
>>> --
>>> ? ?????????,
>>>   ????? ?????????
>>>   ??? "??????"
>>>   ???. +7 (3435) 230001
>>>   ???? +7 (3435) 230005
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon Apr 25 04:46:52 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sun, 24 Apr 2011 17:46:52 -0700 (PDT)
Subject: [Freeswitch-users] mod_enum compilation error
Message-ID: <1303692412837-6301771.post@n2.nabble.com>

I have a local FS git pulled to adffe07f7512b00f7093091778965a119ca81513 and
failed to compile mod_enum for OpenWRT as shown below:

making all mod_enum
make[7]: Entering directory
`/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/mod/applications/mod_enum'
make[7]: *** No rule to make target `Makefile.am', needed by `Makefile.in'. 
Stop.
make[7]: Leaving directory
`/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/mod/applications/mod_enum'
make[6]: *** [mod_enum-all] Error 1
make[6]: Leaving directory
`/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/mod'
make[5]: *** [all-recursive] Error 1
make[5]: Leaving directory
`/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src'
make[4]: *** [all-recursive] Error 1
make[4]: Leaving directory
`/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
make[3]: *** [all] Error 2
make[3]: Leaving directory
`/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
make[2]: ***
[/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/.built]
Error 2
make[2]: Leaving directory
`/opt/tmp/OpenWRT/feeds/packages/net/freeswitch_git'
make[1]: *** [package/feeds/local/freeswitch_git/compile] Error 2
make[1]: Leaving directory `/opt/tmp/openwrt-svn-trunk'
make: *** [package/freeswitch_git/compile] Error 2


-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-enum-compilation-error-tp6301771p6301771.html
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From fieldpeak at gmail.com  Mon Apr 25 09:42:45 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 25 Apr 2011 13:42:45 +0800
Subject: [Freeswitch-users] config ramdisk on CentOS 5.5
In-Reply-To: <C9D7B974.14501%krice@freeswitch.org>
References: <BANLkTinTkAR64kGsZbah5=Z3Hyxf8MwQiw@mail.gmail.com>
	<C9D7B974.14501%krice@freeswitch.org>
Message-ID: <BANLkTim4SX5JTB0-h=JaMsoyBtyW_wbEwA@mail.gmail.com>

Hi Ken,

Thanks for your reply.

if my below understanding correct?

if i add below to the end of /etc/fstab,
*tmpfs /usr/local/freeswitch/db tmpfs defaults 0 0 *

and then reboot machine, the system will auto mount the tmpfs.
i don't need run mannually *mount /usr/local/freeswitch/db* before
start freeswitch...

Thanks.

Regards,
Charles


2011/4/23 Ken Rice <krice at freeswitch.org>

>  Tmpfs is not a program... Read that page a little closer... That?s
> particular line is for your fstab...
>
> If you want to mount it from the command line its
> mount ?o tmpfs tmpfs /usr/local/freeswitch/db
>
> K
>
>
>
> On 4/22/11 10:50 PM, "fieldpeak" <fieldpeak at gmail.com> wrote:
>
> i'm trying tuning the FS to max performance on centos 5.5, and referred to
>
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#FreeSWITCH.27s_core.db_I.2FO_bottleneck
>
> #1,  i configure the DB of FS to ramdisk ,
>
> when i run "tmpfs /opt/freeswitch/db tmpfs defaults 0 0", it output:
> "-bash: tmpfs: command not found"
>
> #2,  i run "ethtool -g eth0", the output is below, what value i should
> config for RX and TX for max performance...
> Ring parameters for eth0:
> Pre-set maximums:
> RX:             4096
> RX Mini:        0
> RX Jumbo:       0
> TX:             4096
> Current hardware settings:
> RX:             256
> RX Mini:        0
> RX Jumbo:       0
> TX:             256
>
> Apprecited if anyone help how to configure it... 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
>
>
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From zetruger at gmail.com  Mon Apr 25 10:06:23 2011
From: zetruger at gmail.com (=?KOI8-R?B?6dfBziD+ydPU0cvP1w==?=)
Date: Mon, 25 Apr 2011 10:06:23 +0400
Subject: [Freeswitch-users] HOLD - signals from phones? How to play MoH?
In-Reply-To: <EFCB5275-7FF4-460A-9381-145B45E73B06@freeswitch.org>
References: <BANLkTikc5sR-7saccTXnVGiut2R7kwEY-w@mail.gmail.com>
	<EFCB5275-7FF4-460A-9381-145B45E73B06@freeswitch.org>
Message-ID: <BANLkTi=x2esd2ForfSkET5NyhRGOD7pjaA@mail.gmail.com>

TDM phones via audiocodes mp-114
And SFLphone

2011/4/22 Brian West <brian at freeswitch.org>:
> What phones are involved?
>
> /b
>
> On Apr 22, 2011, at 3:17 AM, ???? ???????? wrote:
>
>> In a bridge mode, when i push a button hold, FreeSWITCH don't playing MoH.
>> But if I set hold through FS console it's works.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Mon Apr 25 10:17:45 2011
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 25 Apr 2011 01:17:45 -0500
Subject: [Freeswitch-users] config ramdisk on CentOS 5.5
In-Reply-To: <BANLkTim4SX5JTB0-h=JaMsoyBtyW_wbEwA@mail.gmail.com>
Message-ID: <C9DA7C39.1453D%krice@freeswitch.org>

That is correct

Keep in mind that in using a ramdrive for freeswitch/db if you are running
voicemail you?ll need to look at mechanismd for backing up its db

K


On 4/25/11 12:42 AM, "fieldpeak" <fieldpeak at gmail.com> wrote:

> Hi Ken,
> 
> Thanks for your reply.
> 
> if my below understanding correct?
> if i add below to the end of /etc/fstab,
> tmpfs /usr/local/freeswitch/db tmpfs defaults 0 0
> 
> and then reboot machine, the system will auto mount the tmpfs.
> i don't need run mannually mount /usr/local/freeswitch/db before start
> freeswitch...
> 
> Thanks.
> 
> Regards,
> Charles
> 
> 2011/4/23 Ken Rice <krice at freeswitch.org>
>> Tmpfs is not a program... Read that page a little closer... That?s particular
>> line is for your fstab...
>> 
>> If you want to mount it from the command line its
>> mount ?o tmpfs tmpfs /usr/local/freeswitch/db ???
>> 
>> K
>> 
>> 
>> 
>> On 4/22/11 10:50 PM, "fieldpeak" <fieldpeak at gmail.com
>> <http://fieldpeak at gmail.com> > wrote:
>> 
>>> i'm trying tuning the FS to max performance on centos 5.5, and referred to
>>> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#FreeS
>>> WITCH.27s_core.db_I.2FO_bottleneck
>>> 
>>> #1,? i configure the DB of FS to ramdisk ,
>>> 
>>> when i run "tmpfs /opt/freeswitch/db tmpfs defaults 0 0", it output: "-bash:
>>> tmpfs: command not found"
>>> 
>>> #2,? i run "ethtool -g eth0", the output is below, what value i should
>>> config for RX and TX for max performance...
>>> Ring parameters for eth0:
>>> Pre-set maximums:
>>> RX:???????????? 4096
>>> RX Mini:??????? 0
>>> RX Jumbo:?????? 0
>>> TX:???????????? 4096
>>> Current hardware settings:
>>> RX:???????????? 256
>>> RX Mini:??????? 0
>>> RX Jumbo:?????? 0
>>> TX:???????????? 256
>>> 
>>> Apprecited if anyone help how to configure it... thanks!
>>> 
>>> Regards,
>>> Charles
>>> 
>>> 
>>> 
>>> _______________________________________________
>>> 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
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon Apr 25 11:03:54 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 25 Apr 2011 15:03:54 +0800
Subject: [Freeswitch-users] config ramdisk on CentOS 5.5
In-Reply-To: <C9DA7C39.1453D%krice@freeswitch.org>
References: <BANLkTim4SX5JTB0-h=JaMsoyBtyW_wbEwA@mail.gmail.com>
	<C9DA7C39.1453D%krice@freeswitch.org>
Message-ID: <BANLkTi=aDMwRvUv6EdNG6Fx-ctaYPZ+nPA@mail.gmail.com>

thanks Ken. I noticed the voice mail issue when using tmpfs, thanks again.

Regards,
Charles

On 4/25/11, Ken Rice <krice at freeswitch.org> wrote:
> That is correct
>
> Keep in mind that in using a ramdrive for freeswitch/db if you are running
> voicemail you?ll need to look at mechanismd for backing up its db
>
> K
>
>
> On 4/25/11 12:42 AM, "fieldpeak" <fieldpeak at gmail.com> wrote:
>
>> Hi Ken,
>>
>> Thanks for your reply.
>>
>> if my below understanding correct?
>> if i add below to the end of /etc/fstab,
>> tmpfs /usr/local/freeswitch/db tmpfs defaults 0 0
>>
>> and then reboot machine, the system will auto mount the tmpfs.
>> i don't need run mannually mount /usr/local/freeswitch/db before start
>> freeswitch...
>>
>> Thanks.
>>
>> Regards,
>> Charles
>>
>> 2011/4/23 Ken Rice <krice at freeswitch.org>
>>> Tmpfs is not a program... Read that page a little closer... That?s
>>> particular
>>> line is for your fstab...
>>>
>>> If you want to mount it from the command line its
>>> mount ?o tmpfs tmpfs /usr/local/freeswitch/db
>>>
>>> K
>>>
>>>
>>>
>>> On 4/22/11 10:50 PM, "fieldpeak" <fieldpeak at gmail.com
>>> <http://fieldpeak at gmail.com> > wrote:
>>>
>>>> i'm trying tuning the FS to max performance on centos 5.5, and referred
>>>> to
>>>> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations#FreeS
>>>> WITCH.27s_core.db_I.2FO_bottleneck
>>>>
>>>> #1,? i configure the DB of FS to ramdisk ,
>>>>
>>>> when i run "tmpfs /opt/freeswitch/db tmpfs defaults 0 0", it output:
>>>> "-bash:
>>>> tmpfs: command not found"
>>>>
>>>> #2,? i run "ethtool -g eth0", the output is below, what value i should
>>>> config for RX and TX for max performance...
>>>> Ring parameters for eth0:
>>>> Pre-set maximums:
>>>> RX:???????????? 4096
>>>> RX Mini:??????? 0
>>>> RX Jumbo:?????? 0
>>>> TX:???????????? 4096
>>>> Current hardware settings:
>>>> RX:???????????? 256
>>>> RX Mini:??????? 0
>>>> RX Jumbo:?????? 0
>>>> TX:???????????? 256
>>>>
>>>> Apprecited if anyone help how to configure it... thanks!
>>>>
>>>> Regards,
>>>> Charles
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 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
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.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  Mon Apr 25 12:15:19 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 25 Apr 2011 16:15:19 +0800
Subject: [Freeswitch-users] Build and install FS with new customized folder
	on CentOS 5.5
Message-ID: <BANLkTinOFchbfvGW7cDOv=f5S0goFWcmig@mail.gmail.com>

i added a folders(e.g. new_folder) inside the FS source code folder, inside
the new_folder there are some files, like below,

./freeswitch/conf
./freeswitch/bin
./freeswitch/*new_folder*/files1, files2...

i need this folder and the files under it will be installed when i run 'make
install', however, it did not.

the commands i run as below, i assume bootstrap.sh will run autoscan which
will find my new_folder...
./bootstrap.sh
make
make install

anyone can help how to realize it? thanks.

Regards,
Charles
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From fdelawarde at wirelessmundi.com  Mon Apr 25 12:24:01 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Mon, 25 Apr 2011 10:24:01 +0200
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <B40BD7F2-EEB9-4C1E-AE1F-6474FFFBAE71@gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<4DB068C4.90109@xpirio.com>
	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
	<4DB1550C.1050607@xpirio.com>
	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
	<BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>
	<BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com>
	<BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com>
	<BANLkTin6c0q_66bNk-W+1WjVKkf+0w27FA@mail.gmail.com>
	<B40BD7F2-EEB9-4C1E-AE1F-6474FFFBAE71@gmail.com>
Message-ID: <1303719841.19225.29.camel@luna.tc.commsmundi.com>

I also experienced a huge delay issue when attacked by mister
"friendly-scanner". The machine was not overloaded and call setup was
working well but with a 10-20 seconds delay. Note that the delay was
only with SIP answers, and once RTP was setup the audio was perfect.

Following advices from http://wiki.freeswitch.org/wiki/QoS solved it in
my case. Is there a mechanism in FS that could detect huge call setup
delay and simply reject calls?

Fran?ois.



On Sat, 2011-04-23 at 14:46 +0100, Steven Ayre wrote: 
> ESL would be best as you can offload the processing to another server
> - potentially a load balanced cluster.
> 
> Steve on iPhone
> 
> On 23 Apr 2011, at 08:51, Antonio Teixeira <eagle.antonio at gmail.com>
> wrote:
> 
> 
> 
> > Hello Michael & The rest.
> > 
> > What i'm trying to find is the solution with the highest performance
> > for IVR Applications. And squeezing the last bit of performance out
> > of freeswitch is a needed path specially on a industry that does
> > second by second billing :P
> > i'm going to do some performance testing with Lua , ESL And Mod Curl
> > and of course python and measure some performance data and will post
> > it later.
> > 
> > A/T
> > 
> > 
> > 2011/4/22 Michael Collins <msc at freeswitch.org>
> >         You can definitely use the event socket. Heck, you can even
> >         use Python if you want to. The dev team wrote ESL
> >         specifically for cases like these - where you want to
> >         control FS externally. ESL beats the pants off AGI scripts.
> >         It gives you complete control over the system. 
> >         
> >         
> >         That all being said, you can do all sorts of stuff with Lua
> >         dialplan scripts. Check out chapter 7 of the FreeSWITCH book
> >         for some nice examples. Lua is lightweight and easy to
> >         learn. It's a good alternative for many cases. Check it
> >         out... 
> >         
> >         
> >         -MC 
> >         
> >         
> >         
> >         On Fri, Apr 22, 2011 at 12:39 PM, Antonio Teixeira
> >         <eagle.antonio at gmail.com> wrote:
> >                 Well i totally agree with the Dev team decision , it
> >                 would be impossible for a DEV team to "get it right"
> >                 in all the distros that's why i started this post.
> >                 
> >                 But we also need to take into consideration that we
> >                 are talking about IVR Processing ,  not
> >                 auto-attendants or simply call pass trough. 
> >                 This means heavy use of TTS / ASR , Database Logic
> >                 and Scripts , GetDigits and the works.
> >                 
> >                 I use python alot , but i think the mod_python is
> >                 not the best tool for this job i admit that , that's
> >                 why im currently looking for other solutions.
> >                 I'm thinking in using mod_socket , but that scares
> >                 me ( let? say bad experiences with Asterisk AGI) Or
> >                 Mod Curl
> >                 
> >                 The main problem is that some IVR are extremely
> >                 complex , like questionnaires , etc.
> >                 
> >                 It would be great if we could mod_event_zmq to
> >                 control the calls <330.gif>
> >                 
> >                 Just to Sum it UP so far , so i can later add it to
> >                 the wiki.
> >                 
> >                 Use Cent OS 64.
> >                 Use tmpfs for all the databases.
> >                 
> >                 Thank you all for helping and Happy Eastern.
> >                 Ant?nio Teixeira 
> >                 
> >                 
> >                 
> >                 
> >                 
> >                 
> >                 2011/4/22 Michael Collins <msc at freeswitch.org>
> >                         FreeSWITCH runs well on many platforms.
> >                         However, the devs are painfully aware that
> >                         bleeding edge distros have bleeding edge gcc
> >                         compilers and other interesting issues. That
> >                         being said, CentOS 5.x is "stable" in that
> >                         it's old and boring, therefore it has the
> >                         least amount of drama. OTOH, some of our
> >                         users have been having great success with
> >                         Arch Linux (IRC:bougeyman) and FS, even
> >                         though Arch uses very recent kernels. 
> >                         
> >                         
> >                         Bottom line: if you know what you're doing
> >                         then you can probably run FS anywhere. If
> >                         you don't know what you're doing then stick
> >                         with CentOS 5.x or Debian Lenny until you
> >                         do. (I run then both with zero issues,
> >                         compiling latest git each day.) 
> >                         
> >                         
> >                         -MC 
> >                         
> >                         
> >                         
> >                         On Fri, Apr 22, 2011 at 10:03 AM,
> >                         curriegrad2004 <curriegrad2004 at gmail.com>
> >                         wrote:
> >                                 Freeswitch is targeted for CentOS
> >                                 5.3, which in my opinion quite short
> >                                 sighted for the developers to do
> >                                 this. However with the limited size
> >                                 of developers and testers, I'm
> >                                 afraid there's not much platforms we
> >                                 can throughly test and actually say
> >                                 "okay, FS will run flawlessly on X
> >                                 distro"
> >                                 
> >                                 However you can always try messing
> >                                 with the CFLAG's mtune option and
> >                                 see what it produces for you...
> >                                 
> >                                 2011/4/22 Christian L?schenkohl
> >                                 <christian.loeschenkohl at xpirio.com>: 
> >                                 
> >                                 > hi
> >                                 >
> >                                 > if you refer to my e-mail
> >                                 >
> >                                 > yes, we do use tmpfs on both
> >                                 variants but
> >                                 > - delays occur with concurrent
> >                                 calls > 80-100
> >                                 > - cps is limited to 5-10 on
> >                                 debian, with centos 30 cps and more
> >                                 are no problem at all
> >                                 >
> >                                 > also cpu load, stability and
> >                                 overall performace have been much
> >                                 better since using centos
> >                                 >
> >                                 > i just found out for me that
> >                                 debian works not as good for me as
> >                                 centos does.
> >                                 > btw. everywhere else debian is 1st
> >                                 choice (desktop, lamp, db etc.)
> >                                 >
> >                                 > br
> >                                 >
> >                                 >
> >                                 > On 2011-04-21 23:04, Jay Binks
> >                                 wrote:
> >                                 >
> >                                 >> I have no such problems on
> >                                 debian .
> >                                 >>
> >                                 >> I use debian 5 with 2.6.18 kernel
> >                                 which is what Is recommended
> >                                 >>
> >                                 >> Are you using tmpfs ??
> >                                 >>
> >                                 >> Jay
> >                                 >>
> >                                 >>
> >                                 >>
> >                                 >> On 22/04/2011, at 3:26 AM,
> >                                 Christian
> >                                 L?schenkohl<christian.loeschenkohl at xpirio.com>  wrote:
> >                                 >>
> >                                 >>> hi
> >                                 >>>
> >                                 >>> we did use debian too and had
> >                                 such performance issues (sip packet
> >                                 delays, low cps).
> >                                 >>> after using centos 64bit (as
> >                                 advised by the devs) all performance
> >                                 problems are gone.
> >                                 >>>
> >                                 >>> br
> >                                 >>>
> >                                 >>> On 2011-04-21 18:24, Antonio
> >                                 Teixeira wrote:
> >                                 >>>
> >                                 >>>> Hello List.
> >                                 >>>>
> >                                 >>>> I'm currently integrating an
> >                                 IVR in python together with
> >                                 freeswitch using mod_python and ESL
> >                                 and my life has been well until ...
> >                                 >>>> The flow of calls went over 80
> >                                 simultaneous calls.
> >                                 >>>> Now freeswitch starts sending
> >                                 packets with huge delays ( even when
> >                                 establishing the call , mainly the
> >                                 200 ) and firing up the IVR with
> >                                 tons of delay up to 20 seconds.
> >                                 >>>>
> >                                 >>>> So i searched the wiki forums
> >                                 and mailing list:
> >                                 >>>>
> >                                 >>>> Put freeswitch on a diet ,
> >                                 trimmed modules.conf
> >                                 >>>> Played with the ulimit stuff.
> >                                 >>>> Played with the IVRS to reduce
> >                                 load to a minimum and i was able to
> >                                 squeeze more 5 calls of performance.
> >                                 >>>>
> >                                 >>>> The problem is :
> >                                 >>>>
> >                                 >>>> Top shows
> >                                 >>>> top - 16:14:33 up 35 days,
> >                                  8:15,  3 users,  load average:
> >                                 1.92, 1.76, 1.78
> >                                 >>>> Tasks: 133 total,   1 running,
> >                                 132 sleeping,   0 stopped,   0
> >                                 zombie
> >                                 >>>> Cpu(s):  1.4%us,  3.3%sy,  0.0%
> >                                 ni, 94.6%id,  0.0%wa,  0.3%hi,  0.5%
> >                                 si,  0.0%st
> >                                 >>>> Mem:   8193336k total,
> >                                  1639156k used,  6554180k free,
> >                                 177208k buffers
> >                                 >>>> Swap: 19534904k total,
> >                                  0k used, 19534904k free,  1062272k
> >                                 cached
> >                                 >>>>
> >                                 >>>>    PID USER      PR  NI  VIRT
> >                                  RES  SHR S %CPU %MEM    TIME+
> >                                  COMMAND
> >                                 >>>> 31361 yadayada      20   0
> >                                  716m 164m 9628 S   73  2.1
> >                                 155:17.85 freeswitch
> >                                 >>>>
> >                                 >>>> Freeswitch goes up to 150 % and
> >                                 puff there goes the MOS down to 0.
> >                                 >>>>
> >                                 >>>>
> >                                 >>>> Some basic System Info :
> >                                 >>>> Debian 6.0 ( i heard the
> >                                 timming module is affected by
> >                                 Debian , but if the CPU % gets lower
> >                                 than 95% everything will be more
> >                                 stable)
> >                                 >>>> Python 2.5
> >                                 >>>>
> >                                 >>>> 2 x Intel(R) Xeon(R) CPU
> >                                 E5506  @ 2.13GHz
> >                                 >>>> 8 GB of Ram
> >                                 >>>>
> >                                 >>>> as you can see 94 % of the "Cpu
> >                                 Power" is sleeping :\
> >                                 >>>>
> >                                 >>>>
> >                                 >>>> It appears freeswitch is only
> >                                 capable of using let's say "one
> >                                 cpu"/thread ??
> >                                 >>>> Do you guys recommend simply
> >                                 starting more instances or redoing
> >                                 the IVR stuff.
> >                                 >>>>
> >                                 >>>>
> >                                 >>>> Hope you guys can help me out.
> >                                 >>>>
> >                                 >>>> Thanks
> >                                 >>>> Ant?nio Teixeira
> >                                 >>>>
> >                                 >>>>
> >                                 >>>>
> >                                 >>>>
> >                                 >>>>
> >                                 >>>>
> >                                 >>>>
> >                                 >>>>
> >                                 >>>>
> >                                 >>>>
> >                                 >>>>
> >                                 >>>>
> >                                 >>>>
> >                                 >>>>
> >                                 >>>>
> >                                 _______________________________________________
> >                                 >>>> FreeSWITCH-users mailing list
> >                                 >>>>
> >                                 FreeSWITCH-users at lists.freeswitch.org
> >                                 >>>>
> >                                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >                                 >>>>
> >                                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >                                 >>>> http://www.freeswitch.org
> >                                 >>>
> >                                 >>>
> >                                 >>> --
> >                                 >>> Ing. Christian L?schenkohl
> >                                 >>> Technische Leitung, Forschung&
> >                                  Entwicklung VoIP
> >                                 >>>
> >                                 >>> xpirio
> >                                 >>> Telekommunikation&  Service GmbH
> >                                 >>> Lakeside B04
> >                                 >>> 9020 Klagenfurt
> >                                 >>> Austria
> >                                 >>>
> >                                 >>> T  +43 5 77 11 - 1000
> >                                 >>> F  +43 5 77 11 - 1002
> >                                 >>> E
> >                                  christian.loeschenkohl at xpirio.com
> >                                 >>>
> >                                 >>>
> >                                 _______________________________________________
> >                                 >>> FreeSWITCH-users mailing list
> >                                 >>>
> >                                 FreeSWITCH-users at lists.freeswitch.org
> >                                 >>>
> >                                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >                                 >>>
> >                                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >                                 >>> http://www.freeswitch.org
> >                                 >>
> >                                 >>
> >                                 _______________________________________________
> >                                 >> FreeSWITCH-users mailing list
> >                                 >>
> >                                 FreeSWITCH-users at lists.freeswitch.org
> >                                 >>
> >                                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >                                 >>
> >                                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >                                 >> http://www.freeswitch.org
> >                                 >
> >                                 >
> >                                 > --
> >                                 > Ing. Christian L?schenkohl
> >                                 > Technische Leitung, Forschung &
> >                                 Entwicklung VoIP
> >                                 >
> >                                 > xpirio
> >                                 > Telekommunikation & Service GmbH
> >                                 > Lakeside B04
> >                                 > 9020 Klagenfurt
> >                                 > Austria
> >                                 >
> >                                 > T  +43 5 77 11 - 1000
> >                                 > F  +43 5 77 11 - 1002
> >                                 > E
> >                                  christian.loeschenkohl at xpirio.com
> >                                 >
> >                                 >
> >                                 _______________________________________________
> >                                 > FreeSWITCH-users mailing list
> >                                 >
> >                                 FreeSWITCH-users at lists.freeswitch.org
> >                                 >
> >                                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >                                 >
> >                                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >                                 > http://www.freeswitch.org
> >                                 >
> >                                 
> >                                 _______________________________________________
> >                                 FreeSWITCH-users mailing list
> >                                 FreeSWITCH-users at lists.freeswitch.org
> >                                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >                                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >                                 http://www.freeswitch.org
> >                                 
> >                         
> >                         
> >                         
> >                         _______________________________________________
> >                         FreeSWITCH-users mailing list
> >                         FreeSWITCH-users at lists.freeswitch.org
> >                         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >                         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >                         http://www.freeswitch.org
> >                         
> >                 
> >                 
> >                 
> >                 _______________________________________________
> >                 FreeSWITCH-users mailing list
> >                 FreeSWITCH-users at lists.freeswitch.org
> >                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >                 http://www.freeswitch.org
> >                 
> >         
> >         
> >         
> >         _______________________________________________
> >         FreeSWITCH-users mailing list
> >         FreeSWITCH-users at lists.freeswitch.org
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >         
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> > 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From eagle.antonio at gmail.com  Mon Apr 25 13:05:20 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 25 Apr 2011 10:05:20 +0100
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <1303719841.19225.29.camel@luna.tc.commsmundi.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<4DB068C4.90109@xpirio.com>
	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
	<4DB1550C.1050607@xpirio.com>
	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
	<BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>
	<BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com>
	<BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com>
	<BANLkTin6c0q_66bNk-W+1WjVKkf+0w27FA@mail.gmail.com>
	<B40BD7F2-EEB9-4C1E-AE1F-6474FFFBAE71@gmail.com>
	<1303719841.19225.29.camel@luna.tc.commsmundi.com>
Message-ID: <BANLkTikZQNBEFkPqEG86N8QKTLWFk6yegg@mail.gmail.com>

Good Morning.

@ Steven,
Thanks for the idea i will make a prototype in Python , and will send you
guysthe performance data when i have it.

@ Fran?ois

You probably want this.
http://wiki.freeswitch.org/wiki/XML_Switch_Configuration

'sessions-per-second' - throttling mechanism, the switch will only create
this many channels at most, per second.



2011/4/25 Fran?ois Delawarde <fdelawarde at wirelessmundi.com>

> I also experienced a huge delay issue when attacked by mister
> "friendly-scanner". The machine was not overloaded and call setup was
> working well but with a 10-20 seconds delay. Note that the delay was
> only with SIP answers, and once RTP was setup the audio was perfect.
>
> Following advices from http://wiki.freeswitch.org/wiki/QoS solved it in
> my case. Is there a mechanism in FS that could detect huge call setup
> delay and simply reject calls?
>
> Fran?ois.
>
>
>
> On Sat, 2011-04-23 at 14:46 +0100, Steven Ayre wrote:
> > ESL would be best as you can offload the processing to another server
> > - potentially a load balanced cluster.
> >
> > Steve on iPhone
> >
> > On 23 Apr 2011, at 08:51, Antonio Teixeira <eagle.antonio at gmail.com>
> > wrote:
> >
> >
> >
> > > Hello Michael & The rest.
> > >
> > > What i'm trying to find is the solution with the highest performance
> > > for IVR Applications. And squeezing the last bit of performance out
> > > of freeswitch is a needed path specially on a industry that does
> > > second by second billing :P
> > > i'm going to do some performance testing with Lua , ESL And Mod Curl
> > > and of course python and measure some performance data and will post
> > > it later.
> > >
> > > A/T
> > >
> > >
> > > 2011/4/22 Michael Collins <msc at freeswitch.org>
> > >         You can definitely use the event socket. Heck, you can even
> > >         use Python if you want to. The dev team wrote ESL
> > >         specifically for cases like these - where you want to
> > >         control FS externally. ESL beats the pants off AGI scripts.
> > >         It gives you complete control over the system.
> > >
> > >
> > >         That all being said, you can do all sorts of stuff with Lua
> > >         dialplan scripts. Check out chapter 7 of the FreeSWITCH book
> > >         for some nice examples. Lua is lightweight and easy to
> > >         learn. It's a good alternative for many cases. Check it
> > >         out...
> > >
> > >
> > >         -MC
> > >
> > >
> > >
> > >         On Fri, Apr 22, 2011 at 12:39 PM, Antonio Teixeira
> > >         <eagle.antonio at gmail.com> wrote:
> > >                 Well i totally agree with the Dev team decision , it
> > >                 would be impossible for a DEV team to "get it right"
> > >                 in all the distros that's why i started this post.
> > >
> > >                 But we also need to take into consideration that we
> > >                 are talking about IVR Processing ,  not
> > >                 auto-attendants or simply call pass trough.
> > >                 This means heavy use of TTS / ASR , Database Logic
> > >                 and Scripts , GetDigits and the works.
> > >
> > >                 I use python alot , but i think the mod_python is
> > >                 not the best tool for this job i admit that , that's
> > >                 why im currently looking for other solutions.
> > >                 I'm thinking in using mod_socket , but that scares
> > >                 me ( let? say bad experiences with Asterisk AGI) Or
> > >                 Mod Curl
> > >
> > >                 The main problem is that some IVR are extremely
> > >                 complex , like questionnaires , etc.
> > >
> > >                 It would be great if we could mod_event_zmq to
> > >                 control the calls <330.gif>
> > >
> > >                 Just to Sum it UP so far , so i can later add it to
> > >                 the wiki.
> > >
> > >                 Use Cent OS 64.
> > >                 Use tmpfs for all the databases.
> > >
> > >                 Thank you all for helping and Happy Eastern.
> > >                 Ant?nio Teixeira
> > >
> > >
> > >
> > >
> > >
> > >
> > >                 2011/4/22 Michael Collins <msc at freeswitch.org>
> > >                         FreeSWITCH runs well on many platforms.
> > >                         However, the devs are painfully aware that
> > >                         bleeding edge distros have bleeding edge gcc
> > >                         compilers and other interesting issues. That
> > >                         being said, CentOS 5.x is "stable" in that
> > >                         it's old and boring, therefore it has the
> > >                         least amount of drama. OTOH, some of our
> > >                         users have been having great success with
> > >                         Arch Linux (IRC:bougeyman) and FS, even
> > >                         though Arch uses very recent kernels.
> > >
> > >
> > >                         Bottom line: if you know what you're doing
> > >                         then you can probably run FS anywhere. If
> > >                         you don't know what you're doing then stick
> > >                         with CentOS 5.x or Debian Lenny until you
> > >                         do. (I run then both with zero issues,
> > >                         compiling latest git each day.)
> > >
> > >
> > >                         -MC
> > >
> > >
> > >
> > >                         On Fri, Apr 22, 2011 at 10:03 AM,
> > >                         curriegrad2004 <curriegrad2004 at gmail.com>
> > >                         wrote:
> > >                                 Freeswitch is targeted for CentOS
> > >                                 5.3, which in my opinion quite short
> > >                                 sighted for the developers to do
> > >                                 this. However with the limited size
> > >                                 of developers and testers, I'm
> > >                                 afraid there's not much platforms we
> > >                                 can throughly test and actually say
> > >                                 "okay, FS will run flawlessly on X
> > >                                 distro"
> > >
> > >                                 However you can always try messing
> > >                                 with the CFLAG's mtune option and
> > >                                 see what it produces for you...
> > >
> > >                                 2011/4/22 Christian L?schenkohl
> > >                                 <christian.loeschenkohl at xpirio.com>:
> > >
> > >                                 > hi
> > >                                 >
> > >                                 > if you refer to my e-mail
> > >                                 >
> > >                                 > yes, we do use tmpfs on both
> > >                                 variants but
> > >                                 > - delays occur with concurrent
> > >                                 calls > 80-100
> > >                                 > - cps is limited to 5-10 on
> > >                                 debian, with centos 30 cps and more
> > >                                 are no problem at all
> > >                                 >
> > >                                 > also cpu load, stability and
> > >                                 overall performace have been much
> > >                                 better since using centos
> > >                                 >
> > >                                 > i just found out for me that
> > >                                 debian works not as good for me as
> > >                                 centos does.
> > >                                 > btw. everywhere else debian is 1st
> > >                                 choice (desktop, lamp, db etc.)
> > >                                 >
> > >                                 > br
> > >                                 >
> > >                                 >
> > >                                 > On 2011-04-21 23:04, Jay Binks
> > >                                 wrote:
> > >                                 >
> > >                                 >> I have no such problems on
> > >                                 debian .
> > >                                 >>
> > >                                 >> I use debian 5 with 2.6.18 kernel
> > >                                 which is what Is recommended
> > >                                 >>
> > >                                 >> Are you using tmpfs ??
> > >                                 >>
> > >                                 >> Jay
> > >                                 >>
> > >                                 >>
> > >                                 >>
> > >                                 >> On 22/04/2011, at 3:26 AM,
> > >                                 Christian
> > >                                 L?schenkohl<
> christian.loeschenkohl at xpirio.com>  wrote:
> > >                                 >>
> > >                                 >>> hi
> > >                                 >>>
> > >                                 >>> we did use debian too and had
> > >                                 such performance issues (sip packet
> > >                                 delays, low cps).
> > >                                 >>> after using centos 64bit (as
> > >                                 advised by the devs) all performance
> > >                                 problems are gone.
> > >                                 >>>
> > >                                 >>> br
> > >                                 >>>
> > >                                 >>> On 2011-04-21 18:24, Antonio
> > >                                 Teixeira wrote:
> > >                                 >>>
> > >                                 >>>> Hello List.
> > >                                 >>>>
> > >                                 >>>> I'm currently integrating an
> > >                                 IVR in python together with
> > >                                 freeswitch using mod_python and ESL
> > >                                 and my life has been well until ...
> > >                                 >>>> The flow of calls went over 80
> > >                                 simultaneous calls.
> > >                                 >>>> Now freeswitch starts sending
> > >                                 packets with huge delays ( even when
> > >                                 establishing the call , mainly the
> > >                                 200 ) and firing up the IVR with
> > >                                 tons of delay up to 20 seconds.
> > >                                 >>>>
> > >                                 >>>> So i searched the wiki forums
> > >                                 and mailing list:
> > >                                 >>>>
> > >                                 >>>> Put freeswitch on a diet ,
> > >                                 trimmed modules.conf
> > >                                 >>>> Played with the ulimit stuff.
> > >                                 >>>> Played with the IVRS to reduce
> > >                                 load to a minimum and i was able to
> > >                                 squeeze more 5 calls of performance.
> > >                                 >>>>
> > >                                 >>>> The problem is :
> > >                                 >>>>
> > >                                 >>>> Top shows
> > >                                 >>>> top - 16:14:33 up 35 days,
> > >                                  8:15,  3 users,  load average:
> > >                                 1.92, 1.76, 1.78
> > >                                 >>>> Tasks: 133 total,   1 running,
> > >                                 132 sleeping,   0 stopped,   0
> > >                                 zombie
> > >                                 >>>> Cpu(s):  1.4%us,  3.3%sy,  0.0%
> > >                                 ni, 94.6%id,  0.0%wa,  0.3%hi,  0.5%
> > >                                 si,  0.0%st
> > >                                 >>>> Mem:   8193336k total,
> > >                                  1639156k used,  6554180k free,
> > >                                 177208k buffers
> > >                                 >>>> Swap: 19534904k total,
> > >                                  0k used, 19534904k free,  1062272k
> > >                                 cached
> > >                                 >>>>
> > >                                 >>>>    PID USER      PR  NI  VIRT
> > >                                  RES  SHR S %CPU %MEM    TIME+
> > >                                  COMMAND
> > >                                 >>>> 31361 yadayada      20   0
> > >                                  716m 164m 9628 S   73  2.1
> > >                                 155:17.85 freeswitch
> > >                                 >>>>
> > >                                 >>>> Freeswitch goes up to 150 % and
> > >                                 puff there goes the MOS down to 0.
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>> Some basic System Info :
> > >                                 >>>> Debian 6.0 ( i heard the
> > >                                 timming module is affected by
> > >                                 Debian , but if the CPU % gets lower
> > >                                 than 95% everything will be more
> > >                                 stable)
> > >                                 >>>> Python 2.5
> > >                                 >>>>
> > >                                 >>>> 2 x Intel(R) Xeon(R) CPU
> > >                                 E5506  @ 2.13GHz
> > >                                 >>>> 8 GB of Ram
> > >                                 >>>>
> > >                                 >>>> as you can see 94 % of the "Cpu
> > >                                 Power" is sleeping :\
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>> It appears freeswitch is only
> > >                                 capable of using let's say "one
> > >                                 cpu"/thread ??
> > >                                 >>>> Do you guys recommend simply
> > >                                 starting more instances or redoing
> > >                                 the IVR stuff.
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>> Hope you guys can help me out.
> > >                                 >>>>
> > >                                 >>>> Thanks
> > >                                 >>>> Ant?nio Teixeira
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>>
> > >                                 >>>>
> > >
> _______________________________________________
> > >                                 >>>> FreeSWITCH-users mailing list
> > >                                 >>>>
> > >                                 FreeSWITCH-users at lists.freeswitch.org
> > >                                 >>>>
> > >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >                                 >>>>
> > >                                 UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >                                 >>>> http://www.freeswitch.org
> > >                                 >>>
> > >                                 >>>
> > >                                 >>> --
> > >                                 >>> Ing. Christian L?schenkohl
> > >                                 >>> Technische Leitung, Forschung&
> > >                                  Entwicklung VoIP
> > >                                 >>>
> > >                                 >>> xpirio
> > >                                 >>> Telekommunikation&  Service GmbH
> > >                                 >>> Lakeside B04
> > >                                 >>> 9020 Klagenfurt
> > >                                 >>> Austria
> > >                                 >>>
> > >                                 >>> T  +43 5 77 11 - 1000
> > >                                 >>> F  +43 5 77 11 - 1002
> > >                                 >>> E
> > >                                  christian.loeschenkohl at xpirio.com
> > >                                 >>>
> > >                                 >>>
> > >
> _______________________________________________
> > >                                 >>> FreeSWITCH-users mailing list
> > >                                 >>>
> > >                                 FreeSWITCH-users at lists.freeswitch.org
> > >                                 >>>
> > >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >                                 >>>
> > >                                 UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >                                 >>> http://www.freeswitch.org
> > >                                 >>
> > >                                 >>
> > >
> _______________________________________________
> > >                                 >> FreeSWITCH-users mailing list
> > >                                 >>
> > >                                 FreeSWITCH-users at lists.freeswitch.org
> > >                                 >>
> > >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >                                 >>
> > >                                 UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >                                 >> http://www.freeswitch.org
> > >                                 >
> > >                                 >
> > >                                 > --
> > >                                 > Ing. Christian L?schenkohl
> > >                                 > Technische Leitung, Forschung &
> > >                                 Entwicklung VoIP
> > >                                 >
> > >                                 > xpirio
> > >                                 > Telekommunikation & Service GmbH
> > >                                 > Lakeside B04
> > >                                 > 9020 Klagenfurt
> > >                                 > Austria
> > >                                 >
> > >                                 > T  +43 5 77 11 - 1000
> > >                                 > F  +43 5 77 11 - 1002
> > >                                 > E
> > >                                  christian.loeschenkohl at xpirio.com
> > >                                 >
> > >                                 >
> > >
> _______________________________________________
> > >                                 > FreeSWITCH-users mailing list
> > >                                 >
> > >                                 FreeSWITCH-users at lists.freeswitch.org
> > >                                 >
> > >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >                                 >
> > >                                 UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >                                 > http://www.freeswitch.org
> > >                                 >
> > >
> > >
> _______________________________________________
> > >                                 FreeSWITCH-users mailing list
> > >                                 FreeSWITCH-users at lists.freeswitch.org
> > >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >                                 UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >                                 http://www.freeswitch.org
> > >
> > >
> > >
> > >
> > >                         _______________________________________________
> > >                         FreeSWITCH-users mailing list
> > >                         FreeSWITCH-users at lists.freeswitch.org
> > >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >                         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >                         http://www.freeswitch.org
> > >
> > >
> > >
> > >
> > >                 _______________________________________________
> > >                 FreeSWITCH-users mailing list
> > >                 FreeSWITCH-users at lists.freeswitch.org
> > >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >                 UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >                 http://www.freeswitch.org
> > >
> > >
> > >
> > >
> > >         _______________________________________________
> > >         FreeSWITCH-users mailing list
> > >         FreeSWITCH-users at lists.freeswitch.org
> > >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > >         http://www.freeswitch.org
> > >
> > >
> > >
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
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>
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From peter.olsson at visionutveckling.se  Mon Apr 25 13:23:00 2011
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Mon, 25 Apr 2011 11:23:00 +0200
Subject: [Freeswitch-users] Build and install FS with new customized
 folder	on CentOS 5.5
In-Reply-To: <BANLkTinOFchbfvGW7cDOv=f5S0goFWcmig@mail.gmail.com>
References: <BANLkTinOFchbfvGW7cDOv=f5S0goFWcmig@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C58F127618C@cooper>

After bootstrap - ./configure --prefix=/your/path
________________________________________
Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r fieldpeak [fieldpeak at gmail.com]
Skickat: den 25 april 2011 10:15
Till: FreeSWITCH-users
?mne: [Freeswitch-users] Build and install FS with new customized folder        on CentOS 5.5

i added a folders(e.g. new_folder) inside the FS source code folder, inside the new_folder there are some files, like below,

./freeswitch/conf
./freeswitch/bin
./freeswitch/new_folder/files1, files2...

i need this folder and the files under it will be installed when i run 'make install', however, it did not.

the commands i run as below, i assume bootstrap.sh will run autoscan which will find my new_folder...
./bootstrap.sh
make
make install

anyone can help how to realize it? thanks.

Regards,
Charles



!DSPAM:4db52eaf32762004716643!


From fieldpeak at gmail.com  Mon Apr 25 13:33:24 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 25 Apr 2011 17:33:24 +0800
Subject: [Freeswitch-users] Build and install FS with new customized
 folder on CentOS 5.5
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C58F127618C@cooper>
References: <BANLkTinOFchbfvGW7cDOv=f5S0goFWcmig@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F127618C@cooper>
Message-ID: <BANLkTik_t6AkULxctsYW85SLCjrXL0wO_g@mail.gmail.com>

as i know, ./configure --prefix=/your/path, the prefix will define the new
installation path to override the default one, however, my question is that
i added a new folder inside the source code, why it is not install to the
installation path when i excute 'make install', sorry, your answer seemed
not answer the question...thanks.

2011/4/25 Peter Olsson <peter.olsson at visionutveckling.se>

> After bootstrap - ./configure --prefix=/your/path
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r fieldpeak [
> fieldpeak at gmail.com]
> Skickat: den 25 april 2011 10:15
> Till: FreeSWITCH-users
> ?mne: [Freeswitch-users] Build and install FS with new customized folder
>      on CentOS 5.5
>
> i added a folders(e.g. new_folder) inside the FS source code folder, inside
> the new_folder there are some files, like below,
>
> ./freeswitch/conf
> ./freeswitch/bin
> ./freeswitch/new_folder/files1, files2...
>
> i need this folder and the files under it will be installed when i run
> 'make install', however, it did not.
>
> the commands i run as below, i assume bootstrap.sh will run autoscan which
> will find my new_folder...
> ./bootstrap.sh
> make
> make install
>
> anyone can help how to realize it? thanks.
>
> Regards,
> Charles
>
>
>
> !DSPAM:4db52eaf32762004716643!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 25 13:40:01 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 25 Apr 2011 10:40:01 +0100
Subject: [Freeswitch-users] Build and install FS with new customized
	folder on CentOS 5.5
In-Reply-To: <BANLkTik_t6AkULxctsYW85SLCjrXL0wO_g@mail.gmail.com>
References: <BANLkTinOFchbfvGW7cDOv=f5S0goFWcmig@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F127618C@cooper>
	<BANLkTik_t6AkULxctsYW85SLCjrXL0wO_g@mail.gmail.com>
Message-ID: <D0B8A80F-EF21-42F8-9A30-76BB25D5DE0A@gmail.com>

Because the makefile is more complex than that...

What type of directory? A new module? You need to add the module to modules.conf

Steve on iPhone


On 25 Apr 2011, at 10:33, fieldpeak <fieldpeak at gmail.com> wrote:

> as i know, ./configure --prefix=/your/path, the prefix will define the new installation path to override the default one, however, my question is that i added a new folder inside the source code, why it is not install to the installation path when i excute 'make install', sorry, your answer seemed not answer the question...thanks.
> 
> 2011/4/25 Peter Olsson <peter.olsson at visionutveckling.se>
> After bootstrap - ./configure --prefix=/your/path
> ________________________________________
> Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f&#246;r fieldpeak [fieldpeak at gmail.com]
> Skickat: den 25 april 2011 10:15
> Till: FreeSWITCH-users
> ?mne: [Freeswitch-users] Build and install FS with new customized folder        on CentOS 5.5
> 
> i added a folders(e.g. new_folder) inside the FS source code folder, inside the new_folder there are some files, like below,
> 
> ./freeswitch/conf
> ./freeswitch/bin
> ./freeswitch/new_folder/files1, files2...
> 
> i need this folder and the files under it will be installed when i run 'make install', however, it did not.
> 
> the commands i run as below, i assume bootstrap.sh will run autoscan which will find my new_folder...
> ./bootstrap.sh
> make
> make install
> 
> anyone can help how to realize it? thanks.
> 
> Regards,
> Charles
> 
> 
> 
> !DSPAM:4db52eaf32762004716643!
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Mon Apr 25 13:48:37 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Mon, 25 Apr 2011 11:48:37 +0200
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTikZQNBEFkPqEG86N8QKTLWFk6yegg@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<4DB068C4.90109@xpirio.com>
	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
	<4DB1550C.1050607@xpirio.com>
	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
	<BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>
	<BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com>
	<BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com>
	<BANLkTin6c0q_66bNk-W+1WjVKkf+0w27FA@mail.gmail.com>
	<B40BD7F2-EEB9-4C1E-AE1F-6474FFFBAE71@gmail.com>
	<1303719841.19225.29.camel@luna.tc.commsmundi.com>
	<BANLkTikZQNBEFkPqEG86N8QKTLWFk6yegg@mail.gmail.com>
Message-ID: <1303724917.8952.10.camel@luna.tc.commsmundi.com>

Thanks Antonio, I'm already using that along with iptables rate
limiting. It fits my needs but I need to run tests every time I install
FS in a new hardware to find the optimal values.

It's good enough for me, but I was just wondering if there was anything
more automatic like some timer that could drop the call if it takes too
long to process.

It could also be useful for stalled DB connections or similar endless
loops in some badly coded IVR applications.

Fran?ois.


On Mon, 2011-04-25 at 10:05 +0100, Antonio Teixeira wrote:
> Good Morning.
> 
> @ Steven,
> Thanks for the idea i will make a prototype in Python , and will send
> you guysthe performance data when i have it.
> 
> @ Fran?ois
> 
> You probably want this.
> http://wiki.freeswitch.org/wiki/XML_Switch_Configuration
> 
> 'sessions-per-second' - throttling mechanism, the switch will only
> create this many channels at most, per second. 
> 
> 
> 
> 2011/4/25 Fran?ois Delawarde <fdelawarde at wirelessmundi.com>
>         I also experienced a huge delay issue when attacked by mister
>         "friendly-scanner". The machine was not overloaded and call
>         setup was
>         working well but with a 10-20 seconds delay. Note that the
>         delay was
>         only with SIP answers, and once RTP was setup the audio was
>         perfect.
>         
>         Following advices from http://wiki.freeswitch.org/wiki/QoS
>         solved it in
>         my case. Is there a mechanism in FS that could detect huge
>         call setup
>         delay and simply reject calls?
>         
>         Fran?ois. 
>         
>         
>         
>         
>         On Sat, 2011-04-23 at 14:46 +0100, Steven Ayre wrote:
>         > ESL would be best as you can offload the processing to
>         another server
>         > - potentially a load balanced cluster.
>         >
>         > Steve on iPhone
>         >
>         > On 23 Apr 2011, at 08:51, Antonio Teixeira
>         <eagle.antonio at gmail.com>
>         > wrote:
>         >
>         >
>         >
>         > > Hello Michael & The rest.
>         > >
>         > > What i'm trying to find is the solution with the highest
>         performance
>         > > for IVR Applications. And squeezing the last bit of
>         performance out
>         > > of freeswitch is a needed path specially on a industry
>         that does
>         > > second by second billing :P
>         > > i'm going to do some performance testing with Lua , ESL
>         And Mod Curl
>         > > and of course python and measure some performance data and
>         will post
>         > > it later.
>         > >
>         > > A/T
>         > >
>         > >
>         > > 2011/4/22 Michael Collins <msc at freeswitch.org>
>         > >         You can definitely use the event socket. Heck, you
>         can even
>         > >         use Python if you want to. The dev team wrote ESL
>         > >         specifically for cases like these - where you want
>         to
>         > >         control FS externally. ESL beats the pants off AGI
>         scripts.
>         > >         It gives you complete control over the system.
>         > >
>         > >
>         > >         That all being said, you can do all sorts of stuff
>         with Lua
>         > >         dialplan scripts. Check out chapter 7 of the
>         FreeSWITCH book
>         > >         for some nice examples. Lua is lightweight and
>         easy to
>         > >         learn. It's a good alternative for many cases.
>         Check it
>         > >         out...
>         > >
>         > >
>         > >         -MC
>         > >
>         > >
>         > >
>         > >         On Fri, Apr 22, 2011 at 12:39 PM, Antonio Teixeira
>         > >         <eagle.antonio at gmail.com> wrote:
>         > >                 Well i totally agree with the Dev team
>         decision , it
>         > >                 would be impossible for a DEV team to "get
>         it right"
>         > >                 in all the distros that's why i started
>         this post.
>         > >
>         > >                 But we also need to take into
>         consideration that we
>         > >                 are talking about IVR Processing ,  not
>         > >                 auto-attendants or simply call pass
>         trough.
>         > >                 This means heavy use of TTS / ASR ,
>         Database Logic
>         > >                 and Scripts , GetDigits and the works.
>         > >
>         > >                 I use python alot , but i think the
>         mod_python is
>         > >                 not the best tool for this job i admit
>         that , that's
>         > >                 why im currently looking for other
>         solutions.
>         > >                 I'm thinking in using mod_socket , but
>         that scares
>         > >                 me ( let? say bad experiences with
>         Asterisk AGI) Or
>         > >                 Mod Curl
>         > >
>         > >                 The main problem is that some IVR are
>         extremely
>         > >                 complex , like questionnaires , etc.
>         > >
>         > >                 It would be great if we could
>         mod_event_zmq to
>         > >                 control the calls <330.gif>
>         > >
>         > >                 Just to Sum it UP so far , so i can later
>         add it to
>         > >                 the wiki.
>         > >
>         > >                 Use Cent OS 64.
>         > >                 Use tmpfs for all the databases.
>         > >
>         > >                 Thank you all for helping and Happy
>         Eastern.
>         > >                 Ant?nio Teixeira
>         > >
>         > >
>         > >
>         > >
>         > >
>         > >
>         > >                 2011/4/22 Michael Collins
>         <msc at freeswitch.org>
>         > >                         FreeSWITCH runs well on many
>         platforms.
>         > >                         However, the devs are painfully
>         aware that
>         > >                         bleeding edge distros have
>         bleeding edge gcc
>         > >                         compilers and other interesting
>         issues. That
>         > >                         being said, CentOS 5.x is "stable"
>         in that
>         > >                         it's old and boring, therefore it
>         has the
>         > >                         least amount of drama. OTOH, some
>         of our
>         > >                         users have been having great
>         success with
>         > >                         Arch Linux (IRC:bougeyman) and FS,
>         even
>         > >                         though Arch uses very recent
>         kernels.
>         > >
>         > >
>         > >                         Bottom line: if you know what
>         you're doing
>         > >                         then you can probably run FS
>         anywhere. If
>         > >                         you don't know what you're doing
>         then stick
>         > >                         with CentOS 5.x or Debian Lenny
>         until you
>         > >                         do. (I run then both with zero
>         issues,
>         > >                         compiling latest git each day.)
>         > >
>         > >
>         > >                         -MC
>         > >
>         > >
>         > >
>         > >                         On Fri, Apr 22, 2011 at 10:03 AM,
>         > >                         curriegrad2004
>         <curriegrad2004 at gmail.com>
>         > >                         wrote:
>         > >                                 Freeswitch is targeted for
>         CentOS
>         > >                                 5.3, which in my opinion
>         quite short
>         > >                                 sighted for the developers
>         to do
>         > >                                 this. However with the
>         limited size
>         > >                                 of developers and testers,
>         I'm
>         > >                                 afraid there's not much
>         platforms we
>         > >                                 can throughly test and
>         actually say
>         > >                                 "okay, FS will run
>         flawlessly on X
>         > >                                 distro"
>         > >
>         > >                                 However you can always try
>         messing
>         > >                                 with the CFLAG's mtune
>         option and
>         > >                                 see what it produces for
>         you...
>         > >
>         > >                                 2011/4/22 Christian
>         L?schenkohl
>         > >
>         <christian.loeschenkohl at xpirio.com>:
>         > >
>         > >                                 > hi
>         > >                                 >
>         > >                                 > if you refer to my
>         e-mail
>         > >                                 >
>         > >                                 > yes, we do use tmpfs on
>         both
>         > >                                 variants but
>         > >                                 > - delays occur with
>         concurrent
>         > >                                 calls > 80-100
>         > >                                 > - cps is limited to 5-10
>         on
>         > >                                 debian, with centos 30 cps
>         and more
>         > >                                 are no problem at all
>         > >                                 >
>         > >                                 > also cpu load, stability
>         and
>         > >                                 overall performace have
>         been much
>         > >                                 better since using centos
>         > >                                 >
>         > >                                 > i just found out for me
>         that
>         > >                                 debian works not as good
>         for me as
>         > >                                 centos does.
>         > >                                 > btw. everywhere else
>         debian is 1st
>         > >                                 choice (desktop, lamp, db
>         etc.)
>         > >                                 >
>         > >                                 > br
>         > >                                 >
>         > >                                 >
>         > >                                 > On 2011-04-21 23:04, Jay
>         Binks
>         > >                                 wrote:
>         > >                                 >
>         > >                                 >> I have no such problems
>         on
>         > >                                 debian .
>         > >                                 >>
>         > >                                 >> I use debian 5 with
>         2.6.18 kernel
>         > >                                 which is what Is
>         recommended
>         > >                                 >>
>         > >                                 >> Are you using tmpfs ??
>         > >                                 >>
>         > >                                 >> Jay
>         > >                                 >>
>         > >                                 >>
>         > >                                 >>
>         > >                                 >> On 22/04/2011, at 3:26
>         AM,
>         > >                                 Christian
>         > >
>         L?schenkohl<christian.loeschenkohl at xpirio.com>  wrote:
>         > >                                 >>
>         > >                                 >>> hi
>         > >                                 >>>
>         > >                                 >>> we did use debian too
>         and had
>         > >                                 such performance issues
>         (sip packet
>         > >                                 delays, low cps).
>         > >                                 >>> after using centos
>         64bit (as
>         > >                                 advised by the devs) all
>         performance
>         > >                                 problems are gone.
>         > >                                 >>>
>         > >                                 >>> br
>         > >                                 >>>
>         > >                                 >>> On 2011-04-21 18:24,
>         Antonio
>         > >                                 Teixeira wrote:
>         > >                                 >>>
>         > >                                 >>>> Hello List.
>         > >                                 >>>>
>         > >                                 >>>> I'm currently
>         integrating an
>         > >                                 IVR in python together
>         with
>         > >                                 freeswitch using
>         mod_python and ESL
>         > >                                 and my life has been well
>         until ...
>         > >                                 >>>> The flow of calls
>         went over 80
>         > >                                 simultaneous calls.
>         > >                                 >>>> Now freeswitch starts
>         sending
>         > >                                 packets with huge delays
>         ( even when
>         > >                                 establishing the call ,
>         mainly the
>         > >                                 200 ) and firing up the
>         IVR with
>         > >                                 tons of delay up to 20
>         seconds.
>         > >                                 >>>>
>         > >                                 >>>> So i searched the
>         wiki forums
>         > >                                 and mailing list:
>         > >                                 >>>>
>         > >                                 >>>> Put freeswitch on a
>         diet ,
>         > >                                 trimmed modules.conf
>         > >                                 >>>> Played with the
>         ulimit stuff.
>         > >                                 >>>> Played with the IVRS
>         to reduce
>         > >                                 load to a minimum and i
>         was able to
>         > >                                 squeeze more 5 calls of
>         performance.
>         > >                                 >>>>
>         > >                                 >>>> The problem is :
>         > >                                 >>>>
>         > >                                 >>>> Top shows
>         > >                                 >>>> top - 16:14:33 up 35
>         days,
>         > >                                  8:15,  3 users,  load
>         average:
>         > >                                 1.92, 1.76, 1.78
>         > >                                 >>>> Tasks: 133 total,   1
>         running,
>         > >                                 132 sleeping,   0 stopped,
>         0
>         > >                                 zombie
>         > >                                 >>>> Cpu(s):  1.4%us,
>          3.3%sy,  0.0%
>         > >                                 ni, 94.6%id,  0.0%wa,
>          0.3%hi,  0.5%
>         > >                                 si,  0.0%st
>         > >                                 >>>> Mem:   8193336k
>         total,
>         > >                                  1639156k used,  6554180k
>         free,
>         > >                                 177208k buffers
>         > >                                 >>>> Swap: 19534904k
>         total,
>         > >                                  0k used, 19534904k free,
>          1062272k
>         > >                                 cached
>         > >                                 >>>>
>         > >                                 >>>>    PID USER      PR
>          NI  VIRT
>         > >                                  RES  SHR S %CPU %MEM
>          TIME+
>         > >                                  COMMAND
>         > >                                 >>>> 31361 yadayada
>          20   0
>         > >                                  716m 164m 9628 S   73
>          2.1
>         > >                                 155:17.85 freeswitch
>         > >                                 >>>>
>         > >                                 >>>> Freeswitch goes up to
>         150 % and
>         > >                                 puff there goes the MOS
>         down to 0.
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>> Some basic System
>         Info :
>         > >                                 >>>> Debian 6.0 ( i heard
>         the
>         > >                                 timming module is affected
>         by
>         > >                                 Debian , but if the CPU %
>         gets lower
>         > >                                 than 95% everything will
>         be more
>         > >                                 stable)
>         > >                                 >>>> Python 2.5
>         > >                                 >>>>
>         > >                                 >>>> 2 x Intel(R) Xeon(R)
>         CPU
>         > >                                 E5506  @ 2.13GHz
>         > >                                 >>>> 8 GB of Ram
>         > >                                 >>>>
>         > >                                 >>>> as you can see 94 %
>         of the "Cpu
>         > >                                 Power" is sleeping :\
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>> It appears freeswitch
>         is only
>         > >                                 capable of using let's say
>         "one
>         > >                                 cpu"/thread ??
>         > >                                 >>>> Do you guys recommend
>         simply
>         > >                                 starting more instances or
>         redoing
>         > >                                 the IVR stuff.
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>> Hope you guys can
>         help me out.
>         > >                                 >>>>
>         > >                                 >>>> Thanks
>         > >                                 >>>> Ant?nio Teixeira
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>>
>         > >                                 >>>>
>         > >
>         _______________________________________________
>         > >                                 >>>> FreeSWITCH-users
>         mailing list
>         > >                                 >>>>
>         > >
>         FreeSWITCH-users at lists.freeswitch.org
>         > >                                 >>>>
>         > >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         > >                                 >>>>
>         > >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > >                                 >>>>
>         http://www.freeswitch.org
>         > >                                 >>>
>         > >                                 >>>
>         > >                                 >>> --
>         > >                                 >>> Ing. Christian
>         L?schenkohl
>         > >                                 >>> Technische Leitung,
>         Forschung&
>         > >                                  Entwicklung VoIP
>         > >                                 >>>
>         > >                                 >>> xpirio
>         > >                                 >>> Telekommunikation&
>          Service GmbH
>         > >                                 >>> Lakeside B04
>         > >                                 >>> 9020 Klagenfurt
>         > >                                 >>> Austria
>         > >                                 >>>
>         > >                                 >>> T  +43 5 77 11 - 1000
>         > >                                 >>> F  +43 5 77 11 - 1002
>         > >                                 >>> E
>         > >
>          christian.loeschenkohl at xpirio.com
>         > >                                 >>>
>         > >                                 >>>
>         > >
>         _______________________________________________
>         > >                                 >>> FreeSWITCH-users
>         mailing list
>         > >                                 >>>
>         > >
>         FreeSWITCH-users at lists.freeswitch.org
>         > >                                 >>>
>         > >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         > >                                 >>>
>         > >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > >                                 >>>
>         http://www.freeswitch.org
>         > >                                 >>
>         > >                                 >>
>         > >
>         _______________________________________________
>         > >                                 >> FreeSWITCH-users
>         mailing list
>         > >                                 >>
>         > >
>         FreeSWITCH-users at lists.freeswitch.org
>         > >                                 >>
>         > >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         > >                                 >>
>         > >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > >                                 >>
>         http://www.freeswitch.org
>         > >                                 >
>         > >                                 >
>         > >                                 > --
>         > >                                 > Ing. Christian
>         L?schenkohl
>         > >                                 > Technische Leitung,
>         Forschung &
>         > >                                 Entwicklung VoIP
>         > >                                 >
>         > >                                 > xpirio
>         > >                                 > Telekommunikation &
>         Service GmbH
>         > >                                 > Lakeside B04
>         > >                                 > 9020 Klagenfurt
>         > >                                 > Austria
>         > >                                 >
>         > >                                 > T  +43 5 77 11 - 1000
>         > >                                 > F  +43 5 77 11 - 1002
>         > >                                 > E
>         > >
>          christian.loeschenkohl at xpirio.com
>         > >                                 >
>         > >                                 >
>         > >
>         _______________________________________________
>         > >                                 > FreeSWITCH-users mailing
>         list
>         > >                                 >
>         > >
>         FreeSWITCH-users at lists.freeswitch.org
>         > >                                 >
>         > >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         > >                                 >
>         > >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > >                                 >
>         http://www.freeswitch.org
>         > >                                 >
>         > >
>         > >
>         _______________________________________________
>         > >                                 FreeSWITCH-users mailing
>         list
>         > >
>         FreeSWITCH-users at lists.freeswitch.org
>         > >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         > >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > >                                 http://www.freeswitch.org
>         > >
>         > >
>         > >
>         > >
>         > >
>         _______________________________________________
>         > >                         FreeSWITCH-users mailing list
>         > >
>         FreeSWITCH-users at lists.freeswitch.org
>         > >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         > >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > >                         http://www.freeswitch.org
>         > >
>         > >
>         > >
>         > >
>         > >
>         _______________________________________________
>         > >                 FreeSWITCH-users mailing list
>         > >                 FreeSWITCH-users at lists.freeswitch.org
>         > >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         > >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > >                 http://www.freeswitch.org
>         > >
>         > >
>         > >
>         > >
>         > >         _______________________________________________
>         > >         FreeSWITCH-users mailing list
>         > >         FreeSWITCH-users at lists.freeswitch.org
>         > >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         > >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > >         http://www.freeswitch.org
>         > >
>         > >
>         > >
>         > > _______________________________________________
>         > > FreeSWITCH-users mailing list
>         > > FreeSWITCH-users at lists.freeswitch.org
>         > >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         > >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > > http://www.freeswitch.org
>         > >
>         > _______________________________________________
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>         
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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  Mon Apr 25 14:58:41 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Mon, 25 Apr 2011 18:58:41 +0800
Subject: [Freeswitch-users] Build and install FS with new customized
 folder on CentOS 5.5
In-Reply-To: <D0B8A80F-EF21-42F8-9A30-76BB25D5DE0A@gmail.com>
References: <BANLkTinOFchbfvGW7cDOv=f5S0goFWcmig@mail.gmail.com>
	<549CFEF87AEDE841A38E9D15EAB4C04C58F127618C@cooper>
	<BANLkTik_t6AkULxctsYW85SLCjrXL0wO_g@mail.gmail.com>
	<D0B8A80F-EF21-42F8-9A30-76BB25D5DE0A@gmail.com>
Message-ID: <BANLkTimGkwUtVS0na5218ZG1qSaLm2w2SQ@mail.gmail.com>

www folder, i want to put all of the web pages there (same folder as FS)
instead of /var/www/html
should i modify the configure.in or Makefile.am? thanks.

Regards,
Charles

2011/4/25 Steven Ayre <steveayre at gmail.com>

> Because the makefile is more complex than that...
>
> What type of directory? A new module? You need to add the module to
> modules.conf
>
> Steve on iPhone
>
>
> On 25 Apr 2011, at 10:33, fieldpeak <fieldpeak at gmail.com> wrote:
>
> as i know, ./configure --prefix=/your/path, the prefix will define the new
> installation path to override the default one, however, my question is that
> i added a new folder inside the source code, why it is not install to the
> installation path when i excute 'make install', sorry, your answer seemed
> not answer the question...thanks.
>
> 2011/4/25 Peter Olsson < <peter.olsson at visionutveckling.se>
> peter.olsson at visionutveckling.se>
>
>> After bootstrap - ./configure --prefix=/your/path
>> ________________________________________
>> Fr?n: <freeswitch-users-bounces at lists.freeswitch.org>
>> freeswitch-users-bounces at lists.freeswitch.org [<freeswitch-users-bounces at lists.freeswitch.org>
>> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r fieldpeak [<fieldpeak at gmail.com>
>> fieldpeak at gmail.com]
>> Skickat: den 25 april 2011 10:15
>> Till: FreeSWITCH-users
>> ?mne: [Freeswitch-users] Build and install FS with new customized folder
>>      on CentOS 5.5
>>
>> i added a folders(e.g. new_folder) inside the FS source code folder,
>> inside the new_folder there are some files, like below,
>>
>> ./freeswitch/conf
>> ./freeswitch/bin
>> ./freeswitch/new_folder/files1, files2...
>>
>> i need this folder and the files under it will be installed when i run
>> 'make install', however, it did not.
>>
>> the commands i run as below, i assume bootstrap.sh will run autoscan which
>> will find my new_folder...
>> ./bootstrap.sh
>> make
>> make install
>>
>> anyone can help how to realize it? thanks.
>>
>> Regards,
>> Charles
>>
>>
>>
>> !DSPAM:4db52eaf32762004716643!
>>
>> _______________________________________________
>> 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 jgallartm at gmail.com  Mon Apr 25 15:44:34 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Mon, 25 Apr 2011 13:44:34 +0200
Subject: [Freeswitch-users] g729 annexa/annexb interoperability
Message-ID: <BANLkTim6q2QFwESamPZ4v6k_OUaAX+MbhQ@mail.gmail.com>

Hello all

we're using a Sangoma D100 card for transcoding. Our configuration, as far
as the codec policy is concerned, basically does this:

-In the first offer to the b-leg we use the same codec list we receive
(<action application="export"
data="absolute_codec_string=${ep_codec_string}"/>)
-If the call fails with status 488, we repeat the call using all the codecs
available. This includes changing from annexb to annexa and viceversa.

Let's suppose a-leg only supports g729 annexA, and B-leg only supports G729
annexB. My understanding is that those different codec flavours don't
interoperate; in fact I've experienced many times audio problems when trying
to set up a call between endpoints supporting different g729 variants.

In the described case, FS sends initially annexb=no, and B-leg rejects it
with cause 488. We rebuild the offer with annexb=yes, and then the offer is
accepted by the B-leg. Our concern is that no transcoding resources are used
in this case, and we might run into audio problems because of that. The
other concern is that in the answer to the A-leg (in 183 and 200),
annexb=yes is included. I'm not sure if all devices would support different
fmtp parameters in the offer and the answer (the RFC, as usual, won't be
explicit about this).

Is there any way to force the transcoding in a situation like the one I
described?

Thanks in advance

Javi
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From rodrigo.ferrari at cellmidia.com.br  Mon Apr 25 16:45:27 2011
From: rodrigo.ferrari at cellmidia.com.br (Rodrigo Ferrari)
Date: Mon, 25 Apr 2011 09:45:27 -0300
Subject: [Freeswitch-users] Receiving SMS some alert?
Message-ID: <BANLkTimHvjOCLNUXUOSO9K=74=mno2ogTw@mail.gmail.com>

Hello friends,

I just bought a Khomp board to send and receive SMS, the problem is, I'm not
receiving the messages, it's not marking at the console a message incoming.
Is this some configuration that I miss?

Thanks, best regards.
Rodrigo Ferrari.
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From psilvao at gmail.com  Mon Apr 25 16:56:12 2011
From: psilvao at gmail.com (Pablo Silva)
Date: Mon, 25 Apr 2011 09:56:12 -0300
Subject: [Freeswitch-users] No Dial Tone ... Zaptel + FreeSwitch + TDM400P
Message-ID: <BANLkTimyKOrBq8TCDA3imnYS79b74=KgXA@mail.gmail.com>

Hi!


   I've tryed for severals hours to make a phone call from freeswitch
server, using TDM400P.

   I don't know why i can't hear the dial tone, and it's impossible
make a phone call.., i read freeswitch wiki page
about TDM400P configuration, as you can see that's my configuration an log

http://pastebin.com/Y94UqN5R

  if I focus on information from the log, you can see that loads
openzap, the server recognizes the TDM400P card, the call is routed to
the PSTN but no concrete achievements.

When you call, I hear the tone, not the dialing, I guess I'm near the
solution to my problem, I request your cooperation.

In advance, thanks.
-Pablo


From tayeb.meftah at gmail.com  Mon Apr 25 17:50:47 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Mon, 25 Apr 2011 15:50:47 +0200
Subject: [Freeswitch-users] No Dial Tone ... Zaptel + FreeSwitch +
	TDM400P
In-Reply-To: <BANLkTimyKOrBq8TCDA3imnYS79b74=KgXA@mail.gmail.com>
References: <BANLkTimyKOrBq8TCDA3imnYS79b74=KgXA@mail.gmail.com>
Message-ID: <4DB57C37.4020203@gmail.com>

openzap have bean renamed to freetdm
so please try mod_freetdm
thank you.
On 25/04/2011 14:56, Pablo Silva wrote:
> Hi!
>
>
>     I've tryed for severals hours to make a phone call from freeswitch
> server, using TDM400P.
>
>     I don't know why i can't hear the dial tone, and it's impossible
> make a phone call.., i read freeswitch wiki page
> about TDM400P configuration, as you can see that's my configuration an log
>
> http://pastebin.com/Y94UqN5R
>
>    if I focus on information from the log, you can see that loads
> openzap, the server recognizes the TDM400P card, the call is routed to
> the PSTN but no concrete achievements.
>
> When you call, I hear the tone, not the dialing, I guess I'm near the
> solution to my problem, I request your cooperation.
>
> In advance, thanks.
> -Pablo
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 zetruger at gmail.com  Mon Apr 25 17:55:29 2011
From: zetruger at gmail.com (=?KOI8-R?B?6dfBziD+ydPU0cvP1w==?=)
Date: Mon, 25 Apr 2011 17:55:29 +0400
Subject: [Freeswitch-users] Can FreeSWITCH play MoH from remote server?
Message-ID: <BANLkTimHZSaZAsyx0B5OwV1cLtcZDvtKcg@mail.gmail.com>

Can FreeSWITCH play MoH from remote server?


From jmesquita at freeswitch.org  Mon Apr 25 18:07:42 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 25 Apr 2011 11:07:42 -0300
Subject: [Freeswitch-users] Receiving SMS some alert?
In-Reply-To: <BANLkTimHvjOCLNUXUOSO9K=74=mno2ogTw@mail.gmail.com>
References: <BANLkTimHvjOCLNUXUOSO9K=74=mno2ogTw@mail.gmail.com>
Message-ID: <BANLkTim2HO4OhDMSo2yRPWxOfFyQWyCK6Q@mail.gmail.com>

Rodrigo,

Have you tried their official support? They do provide tech support and they
officially support FreeSWITCH. I am sure if you did, they did not reply
because of the holiday.

suporte (at) khomp.com.br should be just fine.

Regards,
Jo?o Mesquita



On Mon, Apr 25, 2011 at 9:45 AM, Rodrigo Ferrari <
rodrigo.ferrari at cellmidia.com.br> wrote:

> Hello friends,
>
> I just bought a Khomp board to send and receive SMS, the problem is, I'm
> not receiving the messages, it's not marking at the console a message
> incoming. Is this some configuration that I miss?
>
> Thanks, best regards.
> Rodrigo Ferrari.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Mon Apr 25 18:15:33 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 25 Apr 2011 16:15:33 +0200
Subject: [Freeswitch-users] call not connecting sometime
In-Reply-To: <4DB47783.5090507@ppmt.org>
References: <4DB47783.5090507@ppmt.org>
Message-ID: <2A5992B4-AABE-4041-A74E-4C7A956C4BD3@ipeva.fr>

Phil,

is there a provider involved ?
Perhaps it's on their side ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 24/04/2011 ? 21:18, Philippe Le Toquin a ?crit :

> Hello,
> 
> I had the problem a few weeks ago but after a reinstall (unrelated to that) the problem had gone so I put it
> done as my messing up my system at the time!
> 
> But today I had the same issue again. When I call sometime the call is not going through completely
> 
> The symptom on my side are that nothing happens (no ring tone) on the other they say that the phones
> rings but when they pick up the phone they can't hear anything.
> 
> Below is a siptrace where I changed the number and IP so I hope I didn't mess it up too much.
> 
> http://pastebin.freeswitch.org/16165
> 
> Can someone let me know if they see something wrong? I tried to understand but it is beyond me :(
> 
> Regards
> 
> /Philippe
> <0x1A0BDC2B.asc>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Mon Apr 25 18:19:39 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 25 Apr 2011 16:19:39 +0200
Subject: [Freeswitch-users] g729 annexa/annexb interoperability
In-Reply-To: <BANLkTim6q2QFwESamPZ4v6k_OUaAX+MbhQ@mail.gmail.com>
References: <BANLkTim6q2QFwESamPZ4v6k_OUaAX+MbhQ@mail.gmail.com>
Message-ID: <74E5FF91-CCFA-4CCA-BA99-725041271DCB@ipeva.fr>

Javier,

I dont think there is anything to transcode.
G729 and G729B are the same, except the latter uses VAD and CNG.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 25/04/2011 ? 13:44, Javier Gallart a ?crit :

> Hello all
> 
> we're using a Sangoma D100 card for transcoding. Our configuration, as far as the codec policy is concerned, basically does this:
> 
> -In the first offer to the b-leg we use the same codec list we receive (<action application="export" data="absolute_codec_string=${ep_codec_string}"/>)
> -If the call fails with status 488, we repeat the call using all the codecs available. This includes changing from annexb to annexa and viceversa.
> 
> Let's suppose a-leg only supports g729 annexA, and B-leg only supports G729 annexB. My understanding is that those different codec flavours don't interoperate; in fact I've experienced many times audio problems when trying to set up a call between endpoints supporting different g729 variants. 
> 
> In the described case, FS sends initially annexb=no, and B-leg rejects it with cause 488. We rebuild the offer with annexb=yes, and then the offer is accepted by the B-leg. Our concern is that no transcoding resources are used in this case, and we might run into audio problems because of that. The other concern is that in the answer to the A-leg (in 183 and 200), annexb=yes is included. I'm not sure if all devices would support different fmtp parameters in the offer and the answer (the RFC, as usual, won't be explicit about this).
> 
> Is there any way to force the transcoding in a situation like the one I described?
> 
> Thanks in advance
> 
> Javi
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 rodrigo.ferrari at cellmidia.com.br  Mon Apr 25 18:25:54 2011
From: rodrigo.ferrari at cellmidia.com.br (Rodrigo Ferrari)
Date: Mon, 25 Apr 2011 11:25:54 -0300
Subject: [Freeswitch-users] Receiving SMS some alert?
In-Reply-To: <BANLkTim2HO4OhDMSo2yRPWxOfFyQWyCK6Q@mail.gmail.com>
References: <BANLkTimHvjOCLNUXUOSO9K=74=mno2ogTw@mail.gmail.com>
	<BANLkTim2HO4OhDMSo2yRPWxOfFyQWyCK6Q@mail.gmail.com>
Message-ID: <BANLkTimmcXeyaHacWZZEEUs+kFLbRSjUJw@mail.gmail.com>

Obrigado Jo?o!

Eu tentei o suporte mas n?o obtive a resposta do que perguntei, tentei
novamente e estou aguardando devido ao feriado mesmo, por?m n?o consigo
ficar parado, fico tentando buscar solu??o por todos os lugares, meu
problema ? na quest?o de receber o SMS, pego meu celular e envio uma
mensagem para um chip da placa, com o console do freeswitch aberto, n?o
recebo nenhum evento de entrada e nem o dialplan ? executado, se ent?o eu
com meu celular ligo, no momento o freeswitch apita no console a entrada de
uma chamada e executa o dialplan, ent?o acredito estar esquecendo de
configurar algo para lidar com as mensagens que entram na placa.

Obrigado!

Abra?os.

Em 25 de abril de 2011 11:07, Jo?o Mesquita <jmesquita at freeswitch.org>escreveu:

> Rodrigo,
>
> Have you tried their official support? They do provide tech support and
> they officially support FreeSWITCH. I am sure if you did, they did not reply
> because of the holiday.
>
> suporte (at) khomp.com.br should be just fine.
>
> Regards,
> Jo?o Mesquita
>
>
>
> On Mon, Apr 25, 2011 at 9:45 AM, Rodrigo Ferrari <
> rodrigo.ferrari at cellmidia.com.br> wrote:
>
>> Hello friends,
>>
>> I just bought a Khomp board to send and receive SMS, the problem is, I'm
>> not receiving the messages, it's not marking at the console a message
>> incoming. Is this some configuration that I miss?
>>
>> Thanks, best regards.
>> Rodrigo Ferrari.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 lfurrea at gmail.com  Mon Apr 25 18:30:10 2011
From: lfurrea at gmail.com (Luis F Urrea)
Date: Mon, 25 Apr 2011 08:30:10 -0600
Subject: [Freeswitch-users] FreeTDM disconnect supervision
In-Reply-To: <BANLkTikNrdU0aFHUxvGJ=sSfPD=_SzZgJg@mail.gmail.com>
References: <BANLkTikwxtkATFaK0jQJhpnsvgOhxJp2RQ@mail.gmail.com>
	<BANLkTinRmBORb9zO0yvt22sWW8JE2E7g3A@mail.gmail.com>
	<BANLkTimk_=m2GVZB=c7PtxqZoBJWR0ELMA@mail.gmail.com>
	<BANLkTikNrdU0aFHUxvGJ=sSfPD=_SzZgJg@mail.gmail.com>
Message-ID: <BANLkTineb1_p9rd1mm2fism=aNbyo6gY5g@mail.gmail.com>

Hi all,

I am updating the thread as per Moises Silva's request so he can answer for
posterity. Not just to bump the thread.

Thx.

Hello Luis,

Can you point to the mailing list conversation and please update the thread
because if I deleted the thread in my mail already I have no way to reply
unless someone else sends a new reply. I'd rather reply there to leave the
response on the internet available for anyone. If you don't have it please
create a new thread in freeswitch-users and I'll reply there.

On Mon, Apr 18, 2011 at 5:38 PM, Michael Collins <msc at freeswitch.org> wrote:

> Generating a tone w/ cadence is a piece of cake. *Detecting* a tone and
> cadence is a bit trickier. Nothing in FreeTDM (that I'm aware of) can detect
> both tone *and* cadence, which is why it works great in the US (which uses a
> combination of tones) as opposed to Mexico (which uses, I think 425Hz for
> everything).
>
> I'd ask Moises Silva if there's any plans to add the cadence detection.
>
> -MC
>
>
> On Mon, Apr 18, 2011 at 4:27 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>
>> Yes guru,
>>
>> The point is that using tone_detect I am not able to specify cadence
>> either.
>>
>> I need to be able to detect a tone that plays 450Hz for 330ms and then
>> silence for 330ms.
>>
>> Strange thing is that on tones.conf generate tones do seem to specify
>> cadence such as US busy
>>
>> v=-7;(500,500,420,680)
>>
>> I would interpret 420Hz for .5s and 680Hz for .5s
>>
>> Is that how it is to be interpreted??
>>
>>
>> TIA
>>
>>
>> On Mon, Apr 18, 2011 at 4:48 PM, guru singh <grsingh750 at gmail.com> wrote:
>>
>>> Hi Luis,
>>>
>>> You could try the dialplan application tone_detect to detect the busy
>>> tone and hangup.
>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>>>
>>> guru
>>>
>>> On Tue, Apr 19, 2011 at 3:34 AM, Luis F Urrea <lfurrea at gmail.com> wrote:
>>> > Hello,
>>> > According to what I have found in regards the tones used for signaling
>>> on
>>> > FreeTDM, it seems that anything set through DAHDI is ignored and specs
>>> from
>>> > tones.conf are used instead.
>>> > However I have not been able to properly detect a busy tone to be able
>>> to
>>> > set an FXO back on hook once a busy tone is sniffed.
>>> > tones.conf has the following references for busy tones [us]:
>>> > generate-busy => v=-7;%(500,500,480,620)
>>> > detect-busy=> 480,620
>>> > I would like to know if detect-busy takes into account the time frame
>>> that
>>> > the frequencies are on (played) or off (not being played) and if there
>>> is
>>> > somehow a way to specify this. Is there any default value for time on
>>> and
>>> > time off?
>>> > I appreciate your help!
>>> > 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
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 rupa at rupa.com  Mon Apr 25 19:03:28 2011
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 25 Apr 2011 10:03:28 -0500
Subject: [Freeswitch-users] Can FreeSWITCH play MoH from remote server?
In-Reply-To: <BANLkTimHZSaZAsyx0B5OwV1cLtcZDvtKcg@mail.gmail.com>
References: <BANLkTimHZSaZAsyx0B5OwV1cLtcZDvtKcg@mail.gmail.com>
Message-ID: <BANLkTimPgo49ALbTsDX1gM46ZhfDiKBt+Q@mail.gmail.com>

You can use mod_shout to play an mp3 stream from a shoutcast or icecast server.

On Mon, Apr 25, 2011 at 8:55 AM, ???? ???????? <zetruger at gmail.com> wrote:
> Can FreeSWITCH play MoH from remote server?

-- 
-Rupa


From boris at tagnet.ru  Mon Apr 25 19:11:54 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Mon, 25 Apr 2011 21:11:54 +0600
Subject: [Freeswitch-users] =?utf-8?b?0K3RhdC+INC90LAg0JHQtdC70L7RgNGD0YE=?=
	=?utf-8?b?0YHQuNGO?=
Message-ID: <4DB58F3A.9030706@tagnet.ru>

?????? ????!

25.04.2011 20:26:06 +375292718781 - ?? ????????? ????? (? ??????????) 
??????? ??????????? ???.

-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



From boris at tagnet.ru  Mon Apr 25 19:16:53 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Mon, 25 Apr 2011 21:16:53 +0600
Subject: [Freeswitch-users]
 =?utf-8?b?0K3RhdC+INC90LAg0JHQtdC70L7RgNGD0YE=?=
 =?utf-8?b?0YHQuNGO?=
In-Reply-To: <4DB58F3A.9030706@tagnet.ru>
References: <4DB58F3A.9030706@tagnet.ru>
Message-ID: <4DB59065.8000607@tagnet.ru>

Ops... sorry, wrong list.

> ?????? ????!
>
> 25.04.2011 20:26:06 +375292718781 - ?? ????????? ????? (? ??????????)
> ??????? ??????????? ???.
>


-- 
Regards,
Boris




From nicolas at medularis.com  Mon Apr 25 19:22:21 2011
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 25 Apr 2011 11:22:21 -0400
Subject: [Freeswitch-users] absolute_codec_string question
Message-ID: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>

Hi,

I'm trying to force the use of a specific codec when originating a call from
the command line.

This is my codecs config in vars.xml:

  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,SPEEX,G729"/>
  <X-PRE-PROCESS cmd="set"
data="outbound_codec_prefs=PCMU,PCMA,SPEEX,G729"/>


If I just originate a call:

originate {ignore_early_media}sofia/gateway/mygateway/444444
&bridge(user/1001)

I get the following SDP:

   v=0
   o=FreeSWITCH 1303727406 1303727407 IN IP4 127.0.0.1
   s=FreeSWITCH
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 17054 RTP/AVP 0 8 98 18 101 13
   a=rtpmap:98 SPEEX/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20


If I originate a call with the absolute_codec_string variable set like this:

originate
{ignore_early_media,absolute_codec_string='G729,PCMU'}sofia/gateway/
mygateway/444444 &bridge(user/1001)

I get the following SDP:

   v=0
   o=FreeSWITCH 1303719860 1303719861 IN IP4 127.0.0.1
   s=FreeSWITCH
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 24894 RTP/AVP 0 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20


And it doesn't matter the value I give to the absolute_codec_string, I will
get the same SDP, except if I specify SPEEX, then I'll get the same SDP as
the one above. Am I missing something, or is this the way it's supposed to
work?

I also tried with the codec_string variable, and I get the same results.

I'm using the following version from the git repository: head-git-244fd68
2011-03-21 14-27-57 -0400


Thanks for your help,


Nicolas
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From paul at cupis.co.uk  Mon Apr 25 20:06:30 2011
From: paul at cupis.co.uk (Paul Cupis)
Date: Mon, 25 Apr 2011 17:06:30 +0100
Subject: [Freeswitch-users] absolute_codec_string question
In-Reply-To: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>
References: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>
Message-ID: <4DB59C06.3090706@cupis.co.uk>

On 25/04/11 16:22, Nicolas Brenner wrote:
> I'm trying to force the use of a specific codec when originating a call from
> the command line.

>    m=audio 17054 RTP/AVP 0 8 98 18 101 13

>    m=audio 24894 RTP/AVP 0 101 13

> And it doesn't matter the value I give to the absolute_codec_string, I will
> get the same SDP, except if I specify SPEEX, then I'll get the same SDP as
> the one above. Am I missing something, or is this the way it's supposed to
> work?

Looks okay to me. Perhaps you are not looking at the m=audio line in the
SDP which you will is different - this reflects the difference in the
offered codecs.

You might be expecting to see all of the codecs itemised in the a=fmtp
lines - this is known as verbose-sdp and is disabled by default.

Have a look at:
http://wiki.freeswitch.org/wiki/Variable_verbose_sdp

Regards,


From robert.hadley at teotech.com  Mon Apr 25 20:19:35 2011
From: robert.hadley at teotech.com (Robert Hadley)
Date: Mon, 25 Apr 2011 09:19:35 -0700
Subject: [Freeswitch-users] config ramdisk on CentOS 5.5
In-Reply-To: <BANLkTi=aDMwRvUv6EdNG6Fx-ctaYPZ+nPA@mail.gmail.com>
References: <BANLkTim4SX5JTB0-h=JaMsoyBtyW_wbEwA@mail.gmail.com>
	<C9DA7C39.1453D%krice@freeswitch.org>
	<BANLkTi=aDMwRvUv6EdNG6Fx-ctaYPZ+nPA@mail.gmail.com>
Message-ID: <E826C94A3C6BE44198FAF4D66B5AB4D814A48A9240@VA3DIAXVS351.RED001.local>


On CentOS the mount command executes the /etc/fstab file, to make the tmpfs without rebooting try:
mount -a

-Robert

-----Original Message-----
From: fieldpeak [mailto:fieldpeak at gmail.com] 
Sent: Monday, April 25, 2011 12:04 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] config ramdisk on CentOS 5.5

thanks Ken. I noticed the voice mail issue when using tmpfs, thanks again.

Regards,
Charles

On 4/25/11, Ken Rice <krice at freeswitch.org> wrote:
> That is correct
>
> Keep in mind that in using a ramdrive for freeswitch/db if you are 
> running voicemail you?ll need to look at mechanismd for backing up its 
> db
>
> K
>
>
> On 4/25/11 12:42 AM, "fieldpeak" <fieldpeak at gmail.com> wrote:
>
>> Hi Ken,
>>
>> Thanks for your reply.
>>
>> if my below understanding correct?
>> if i add below to the end of /etc/fstab, tmpfs 
>> /usr/local/freeswitch/db tmpfs defaults 0 0
>>
>> and then reboot machine, the system will auto mount the tmpfs.
>> i don't need run mannually mount /usr/local/freeswitch/db before 
>> start freeswitch...
>>
>> Thanks.
>>
>> Regards,
>> Charles
>>
>> 2011/4/23 Ken Rice <krice at freeswitch.org>
>>> Tmpfs is not a program... Read that page a little closer... That?s 
>>> particular line is for your fstab...
>>>
>>> If you want to mount it from the command line its mount ?o tmpfs 
>>> tmpfs /usr/local/freeswitch/db
>>>
>>> K
>>>
>>>
>>>
>>> On 4/22/11 10:50 PM, "fieldpeak" <fieldpeak at gmail.com 
>>> <http://fieldpeak at gmail.com> > wrote:
>>>
>>>> i'm trying tuning the FS to max performance on centos 5.5, and 
>>>> referred to 
>>>> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurati
>>>> ons#FreeS
>>>> WITCH.27s_core.db_I.2FO_bottleneck
>>>>
>>>> #1,? i configure the DB of FS to ramdisk ,
>>>>
>>>> when i run "tmpfs /opt/freeswitch/db tmpfs defaults 0 0", it output:
>>>> "-bash:
>>>> tmpfs: command not found"
>>>>
>>>> #2,? i run "ethtool -g eth0", the output is below, what value i 
>>>> should config for RX and TX for max performance...
>>>> Ring parameters for eth0:
>>>> Pre-set maximums:
>>>> RX:???????????? 4096
>>>> RX Mini:??????? 0
>>>> RX Jumbo:?????? 0
>>>> TX:???????????? 4096
>>>> Current hardware settings:
>>>> RX:???????????? 256
>>>> RX Mini:??????? 0
>>>> RX Jumbo:?????? 0
>>>> TX:???????????? 256
>>>>
>>>> Apprecited if anyone help how to configure it... thanks!
>>>>
>>>> Regards,
>>>> Charles
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> 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-u
>>> sers
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>> ers
>> http://www.freeswitch.org
>
>





From nico at clickfono.com  Mon Apr 25 20:22:01 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Mon, 25 Apr 2011 12:22:01 -0400
Subject: [Freeswitch-users] absolute_codec_string question
In-Reply-To: <4DB59C06.3090706@cupis.co.uk>
References: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>
	<4DB59C06.3090706@cupis.co.uk>
Message-ID: <BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>

Thanks Paul!

You are totally right, I was looking for the a=rtpmap:18 G729/8000 line. I
find it weird that it is automatically inserted for SPEEX, but not for other
codecs.

Another thing I noticed, is I can't specify more than one codec, according
to what I've read about other issues, this has something to do with a comma
parsing problem on the console (for some reason commas can't be escaped,
this also affects trying to originate calls from ESL).

So for absolute_codec_string=G729 I get this SDP (with verbose_sdp=true):

   v=0
   o=FreeSWITCH 1303718366 1303718367 IN IP4 127.0.0.1
   s=FreeSWITCH
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 29882 RTP/AVP 18 101 13
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20


But for absolute_codec_string=G729,PCMU I get no additional m= lines in the
SDP:

   v=0
   o=FreeSWITCH 1303716332 1303716333 IN IP4 127.0.0.1
   s=FreeSWITCH
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 32008 RTP/AVP 18 101 13
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20




On Mon, Apr 25, 2011 at 12:06 PM, Paul Cupis <paul at cupis.co.uk> wrote:

> On 25/04/11 16:22, Nicolas Brenner wrote:
> > I'm trying to force the use of a specific codec when originating a call
> from
> > the command line.
>
> >    m=audio 17054 RTP/AVP 0 8 98 18 101 13
>
> >    m=audio 24894 RTP/AVP 0 101 13
>
> > And it doesn't matter the value I give to the absolute_codec_string, I
> will
> > get the same SDP, except if I specify SPEEX, then I'll get the same SDP
> as
> > the one above. Am I missing something, or is this the way it's supposed
> to
> > work?
>
> Looks okay to me. Perhaps you are not looking at the m=audio line in the
> SDP which you will is different - this reflects the difference in the
> offered codecs.
>
> You might be expecting to see all of the codecs itemised in the a=fmtp
> lines - this is known as verbose-sdp and is disabled by default.
>
> Have a look at:
> http://wiki.freeswitch.org/wiki/Variable_verbose_sdp
>
> 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 david.ponzone at ipeva.fr  Mon Apr 25 20:33:20 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Mon, 25 Apr 2011 18:33:20 +0200
Subject: [Freeswitch-users] absolute_codec_string question
In-Reply-To: <BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>
References: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>
	<4DB59C06.3090706@cupis.co.uk>
	<BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>
Message-ID: <AE478CA0-8AA1-48CB-A439-8B39AB968C54@ipeva.fr>

Nicolas,

it's because the payload type for G729 is an old static one.
Speex doesnt have its own static payload type, as all the recent codecs. It's dynamic, so a fmtp line is required.

For your issue with 2 codecs, that's not normal.
Are you sure you did reloadxml ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 25/04/2011 ? 18:22, Nicolas Brenner a ?crit :

> Thanks Paul!
> 
> You are totally right, I was looking for the a=rtpmap:18 G729/8000 line. I find it weird that it is automatically inserted for SPEEX, but not for other codecs.
> 
> Another thing I noticed, is I can't specify more than one codec, according to what I've read about other issues, this has something to do with a comma parsing problem on the console (for some reason commas can't be escaped, this also affects trying to originate calls from ESL).
> 
> So for absolute_codec_string=G729 I get this SDP (with verbose_sdp=true):
> 
>    v=0
>    o=FreeSWITCH 1303718366 1303718367 IN IP4 127.0.0.1
>    s=FreeSWITCH
>    c=IN IP4 127.0.0.1
>    t=0 0
>    m=audio 29882 RTP/AVP 18 101 13
>    a=rtpmap:18 G729/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
> 
> 
> But for absolute_codec_string=G729,PCMU I get no additional m= lines in the SDP:
> 
>    v=0
>    o=FreeSWITCH 1303716332 1303716333 IN IP4 127.0.0.1
>    s=FreeSWITCH
>    c=IN IP4 127.0.0.1
>    t=0 0
>    m=audio 32008 RTP/AVP 18 101 13
>    a=rtpmap:18 G729/8000
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
> 
> 
> 
> 
> On Mon, Apr 25, 2011 at 12:06 PM, Paul Cupis <paul at cupis.co.uk> wrote:
> On 25/04/11 16:22, Nicolas Brenner wrote:
> > I'm trying to force the use of a specific codec when originating a call from
> > the command line.
> 
> >    m=audio 17054 RTP/AVP 0 8 98 18 101 13
> 
> >    m=audio 24894 RTP/AVP 0 101 13
> 
> > And it doesn't matter the value I give to the absolute_codec_string, I will
> > get the same SDP, except if I specify SPEEX, then I'll get the same SDP as
> > the one above. Am I missing something, or is this the way it's supposed to
> > work?
> 
> Looks okay to me. Perhaps you are not looking at the m=audio line in the
> SDP which you will is different - this reflects the difference in the
> offered codecs.
> 
> You might be expecting to see all of the codecs itemised in the a=fmtp
> lines - this is known as verbose-sdp and is disabled by default.
> 
> Have a look at:
> http://wiki.freeswitch.org/wiki/Variable_verbose_sdp
> 
> 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
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ctroncoso at redvoiss.net  Mon Apr 25 20:29:38 2011
From: ctroncoso at redvoiss.net (Camila Troncoso)
Date: Mon, 25 Apr 2011 13:29:38 -0300
Subject: [Freeswitch-users] mod_radius_cdr
Message-ID: <b7227254510d17dfbe03483165f1f696@mail.gmail.com>

Hi,



I?m working with radius accounting using the mod_radius_cdr module. I?m
having trouble with the date format that Radius server receive.  An example
of what the radius server receive is:



?Freeswitch-Callstartdate = "2011-04-21T18:23:59.780945-0400"?



This date format is very difficult to read and I want to change it  to make
the accounting easier. I search all around for some param or configuration
that allows me to do so, but I only find that this date format is define in
mod_radius_cdr.c module.



Please some help with this matter.



Regards,



*Camila Troncoso **|* Ingeniero de Desarrollo

RedVoiss *|*ctroncoso at redvoiss.net

Santiago - Chile *|* +56 2 2408535

www.redvoiss.net
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From paul at cupis.co.uk  Mon Apr 25 20:45:49 2011
From: paul at cupis.co.uk (Paul Cupis)
Date: Mon, 25 Apr 2011 17:45:49 +0100
Subject: [Freeswitch-users] absolute_codec_string question
In-Reply-To: <BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>
References: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>	<4DB59C06.3090706@cupis.co.uk>
	<BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>
Message-ID: <4DB5A53D.50201@cupis.co.uk>

On 25/04/11 17:22, Nicolas Brenner wrote:
> You are totally right, I was looking for the a=rtpmap:18 G729/8000 line. I
> find it weird that it is automatically inserted for SPEEX, but not for other
> codecs.

SPEEX does not have an IANA allocated mapping, so used dynamic mapping
(98 is within the dynamic range) - dynamically mapped codecs must be
specified verbosely in the SDP.

> Another thing I noticed, is I can't specify more than one codec, according
> to what I've read about other issues, this has something to do with a comma
> parsing problem on the console (for some reason commas can't be escaped,
> this also affects trying to originate calls from ESL).
> 
> So for absolute_codec_string=G729 I get this SDP (with verbose_sdp=true):

Try something like:

 {absolute_codec_string='G729,PCMU'}

The quoting of the codecs should mean that the comma is "escaped" and
therefore both codecs will be offered.

Regards,


From nico at clickfono.com  Mon Apr 25 21:08:20 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Mon, 25 Apr 2011 13:08:20 -0400
Subject: [Freeswitch-users] absolute_codec_string question
In-Reply-To: <4DB5A53D.50201@cupis.co.uk>
References: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>
	<4DB59C06.3090706@cupis.co.uk>
	<BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>
	<4DB5A53D.50201@cupis.co.uk>
Message-ID: <BANLkTi=gh6AyV9C-fQaR6cY-x0=VX=XDeg@mail.gmail.com>

Thanks for the information, good to know FreeSWITCH is behaving well :P

David, reloadxml doesn't do anything, since I'm trying to use the
absolute_codec_string with originate from the console, like so:

originate
{ignore_early_media,verbose_sdp=true,absolute_codec_string='G729,PMCU'}sofia/gateway/mygateway/444444
&bridge(user/1001)

Paul, I am using {absolute_codec_string='G729,PCMU'}, and I get the same as
if I don't quote the string, or if I just specify one codec:


   v=0
   o=FreeSWITCH 1303728783 1303728784 IN IP4 127.0.0.1
   s=FreeSWITCH
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 22426 RTP/AVP 18 101 13
   a=rtpmap:18 G729/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20



On Mon, Apr 25, 2011 at 12:45 PM, Paul Cupis <paul at cupis.co.uk> wrote:

> On 25/04/11 17:22, Nicolas Brenner wrote:
> > You are totally right, I was looking for the a=rtpmap:18 G729/8000 line.
> I
> > find it weird that it is automatically inserted for SPEEX, but not for
> other
> > codecs.
>
> SPEEX does not have an IANA allocated mapping, so used dynamic mapping
> (98 is within the dynamic range) - dynamically mapped codecs must be
> specified verbosely in the SDP.
>
> > Another thing I noticed, is I can't specify more than one codec,
> according
> > to what I've read about other issues, this has something to do with a
> comma
> > parsing problem on the console (for some reason commas can't be escaped,
> > this also affects trying to originate calls from ESL).
> >
> > So for absolute_codec_string=G729 I get this SDP (with verbose_sdp=true):
>
> Try something like:
>
>  {absolute_codec_string='G729,PCMU'}
>
> The quoting of the codecs should mean that the comma is "escaped" and
> therefore both codecs will be offered.
>
> 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 Nabble at slickdeals.endjunk.com  Mon Apr 25 21:21:21 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Mon, 25 Apr 2011 10:21:21 -0700 (PDT)
Subject: [Freeswitch-users] mod_spandsp fails to cross compile
Message-ID: <1303752081792-6303345.post@n2.nabble.com>

I don't remember when this problem started; however, I have been doing a git
pull every time trying to re-compile my local FS source for the past two
weeks and always ended up with the same error messages on mod_spandsp
compilation when trying to cross compile FS as shown 
http://pastebin.com/7uKGPWxQ here  on line #13 (mod_spandsp_fax.c:1656:13:
error: implicit declaration of function 't38_gateway_rx_fillin').

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-spandsp-fails-to-cross-compile-tp6303345p6303345.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Mon Apr 25 21:36:19 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Apr 2011 10:36:19 -0700
Subject: [Freeswitch-users] Community Thank Yous
Message-ID: <BANLkTi=iG8PPnRZc=aFQxFhRdJsyt4QOyA@mail.gmail.com>

Happy Monday to you all!

I just wanted to take a moment to say thank you to the many folks who have
been stepping up to help out with the not-so-glamorous side of running an
open source project: documentation and community support. We all know that
hacking code is the fun part, but writing documentation and helping out with
IRC and mailing list questions are often thankless tasks. I wanted to
personally say thank you to all those who've been stepping up.

Regarding documentation, I've seen dozens of new users on the wiki, each of
whom is adding something. This is indicative of people "paying the wiki tax"
when a question is answered via ML or IRC. So to those who have been adding
knowledge to the wiki: thank you! It is much appreciated. Please keep up the
good work.

In the IRC channel we've been having a good number of community members come
in and make themselves available for answering questions. This can be very
time-consuming, so again we say thank you. The same thing goes for the
mailing list. It takes a lot of time to read through all the posts and
respond. The following users have been especially active in helping others
of late and I feel it is only fair to acknowledge their fine efforts. In no
particular order...

Ken Rice
Steven Ayre
Jeff Link
Avi Marcus
Kristian Kielhofner
Steve Underwood
Peter Olsson
Stefan Knobloch
Giovanni Maruzzelli
Christopher Rienzo

Every time a community member helps another community member we grow
stronger. Please keep up the good work!

-Michael
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From msc at freeswitch.org  Mon Apr 25 21:43:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Apr 2011 10:43:40 -0700
Subject: [Freeswitch-users] mod_enum compilation error
In-Reply-To: <1303692412837-6301771.post@n2.nabble.com>
References: <1303692412837-6301771.post@n2.nabble.com>
Message-ID: <BANLkTi=WxghpfjwA6vjWMNqzZbatB2e3YQ@mail.gmail.com>

open a jira please.
-MC

On Sun, Apr 24, 2011 at 5:46 PM, mazilo <Nabble at slickdeals.endjunk.com>wrote:

> I have a local FS git pulled to adffe07f7512b00f7093091778965a119ca81513
> and
> failed to compile mod_enum for OpenWRT as shown below:
>
> making all mod_enum
> make[7]: Entering directory
>
> `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/mod/applications/mod_enum'
> make[7]: *** No rule to make target `Makefile.am', needed by `Makefile.in'.
> Stop.
> make[7]: Leaving directory
>
> `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/mod/applications/mod_enum'
> make[6]: *** [mod_enum-all] Error 1
> make[6]: Leaving directory
>
> `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src/mod'
> make[5]: *** [all-recursive] Error 1
> make[5]: Leaving directory
>
> `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/src'
> make[4]: *** [all-recursive] Error 1
> make[4]: Leaving directory
>
> `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[3]: *** [all] Error 2
> make[3]: Leaving directory
>
> `/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git'
> make[2]: ***
>
> [/opt/tmp/openwrt-svn-trunk/build_dir/target-arm_v5te_uClibc-0.9.32_eabi/freeswitch_git/.built]
> Error 2
> make[2]: Leaving directory
> `/opt/tmp/OpenWRT/feeds/packages/net/freeswitch_git'
> make[1]: *** [package/feeds/local/freeswitch_git/compile] Error 2
> make[1]: Leaving directory `/opt/tmp/openwrt-svn-trunk'
> make: *** [package/freeswitch_git/compile] Error 2
>
>
> -----
> FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/mod-enum-compilation-error-tp6301771p6301771.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  Mon Apr 25 21:52:18 2011
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 25 Apr 2011 10:52:18 -0700
Subject: [Freeswitch-users] conference_set_auto_outcall No Audio
In-Reply-To: <397737.87013.qm@web59401.mail.ac4.yahoo.com>
References: <397737.87013.qm@web59401.mail.ac4.yahoo.com>
Message-ID: <BANLkTik8E2ebEsS9hV4yKwo9ybMBiBN5KA@mail.gmail.com>

Welcome to the FreeSWITCH community!

Okay, in cases like this you will need to break out the information
gathering tools. Start with the stuff here:

http://wiki.freeswitch.org/wiki/Reporting_Bugs

You'll need a FreeSWITCH console log. You might also want to do a tcpdump on
the RTP and see if it's really flowing. Hopefully that will yield a clue as
to what you need to do...

You can also dump all this into pastebin and put the URL in this thread. The
link I provided has all the information you need on how to collect
information and put it on pastebin.

Hope it goes well!

-MC

On Fri, Apr 22, 2011 at 7:38 PM, Chris Monkey <ckmonkey158 at yahoo.com> wrote:

> Hi,
>
> I am am relatively new to FS and have setup a home PBX with about 5 Snom
> 320s with TLS. Everything works fine, except when I try to use the mad_boss
> example, or any auto_outcall conference, the phones ring and will auto
> answer, but there is no audio. However, if I dial in to the conference,
> everything works fine.
>
> I've tried adding <action application="set" data="sip_secure_media=true"/>
> to the conference extension, but to no avail. I don't see any errors in the
> log, and I just did a make clean this afternoon. What am I missing?
>
> 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 eagle.antonio at gmail.com  Mon Apr 25 21:53:44 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Mon, 25 Apr 2011 18:53:44 +0100
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <1303724917.8952.10.camel@luna.tc.commsmundi.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<4DB068C4.90109@xpirio.com>
	<6294202E-7ACB-49C3-89B8-2DE4631EA66B@gmail.com>
	<4DB1550C.1050607@xpirio.com>
	<BANLkTinuw7xoNB_8rfrsZuj=02MW_JiN1g@mail.gmail.com>
	<BANLkTimBWSwvjk3=C3oEZLe4U3ig94_Gqg@mail.gmail.com>
	<BANLkTik4ZpAP2uK0iFoxMfGW2xaEy+8w9g@mail.gmail.com>
	<BANLkTi=e3w6WH=ss4CvuO5W1Qn8QXzqXqA@mail.gmail.com>
	<BANLkTin6c0q_66bNk-W+1WjVKkf+0w27FA@mail.gmail.com>
	<B40BD7F2-EEB9-4C1E-AE1F-6474FFFBAE71@gmail.com>
	<1303719841.19225.29.camel@luna.tc.commsmundi.com>
	<BANLkTikZQNBEFkPqEG86N8QKTLWFk6yegg@mail.gmail.com>
	<1303724917.8952.10.camel@luna.tc.commsmundi.com>
Message-ID: <BANLkTi=XfkUrNKCmGNG2yMdKQyjYtzH9Tg@mail.gmail.com>

Humm What about this one.

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup

and / or

there is also a possibility to hangup if you do not recveive any media (RTP)
just be careful with Voice Activation.



2011/4/25 Fran?ois Delawarde <fdelawarde at wirelessmundi.com>

> Thanks Antonio, I'm already using that along with iptables rate
> limiting. It fits my needs but I need to run tests every time I install
> FS in a new hardware to find the optimal values.
>
> It's good enough for me, but I was just wondering if there was anything
> more automatic like some timer that could drop the call if it takes too
> long to process.
>
> It could also be useful for stalled DB connections or similar endless
> loops in some badly coded IVR applications.
>
> Fran?ois.
>
>
> On Mon, 2011-04-25 at 10:05 +0100, Antonio Teixeira wrote:
> > Good Morning.
> >
> > @ Steven,
> > Thanks for the idea i will make a prototype in Python , and will send
> > you guysthe performance data when i have it.
> >
> > @ Fran?ois
> >
> > You probably want this.
> > http://wiki.freeswitch.org/wiki/XML_Switch_Configuration
> >
> > 'sessions-per-second' - throttling mechanism, the switch will only
> > create this many channels at most, per second.
> >
> >
> >
> > 2011/4/25 Fran?ois Delawarde <fdelawarde at wirelessmundi.com>
> >         I also experienced a huge delay issue when attacked by mister
> >         "friendly-scanner". The machine was not overloaded and call
> >         setup was
> >         working well but with a 10-20 seconds delay. Note that the
> >         delay was
> >         only with SIP answers, and once RTP was setup the audio was
> >         perfect.
> >
> >         Following advices from http://wiki.freeswitch.org/wiki/QoS
> >         solved it in
> >         my case. Is there a mechanism in FS that could detect huge
> >         call setup
> >         delay and simply reject calls?
> >
> >         Fran?ois.
> >
> >
> >
> >
> >         On Sat, 2011-04-23 at 14:46 +0100, Steven Ayre wrote:
> >         > ESL would be best as you can offload the processing to
> >         another server
> >         > - potentially a load balanced cluster.
> >         >
> >         > Steve on iPhone
> >         >
> >         > On 23 Apr 2011, at 08:51, Antonio Teixeira
> >         <eagle.antonio at gmail.com>
> >         > wrote:
> >         >
> >         >
> >         >
> >         > > Hello Michael & The rest.
> >         > >
> >         > > What i'm trying to find is the solution with the highest
> >         performance
> >         > > for IVR Applications. And squeezing the last bit of
> >         performance out
> >         > > of freeswitch is a needed path specially on a industry
> >         that does
> >         > > second by second billing :P
> >         > > i'm going to do some performance testing with Lua , ESL
> >         And Mod Curl
> >         > > and of course python and measure some performance data and
> >         will post
> >         > > it later.
> >         > >
> >         > > A/T
> >         > >
> >         > >
> >         > > 2011/4/22 Michael Collins <msc at freeswitch.org>
> >         > >         You can definitely use the event socket. Heck, you
> >         can even
> >         > >         use Python if you want to. The dev team wrote ESL
> >         > >         specifically for cases like these - where you want
> >         to
> >         > >         control FS externally. ESL beats the pants off AGI
> >         scripts.
> >         > >         It gives you complete control over the system.
> >         > >
> >         > >
> >         > >         That all being said, you can do all sorts of stuff
> >         with Lua
> >         > >         dialplan scripts. Check out chapter 7 of the
> >         FreeSWITCH book
> >         > >         for some nice examples. Lua is lightweight and
> >         easy to
> >         > >         learn. It's a good alternative for many cases.
> >         Check it
> >         > >         out...
> >         > >
> >         > >
> >         > >         -MC
> >         > >
> >         > >
> >         > >
> >         > >         On Fri, Apr 22, 2011 at 12:39 PM, Antonio Teixeira
> >         > >         <eagle.antonio at gmail.com> wrote:
> >         > >                 Well i totally agree with the Dev team
> >         decision , it
> >         > >                 would be impossible for a DEV team to "get
> >         it right"
> >         > >                 in all the distros that's why i started
> >         this post.
> >         > >
> >         > >                 But we also need to take into
> >         consideration that we
> >         > >                 are talking about IVR Processing ,  not
> >         > >                 auto-attendants or simply call pass
> >         trough.
> >         > >                 This means heavy use of TTS / ASR ,
> >         Database Logic
> >         > >                 and Scripts , GetDigits and the works.
> >         > >
> >         > >                 I use python alot , but i think the
> >         mod_python is
> >         > >                 not the best tool for this job i admit
> >         that , that's
> >         > >                 why im currently looking for other
> >         solutions.
> >         > >                 I'm thinking in using mod_socket , but
> >         that scares
> >         > >                 me ( let? say bad experiences with
> >         Asterisk AGI) Or
> >         > >                 Mod Curl
> >         > >
> >         > >                 The main problem is that some IVR are
> >         extremely
> >         > >                 complex , like questionnaires , etc.
> >         > >
> >         > >                 It would be great if we could
> >         mod_event_zmq to
> >         > >                 control the calls <330.gif>
> >         > >
> >         > >                 Just to Sum it UP so far , so i can later
> >         add it to
> >         > >                 the wiki.
> >         > >
> >         > >                 Use Cent OS 64.
> >         > >                 Use tmpfs for all the databases.
> >         > >
> >         > >                 Thank you all for helping and Happy
> >         Eastern.
> >         > >                 Ant?nio Teixeira
> >         > >
> >         > >
> >         > >
> >         > >
> >         > >
> >         > >
> >         > >                 2011/4/22 Michael Collins
> >         <msc at freeswitch.org>
> >         > >                         FreeSWITCH runs well on many
> >         platforms.
> >         > >                         However, the devs are painfully
> >         aware that
> >         > >                         bleeding edge distros have
> >         bleeding edge gcc
> >         > >                         compilers and other interesting
> >         issues. That
> >         > >                         being said, CentOS 5.x is "stable"
> >         in that
> >         > >                         it's old and boring, therefore it
> >         has the
> >         > >                         least amount of drama. OTOH, some
> >         of our
> >         > >                         users have been having great
> >         success with
> >         > >                         Arch Linux (IRC:bougeyman) and FS,
> >         even
> >         > >                         though Arch uses very recent
> >         kernels.
> >         > >
> >         > >
> >         > >                         Bottom line: if you know what
> >         you're doing
> >         > >                         then you can probably run FS
> >         anywhere. If
> >         > >                         you don't know what you're doing
> >         then stick
> >         > >                         with CentOS 5.x or Debian Lenny
> >         until you
> >         > >                         do. (I run then both with zero
> >         issues,
> >         > >                         compiling latest git each day.)
> >         > >
> >         > >
> >         > >                         -MC
> >         > >
> >         > >
> >         > >
> >         > >                         On Fri, Apr 22, 2011 at 10:03 AM,
> >         > >                         curriegrad2004
> >         <curriegrad2004 at gmail.com>
> >         > >                         wrote:
> >         > >                                 Freeswitch is targeted for
> >         CentOS
> >         > >                                 5.3, which in my opinion
> >         quite short
> >         > >                                 sighted for the developers
> >         to do
> >         > >                                 this. However with the
> >         limited size
> >         > >                                 of developers and testers,
> >         I'm
> >         > >                                 afraid there's not much
> >         platforms we
> >         > >                                 can throughly test and
> >         actually say
> >         > >                                 "okay, FS will run
> >         flawlessly on X
> >         > >                                 distro"
> >         > >
> >         > >                                 However you can always try
> >         messing
> >         > >                                 with the CFLAG's mtune
> >         option and
> >         > >                                 see what it produces for
> >         you...
> >         > >
> >         > >                                 2011/4/22 Christian
> >         L?schenkohl
> >         > >
> >         <christian.loeschenkohl at xpirio.com>:
> >         > >
> >         > >                                 > hi
> >         > >                                 >
> >         > >                                 > if you refer to my
> >         e-mail
> >         > >                                 >
> >         > >                                 > yes, we do use tmpfs on
> >         both
> >         > >                                 variants but
> >         > >                                 > - delays occur with
> >         concurrent
> >         > >                                 calls > 80-100
> >         > >                                 > - cps is limited to 5-10
> >         on
> >         > >                                 debian, with centos 30 cps
> >         and more
> >         > >                                 are no problem at all
> >         > >                                 >
> >         > >                                 > also cpu load, stability
> >         and
> >         > >                                 overall performace have
> >         been much
> >         > >                                 better since using centos
> >         > >                                 >
> >         > >                                 > i just found out for me
> >         that
> >         > >                                 debian works not as good
> >         for me as
> >         > >                                 centos does.
> >         > >                                 > btw. everywhere else
> >         debian is 1st
> >         > >                                 choice (desktop, lamp, db
> >         etc.)
> >         > >                                 >
> >         > >                                 > br
> >         > >                                 >
> >         > >                                 >
> >         > >                                 > On 2011-04-21 23:04, Jay
> >         Binks
> >         > >                                 wrote:
> >         > >                                 >
> >         > >                                 >> I have no such problems
> >         on
> >         > >                                 debian .
> >         > >                                 >>
> >         > >                                 >> I use debian 5 with
> >         2.6.18 kernel
> >         > >                                 which is what Is
> >         recommended
> >         > >                                 >>
> >         > >                                 >> Are you using tmpfs ??
> >         > >                                 >>
> >         > >                                 >> Jay
> >         > >                                 >>
> >         > >                                 >>
> >         > >                                 >>
> >         > >                                 >> On 22/04/2011, at 3:26
> >         AM,
> >         > >                                 Christian
> >         > >
> >         L?schenkohl<christian.loeschenkohl at xpirio.com>  wrote:
> >         > >                                 >>
> >         > >                                 >>> hi
> >         > >                                 >>>
> >         > >                                 >>> we did use debian too
> >         and had
> >         > >                                 such performance issues
> >         (sip packet
> >         > >                                 delays, low cps).
> >         > >                                 >>> after using centos
> >         64bit (as
> >         > >                                 advised by the devs) all
> >         performance
> >         > >                                 problems are gone.
> >         > >                                 >>>
> >         > >                                 >>> br
> >         > >                                 >>>
> >         > >                                 >>> On 2011-04-21 18:24,
> >         Antonio
> >         > >                                 Teixeira wrote:
> >         > >                                 >>>
> >         > >                                 >>>> Hello List.
> >         > >                                 >>>>
> >         > >                                 >>>> I'm currently
> >         integrating an
> >         > >                                 IVR in python together
> >         with
> >         > >                                 freeswitch using
> >         mod_python and ESL
> >         > >                                 and my life has been well
> >         until ...
> >         > >                                 >>>> The flow of calls
> >         went over 80
> >         > >                                 simultaneous calls.
> >         > >                                 >>>> Now freeswitch starts
> >         sending
> >         > >                                 packets with huge delays
> >         ( even when
> >         > >                                 establishing the call ,
> >         mainly the
> >         > >                                 200 ) and firing up the
> >         IVR with
> >         > >                                 tons of delay up to 20
> >         seconds.
> >         > >                                 >>>>
> >         > >                                 >>>> So i searched the
> >         wiki forums
> >         > >                                 and mailing list:
> >         > >                                 >>>>
> >         > >                                 >>>> Put freeswitch on a
> >         diet ,
> >         > >                                 trimmed modules.conf
> >         > >                                 >>>> Played with the
> >         ulimit stuff.
> >         > >                                 >>>> Played with the IVRS
> >         to reduce
> >         > >                                 load to a minimum and i
> >         was able to
> >         > >                                 squeeze more 5 calls of
> >         performance.
> >         > >                                 >>>>
> >         > >                                 >>>> The problem is :
> >         > >                                 >>>>
> >         > >                                 >>>> Top shows
> >         > >                                 >>>> top - 16:14:33 up 35
> >         days,
> >         > >                                  8:15,  3 users,  load
> >         average:
> >         > >                                 1.92, 1.76, 1.78
> >         > >                                 >>>> Tasks: 133 total,   1
> >         running,
> >         > >                                 132 sleeping,   0 stopped,
> >         0
> >         > >                                 zombie
> >         > >                                 >>>> Cpu(s):  1.4%us,
> >          3.3%sy,  0.0%
> >         > >                                 ni, 94.6%id,  0.0%wa,
> >          0.3%hi,  0.5%
> >         > >                                 si,  0.0%st
> >         > >                                 >>>> Mem:   8193336k
> >         total,
> >         > >                                  1639156k used,  6554180k
> >         free,
> >         > >                                 177208k buffers
> >         > >                                 >>>> Swap: 19534904k
> >         total,
> >         > >                                  0k used, 19534904k free,
> >          1062272k
> >         > >                                 cached
> >         > >                                 >>>>
> >         > >                                 >>>>    PID USER      PR
> >          NI  VIRT
> >         > >                                  RES  SHR S %CPU %MEM
> >          TIME+
> >         > >                                  COMMAND
> >         > >                                 >>>> 31361 yadayada
> >          20   0
> >         > >                                  716m 164m 9628 S   73
> >          2.1
> >         > >                                 155:17.85 freeswitch
> >         > >                                 >>>>
> >         > >                                 >>>> Freeswitch goes up to
> >         150 % and
> >         > >                                 puff there goes the MOS
> >         down to 0.
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>> Some basic System
> >         Info :
> >         > >                                 >>>> Debian 6.0 ( i heard
> >         the
> >         > >                                 timming module is affected
> >         by
> >         > >                                 Debian , but if the CPU %
> >         gets lower
> >         > >                                 than 95% everything will
> >         be more
> >         > >                                 stable)
> >         > >                                 >>>> Python 2.5
> >         > >                                 >>>>
> >         > >                                 >>>> 2 x Intel(R) Xeon(R)
> >         CPU
> >         > >                                 E5506  @ 2.13GHz
> >         > >                                 >>>> 8 GB of Ram
> >         > >                                 >>>>
> >         > >                                 >>>> as you can see 94 %
> >         of the "Cpu
> >         > >                                 Power" is sleeping :\
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>> It appears freeswitch
> >         is only
> >         > >                                 capable of using let's say
> >         "one
> >         > >                                 cpu"/thread ??
> >         > >                                 >>>> Do you guys recommend
> >         simply
> >         > >                                 starting more instances or
> >         redoing
> >         > >                                 the IVR stuff.
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>> Hope you guys can
> >         help me out.
> >         > >                                 >>>>
> >         > >                                 >>>> Thanks
> >         > >                                 >>>> Ant?nio Teixeira
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >                                 >>>>
> >         > >
> >         _______________________________________________
> >         > >                                 >>>> FreeSWITCH-users
> >         mailing list
> >         > >                                 >>>>
> >         > >
> >         FreeSWITCH-users at lists.freeswitch.org
> >         > >                                 >>>>
> >         > >
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         > >                                 >>>>
> >         > >
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > >                                 >>>>
> >         http://www.freeswitch.org
> >         > >                                 >>>
> >         > >                                 >>>
> >         > >                                 >>> --
> >         > >                                 >>> Ing. Christian
> >         L?schenkohl
> >         > >                                 >>> Technische Leitung,
> >         Forschung&
> >         > >                                  Entwicklung VoIP
> >         > >                                 >>>
> >         > >                                 >>> xpirio
> >         > >                                 >>> Telekommunikation&
> >          Service GmbH
> >         > >                                 >>> Lakeside B04
> >         > >                                 >>> 9020 Klagenfurt
> >         > >                                 >>> Austria
> >         > >                                 >>>
> >         > >                                 >>> T  +43 5 77 11 - 1000
> >         > >                                 >>> F  +43 5 77 11 - 1002
> >         > >                                 >>> E
> >         > >
> >          christian.loeschenkohl at xpirio.com
> >         > >                                 >>>
> >         > >                                 >>>
> >         > >
> >         _______________________________________________
> >         > >                                 >>> FreeSWITCH-users
> >         mailing list
> >         > >                                 >>>
> >         > >
> >         FreeSWITCH-users at lists.freeswitch.org
> >         > >                                 >>>
> >         > >
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         > >                                 >>>
> >         > >
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > >                                 >>>
> >         http://www.freeswitch.org
> >         > >                                 >>
> >         > >                                 >>
> >         > >
> >         _______________________________________________
> >         > >                                 >> FreeSWITCH-users
> >         mailing list
> >         > >                                 >>
> >         > >
> >         FreeSWITCH-users at lists.freeswitch.org
> >         > >                                 >>
> >         > >
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         > >                                 >>
> >         > >
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > >                                 >>
> >         http://www.freeswitch.org
> >         > >                                 >
> >         > >                                 >
> >         > >                                 > --
> >         > >                                 > Ing. Christian
> >         L?schenkohl
> >         > >                                 > Technische Leitung,
> >         Forschung &
> >         > >                                 Entwicklung VoIP
> >         > >                                 >
> >         > >                                 > xpirio
> >         > >                                 > Telekommunikation &
> >         Service GmbH
> >         > >                                 > Lakeside B04
> >         > >                                 > 9020 Klagenfurt
> >         > >                                 > Austria
> >         > >                                 >
> >         > >                                 > T  +43 5 77 11 - 1000
> >         > >                                 > F  +43 5 77 11 - 1002
> >         > >                                 > E
> >         > >
> >          christian.loeschenkohl at xpirio.com
> >         > >                                 >
> >         > >                                 >
> >         > >
> >         _______________________________________________
> >         > >                                 > FreeSWITCH-users mailing
> >         list
> >         > >                                 >
> >         > >
> >         FreeSWITCH-users at lists.freeswitch.org
> >         > >                                 >
> >         > >
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         > >                                 >
> >         > >
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > >                                 >
> >         http://www.freeswitch.org
> >         > >                                 >
> >         > >
> >         > >
> >         _______________________________________________
> >         > >                                 FreeSWITCH-users mailing
> >         list
> >         > >
> >         FreeSWITCH-users at lists.freeswitch.org
> >         > >
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         > >
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > >                                 http://www.freeswitch.org
> >         > >
> >         > >
> >         > >
> >         > >
> >         > >
> >         _______________________________________________
> >         > >                         FreeSWITCH-users mailing list
> >         > >
> >         FreeSWITCH-users at lists.freeswitch.org
> >         > >
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         > >
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > >                         http://www.freeswitch.org
> >         > >
> >         > >
> >         > >
> >         > >
> >         > >
> >         _______________________________________________
> >         > >                 FreeSWITCH-users mailing list
> >         > >                 FreeSWITCH-users at lists.freeswitch.org
> >         > >
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         > >
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > >                 http://www.freeswitch.org
> >         > >
> >         > >
> >         > >
> >         > >
> >         > >         _______________________________________________
> >         > >         FreeSWITCH-users mailing list
> >         > >         FreeSWITCH-users at lists.freeswitch.org
> >         > >
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         > >
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > >         http://www.freeswitch.org
> >         > >
> >         > >
> >         > >
> >         > > _______________________________________________
> >         > > FreeSWITCH-users mailing list
> >         > > FreeSWITCH-users at lists.freeswitch.org
> >         > >
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         > >
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > > http://www.freeswitch.org
> >         > >
> >         > _______________________________________________
> >         > FreeSWITCH-users mailing list
> >         > FreeSWITCH-users at lists.freeswitch.org
> >         >
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         >
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > http://www.freeswitch.org
> >
> >
> >
> >         _______________________________________________
> >         FreeSWITCH-users mailing list
> >         FreeSWITCH-users at lists.freeswitch.org
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at 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
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>
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From jeff at jefflenk.com  Mon Apr 25 21:58:58 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 25 Apr 2011 10:58:58 -0700 (PDT)
Subject: [Freeswitch-users] mod_spandsp fails to cross compile
In-Reply-To: <1303752081792-6303345.post@n2.nabble.com>
References: <1303752081792-6303345.post@n2.nabble.com>
Message-ID: <1303754338745-6303446.post@n2.nabble.com>

report this to Jira

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-spandsp-fails-to-cross-compile-tp6303345p6303446.html
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From infos at madovsky.org  Mon Apr 25 22:13:37 2011
From: infos at madovsky.org (Madovsky)
Date: Mon, 25 Apr 2011 14:13:37 -0400
Subject: [Freeswitch-users] presence in profiles
Message-ID: <84FA4FE9DD1F4E51BE172B2477CC96D6@e1705>

Is it a problem to turn off presence in a FS cluster environment ?

Thanks
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From paul at cupis.co.uk  Mon Apr 25 22:20:43 2011
From: paul at cupis.co.uk (Paul Cupis)
Date: Mon, 25 Apr 2011 19:20:43 +0100
Subject: [Freeswitch-users] absolute_codec_string question
In-Reply-To: <BANLkTi=gh6AyV9C-fQaR6cY-x0=VX=XDeg@mail.gmail.com>
References: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>	<4DB59C06.3090706@cupis.co.uk>	<BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>	<4DB5A53D.50201@cupis.co.uk>
	<BANLkTi=gh6AyV9C-fQaR6cY-x0=VX=XDeg@mail.gmail.com>
Message-ID: <4DB5BB7B.4060409@cupis.co.uk>

On 25/04/11 18:08, Nicolas Brenner wrote:
> I'm trying to use the
> absolute_codec_string with originate from the console, like so:
> 
> originate
> {ignore_early_media,verbose_sdp=true,absolute_codec_string='G729,PMCU'}sofia/gateway/mygateway/444444
> &bridge(user/1001)
> 
> Paul, I am using {absolute_codec_string='G729,PCMU'}, and I get the same as
> if I don't quote the string, or if I just specify one codec:

Can you provide (on pastebin.freeswitch.org) a complete log of a call,
please?

Regards,



From nico at clickfono.com  Mon Apr 25 22:23:19 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Mon, 25 Apr 2011 14:23:19 -0400
Subject: [Freeswitch-users] Community Thank Yous
In-Reply-To: <BANLkTi=iG8PPnRZc=aFQxFhRdJsyt4QOyA@mail.gmail.com>
References: <BANLkTi=iG8PPnRZc=aFQxFhRdJsyt4QOyA@mail.gmail.com>
Message-ID: <BANLkTimpNoanPzMw=P4iDo9CgZ5e48wg+w@mail.gmail.com>

I would like to join Michael in saying thank you to everyone he named, plus
thanking Michael himself, the development team and everybody who has given
back. I know I have personally benefited from being a part of this
community. Thanks!


On Mon, Apr 25, 2011 at 1:36 PM, Michael Collins <msc at freeswitch.org> wrote:

> Happy Monday to you all!
>
> I just wanted to take a moment to say thank you to the many folks who have
> been stepping up to help out with the not-so-glamorous side of running an
> open source project: documentation and community support. We all know that
> hacking code is the fun part, but writing documentation and helping out with
> IRC and mailing list questions are often thankless tasks. I wanted to
> personally say thank you to all those who've been stepping up.
>
> Regarding documentation, I've seen dozens of new users on the wiki, each of
> whom is adding something. This is indicative of people "paying the wiki tax"
> when a question is answered via ML or IRC. So to those who have been adding
> knowledge to the wiki: thank you! It is much appreciated. Please keep up the
> good work.
>
> In the IRC channel we've been having a good number of community members
> come in and make themselves available for answering questions. This can be
> very time-consuming, so again we say thank you. The same thing goes for the
> mailing list. It takes a lot of time to read through all the posts and
> respond. The following users have been especially active in helping others
> of late and I feel it is only fair to acknowledge their fine efforts. In no
> particular order...
>
> Ken Rice
> Steven Ayre
> Jeff Link
> Avi Marcus
> Kristian Kielhofner
> Steve Underwood
> Peter Olsson
> Stefan Knobloch
> Giovanni Maruzzelli
> Christopher Rienzo
>
> Every time a community member helps another community member we grow
> stronger. Please keep up the good work!
>
> -Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From grsingh750 at gmail.com  Mon Apr 25 23:18:29 2011
From: grsingh750 at gmail.com (guru singh)
Date: Tue, 26 Apr 2011 00:48:29 +0530
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
In-Reply-To: <BANLkTi=_nMDp_kjCrDmUkfLM-tMXt+nZ1g@mail.gmail.com>
References: <4DAECCA8.1050203@gmx.net>
	<BANLkTikH9QMuhvKXqfBu78fOzqxoME3X8A@mail.gmail.com>
	<4DB328AF.8090107@telefaks.de>
	<BANLkTikz_PSaE6T99jnrJSgEFJSWBpyxvw@mail.gmail.com>
	<4DB3E075.1050202@telefaks.de>
	<BANLkTi=_nMDp_kjCrDmUkfLM-tMXt+nZ1g@mail.gmail.com>
Message-ID: <BANLkTi=aKo+E1Dj1gC2Q+OZLeegr6V0-rQ@mail.gmail.com>

Hi Peter,

Please try latest git. Moc's commit has fixed the issue.

Regards,
guru

On Sun, Apr 24, 2011 at 6:09 PM, guru singh <grsingh750 at gmail.com> wrote:
> Hi Peter,
>
> You're right. Please ignore my previous message, status 'Available (On
> Demand)' should also be fine.
> I've tried it and I see the same behavior as you. Reading the logs, I
> can see that nothing after playback gets executed once the call is
> hungup. It's not just transfer, any other application also is not
> getting executing. Something is amiss.
> Maybe moc or somebody else will point it out.
>
> Regards,
> guru
>
> On Sun, Apr 24, 2011 at 2:03 PM, Peter Steinbach <lists at telefaks.de> wrote:
>> Hello Guru,
>>
>> thanks for your hint, however this did not help.
>> The point is that according to the dialplan the agent should be transferred
>> to the same extension again an wait. In my case, when the call hanges up,
>> there is no attempt to continue with the dialplan.
>> So I expect it does not have to do with the agent's setings.
>>
>> Best regards
>> Peter
>>
>>
>> guru singh schrieb:
>>
>> Hi Peter
>>
>> Try setting the status as 'Available' instead of 'Available (On Demand)'
>> In case of 'Available (On Demand)' after the call ends, the agent's
>> status is set to 'idle', so therefore no calls are given to the
>> specific agent. I'm not too sure if this is the only change required
>> to get the behavior you expect. I've not tried it on my box yet. I can
>> only do it Monday and let you know.
>>
>> Regards,
>> guru
>>
>> On Sun, Apr 24, 2011 at 12:59 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>
>>
>> Thank you guru,
>>
>> I tried the example in the wiki. This worked.
>> I wanted the agent also to wait again with MOH after the caller hung up.
>> This did not work in my environment (Freeswitch git April 2011). The
>> agent was always hungup after the caller hung up and was not transferred
>> to the same dialplan extension again.
>> Also hangup after bridge =false did not work.
>>
>> Does this work in your environment?
>>
>> Best regards
>> Peter
>>
>>
>> guru singh schrieb:
>>
>>
>> Hi Peter,
>>
>> I've been using mod_callcenter for a while and must say it works really
>> well.
>> I just tried the uuid-standby strategy and basically it's exactly what
>> you say the asterisk thing does.
>> See the dialplan example.
>> http://wiki.freeswitch.org/wiki/Mod_callcenter#uuid-standby
>> Agent is dialing 4099 and listening to MOH. When a call arrives, it's
>> bridged directly to the agent.
>>
>> Regards
>> guru
>>
>> On Wed, Apr 20, 2011 at 5:38 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>
>>
>> Hello,
>>
>> I am trying to use the mod_callcenter functionality. This works nicely
>> so far so thank you to everybody involved for programming this nice module!
>> But I am stuck somehow with uuid-standby. Can anybody explain how
>> uuid-standby works?
>>
>> Another question: In the Asterisk based callcenter solution named
>> "Vicidial", an agent can be held permanently in a conference, waiting
>> for calls who are bridged to his uuid in the conference. Can this be
>> haviour be done with mod_callcenter?
>>
>>
>> Best regards
>> Peter
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> With kind regards
>> Peter Steinbach
>>
>> Telefaks Services GmbH
>> mailto:lists (att) telefaks.de
>> Internet: www.telefaks.de
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From nico at clickfono.com  Mon Apr 25 23:25:59 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Mon, 25 Apr 2011 15:25:59 -0400
Subject: [Freeswitch-users] absolute_codec_string question
In-Reply-To: <4DB5BB7B.4060409@cupis.co.uk>
References: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>
	<4DB59C06.3090706@cupis.co.uk>
	<BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>
	<4DB5A53D.50201@cupis.co.uk>
	<BANLkTi=gh6AyV9C-fQaR6cY-x0=VX=XDeg@mail.gmail.com>
	<4DB5BB7B.4060409@cupis.co.uk>
Message-ID: <BANLkTini7nLgyNstmf2GxzK9mcwOdG03Dw@mail.gmail.com>

Thanks, yes, here you go:

I'm making calls from the console to an X-Lite registered on extension 1001
which doesn't support G729. Calls 1 and 3 below fail because the codec
options are not supported, and even though on the third call the
absolute_codec_string variable should be 'G729,PCMU', it is not offering
both codecs.


On http://pastebin.freeswitch.org/16167 the SIP trace for:

originate {ignore_early_media,absolute_codec_string=G729}user/1001
&bridge(user/1000)

This is the SDP:

   v=0
   o=FreeSWITCH 1303734153 1303734154 IN IP4 127.0.0.1
   s=FreeSWITCH
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 24036 RTP/AVP 18 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20


On http://pastebin.freeswitch.org/16169 the SIP trace for:

originate {ignore_early_media,absolute_codec_string=PCMU}user/1001
&bridge(user/1000)

This is the SDP:

   v=0
   o=FreeSWITCH 1303733515 1303733516 IN IP4 127.0.0.1
   s=FreeSWITCH
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 24896 RTP/AVP 0 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20


On http://pastebin.freeswitch.org/16170 the SIP trace for:

originate {ignore_early_media,absolute_codec_string='G729,PCMU'}user/1001
&bridge(user/1000)

This is the SDP:

   v=0
   o=FreeSWITCH 1303733520 1303733521 IN IP4 127.0.0.1
   s=FreeSWITCH
   c=IN IP4 127.0.0.1
   t=0 0
   m=audio 25138 RTP/AVP 18 101 13
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20




On Mon, Apr 25, 2011 at 2:20 PM, Paul Cupis <paul at cupis.co.uk> wrote:

> On 25/04/11 18:08, Nicolas Brenner wrote:
> > I'm trying to use the
> > absolute_codec_string with originate from the console, like so:
> >
> > originate
> >
> {ignore_early_media,verbose_sdp=true,absolute_codec_string='G729,PMCU'}sofia/gateway/mygateway/444444
> > &bridge(user/1001)
> >
> > Paul, I am using {absolute_codec_string='G729,PCMU'}, and I get the same
> as
> > if I don't quote the string, or if I just specify one codec:
>
> Can you provide (on pastebin.freeswitch.org) a complete log of a call,
> please?
>
> 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 vetali100 at gmail.com  Mon Apr 25 23:35:13 2011
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Mon, 25 Apr 2011 22:35:13 +0300
Subject: [Freeswitch-users] absolute_codec_string question
In-Reply-To: <BANLkTini7nLgyNstmf2GxzK9mcwOdG03Dw@mail.gmail.com>
References: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>
	<4DB59C06.3090706@cupis.co.uk>
	<BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>
	<4DB5A53D.50201@cupis.co.uk>
	<BANLkTi=gh6AyV9C-fQaR6cY-x0=VX=XDeg@mail.gmail.com>
	<4DB5BB7B.4060409@cupis.co.uk>
	<BANLkTini7nLgyNstmf2GxzK9mcwOdG03Dw@mail.gmail.com>
Message-ID: <BANLkTikDOK0YJoF8s0wnrz00Ah6MXF5-Gg@mail.gmail.com>

Maybe I am wrong, but I see a typo in the following string, after "
ignore_early_media":

originate {ignore_early_media,absolute_codec_string=G729}user/1001
&bridge(user/1000)

It should be:

originate {ignore_early_media=true,absolute_codec_string=G729}user/1001
&bridge(user/1000)

Vitalie


2011/4/25 Nicolas Brenner <nico at clickfono.com>

> Thanks, yes, here you go:
>
> I'm making calls from the console to an X-Lite registered on extension 1001
> which doesn't support G729. Calls 1 and 3 below fail because the codec
> options are not supported, and even though on the third call the
> absolute_codec_string variable should be 'G729,PCMU', it is not offering
> both codecs.
>
>
> On http://pastebin.freeswitch.org/16167 the SIP trace for:
>
> originate {ignore_early_media,absolute_codec_string=G729}user/1001
> &bridge(user/1000)
>
> This is the SDP:
>
>    v=0
>    o=FreeSWITCH 1303734153 1303734154 IN IP4 127.0.0.1
>    s=FreeSWITCH
>    c=IN IP4 127.0.0.1
>    t=0 0
>    m=audio 24036 RTP/AVP 18 101 13
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>
>
> On http://pastebin.freeswitch.org/16169 the SIP trace for:
>
> originate {ignore_early_media,absolute_codec_string=PCMU}user/1001
> &bridge(user/1000)
>
> This is the SDP:
>
>    v=0
>    o=FreeSWITCH 1303733515 1303733516 IN IP4 127.0.0.1
>    s=FreeSWITCH
>    c=IN IP4 127.0.0.1
>    t=0 0
>    m=audio 24896 RTP/AVP 0 101 13
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>
>
> On http://pastebin.freeswitch.org/16170 the SIP trace for:
>
> originate {ignore_early_media,absolute_codec_string='G729,PCMU'}user/1001
> &bridge(user/1000)
>
> This is the SDP:
>
>    v=0
>    o=FreeSWITCH 1303733520 1303733521 IN IP4 127.0.0.1
>    s=FreeSWITCH
>    c=IN IP4 127.0.0.1
>    t=0 0
>    m=audio 25138 RTP/AVP 18 101 13
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
>
>
>
>
> On Mon, Apr 25, 2011 at 2:20 PM, Paul Cupis <paul at cupis.co.uk> wrote:
>
>> On 25/04/11 18:08, Nicolas Brenner wrote:
>> > I'm trying to use the
>> > absolute_codec_string with originate from the console, like so:
>> >
>> > originate
>> >
>> {ignore_early_media,verbose_sdp=true,absolute_codec_string='G729,PMCU'}sofia/gateway/mygateway/444444
>> > &bridge(user/1001)
>> >
>> > Paul, I am using {absolute_codec_string='G729,PCMU'}, and I get the same
>> as
>> > if I don't quote the string, or if I just specify one codec:
>>
>> Can you provide (on pastebin.freeswitch.org) a complete log of a call,
>> please?
>>
>> 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
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 26 00:12:15 2011
From: jgallartm at gmail.com (Javier Gallart)
Date: Mon, 25 Apr 2011 22:12:15 +0200
Subject: [Freeswitch-users] g729 annexa/annexb interoperability
Message-ID: <BANLkTi=U0RbZAkV-j26iapmBX7_vnz=s7g@mail.gmail.com>

Thanks David

this is how Cisco sees the g729 issue:
http://www.cisco.com/en/US/tech/tk1077/technologies_tech_note09186a00800b6710.shtml#g729
IIRC the cng packets have 2 bytes payload, if an endpoint does not support
annexb, what should it do about those packets?

Regards

Javi


---------- Forwarded message ----------
> From: David Ponzone <david.ponzone at ipeva.fr>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 25 Apr 2011 16:19:39 +0200
> Subject: Re: [Freeswitch-users] g729 annexa/annexb interoperability
> Javier,
>
> I dont think there is anything to transcode.
> G729 and G729B are the same, except the latter uses VAD and CNG.
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 25/04/2011 ? 13:44, Javier Gallart a ?crit :
>
> Hello all
>
> we're using a Sangoma D100 card for transcoding. Our configuration, as far
> as the codec policy is concerned, basically does this:
>
> -In the first offer to the b-leg we use the same codec list we receive
> (<action application="export"
> data="absolute_codec_string=${ep_codec_string}"/>)
> -If the call fails with status 488, we repeat the call using all the codecs
> available. This includes changing from annexb to annexa and viceversa.
>
> Let's suppose a-leg only supports g729 annexA, and B-leg only supports G729
> annexB. My understanding is that those different codec flavours don't
> interoperate; in fact I've experienced many times audio problems when trying
> to set up a call between endpoints supporting different g729 variants.
>
> In the described case, FS sends initially annexb=no, and B-leg rejects it
> with cause 488. We rebuild the offer with annexb=yes, and then the offer is
> accepted by the B-leg. Our concern is that no transcoding resources are used
> in this case, and we might run into audio problems because of that. The
> other concern is that in the answer to the A-leg (in 183 and 200),
> annexb=yes is included. I'm not sure if all devices would support different
> fmtp parameters in the offer and the answer (the RFC, as usual, won't be
> explicit about this).
>
> Is there any way to force the transcoding in a situation like the one I
> described?
>
> Thanks in advance
>
> Javi
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 nico at clickfono.com  Tue Apr 26 00:44:19 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Mon, 25 Apr 2011 16:44:19 -0400
Subject: [Freeswitch-users] absolute_codec_string question
In-Reply-To: <BANLkTikDOK0YJoF8s0wnrz00Ah6MXF5-Gg@mail.gmail.com>
References: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>
	<4DB59C06.3090706@cupis.co.uk>
	<BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>
	<4DB5A53D.50201@cupis.co.uk>
	<BANLkTi=gh6AyV9C-fQaR6cY-x0=VX=XDeg@mail.gmail.com>
	<4DB5BB7B.4060409@cupis.co.uk>
	<BANLkTini7nLgyNstmf2GxzK9mcwOdG03Dw@mail.gmail.com>
	<BANLkTikDOK0YJoF8s0wnrz00Ah6MXF5-Gg@mail.gmail.com>
Message-ID: <BANLkTik4mv7zWRndhM4_KxvvU-ybC-cLUQ@mail.gmail.com>

Oops, sorry you are right, I copied it wrong, doesn't fix it though.


On Mon, Apr 25, 2011 at 3:35 PM, Vitalie Colosov <vetali100 at gmail.com>wrote:

> Maybe I am wrong, but I see a typo in the following string, after "
> ignore_early_media":
>
> originate {ignore_early_media,absolute_codec_string=G729}user/1001
> &bridge(user/1000)
>
> It should be:
>
> originate {ignore_early_media=true,absolute_codec_string=G729}user/1001
> &bridge(user/1000)
>
> Vitalie
>
>
> 2011/4/25 Nicolas Brenner <nico at clickfono.com>
>
>> Thanks, yes, here you go:
>>
>> I'm making calls from the console to an X-Lite registered on extension
>> 1001 which doesn't support G729. Calls 1 and 3 below fail because the codec
>> options are not supported, and even though on the third call the
>> absolute_codec_string variable should be 'G729,PCMU', it is not offering
>> both codecs.
>>
>>
>> On http://pastebin.freeswitch.org/16167 the SIP trace for:
>>
>> originate {ignore_early_media,absolute_codec_string=G729}user/1001
>> &bridge(user/1000)
>>
>> This is the SDP:
>>
>>    v=0
>>    o=FreeSWITCH 1303734153 1303734154 IN IP4 127.0.0.1
>>    s=FreeSWITCH
>>    c=IN IP4 127.0.0.1
>>    t=0 0
>>     m=audio 24036 RTP/AVP 18 101 13
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-16
>>    a=ptime:20
>>
>>
>> On http://pastebin.freeswitch.org/16169 the SIP trace for:
>>
>> originate {ignore_early_media,absolute_codec_string=PCMU}user/1001
>> &bridge(user/1000)
>>
>> This is the SDP:
>>
>>    v=0
>>    o=FreeSWITCH 1303733515 1303733516 IN IP4 127.0.0.1
>>     s=FreeSWITCH
>>    c=IN IP4 127.0.0.1
>>    t=0 0
>>    m=audio 24896 RTP/AVP 0 101 13
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-16
>>    a=ptime:20
>>
>>
>> On http://pastebin.freeswitch.org/16170 the SIP trace for:
>>
>> originate {ignore_early_media,absolute_codec_string='G729,PCMU'}user/1001
>> &bridge(user/1000)
>>
>> This is the SDP:
>>
>>    v=0
>>    o=FreeSWITCH 1303733520 1303733521 IN IP4 127.0.0.1
>>    s=FreeSWITCH
>>    c=IN IP4 127.0.0.1
>>    t=0 0
>>     m=audio 25138 RTP/AVP 18 101 13
>>    a=rtpmap:101 telephone-event/8000
>>    a=fmtp:101 0-16
>>    a=ptime:20
>>
>>
>>
>>
>> On Mon, Apr 25, 2011 at 2:20 PM, Paul Cupis <paul at cupis.co.uk> wrote:
>>
>>> On 25/04/11 18:08, Nicolas Brenner wrote:
>>> > I'm trying to use the
>>> > absolute_codec_string with originate from the console, like so:
>>> >
>>> > originate
>>> >
>>> {ignore_early_media,verbose_sdp=true,absolute_codec_string='G729,PMCU'}sofia/gateway/mygateway/444444
>>> > &bridge(user/1001)
>>> >
>>> > Paul, I am using {absolute_codec_string='G729,PCMU'}, and I get the
>>> same as
>>> > if I don't quote the string, or if I just specify one codec:
>>>
>>> Can you provide (on pastebin.freeswitch.org) a complete log of a call,
>>> please?
>>>
>>> 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
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 sam.oredoyin at gmail.com  Tue Apr 26 00:47:14 2011
From: sam.oredoyin at gmail.com (Sam Oredoyin)
Date: Mon, 25 Apr 2011 21:47:14 +0100
Subject: [Freeswitch-users] Forwarding Remote-Party-ID from Callee to Caller
Message-ID: <4db5deb8.0d26e30a.4771.11a7@mx.google.com>

Hi All,

I would like to pass the Remote-Party-ID field of the called party (sent by
the callee in the Ringing & OK messages when answering the call) back to the
originator's phone. 

I use the dialplan action ["set" data="sip_callee_id_name] as below but this
only gives static information, we want to use the information passed back by
the actual callee phone.

The destination sends the Remote-Party-ID in the Ringing and OK replies, but
they are not relayed to the original caller ("Outbound call" is sent
instead). 

Is it at all possible to keep the SIP headers from the B-Leg of a call so we
can pick the value from there, although it will be better if FS forwards the
value rather than replacing it with "Outbound call".

I know someone asked this question a couple of months ago but didn't seem to
get the resolution.

I'm running :>     FreeSWITCH Version 1.0.head (git-4c435ec 2011-03-14
11-54-08 -0500)

Regards
Sam






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From moises.silva at gmail.com  Tue Apr 26 02:16:20 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 26 Apr 2011 00:16:20 +0200
Subject: [Freeswitch-users] FreeTDM disconnect supervision
In-Reply-To: <BANLkTineb1_p9rd1mm2fism=aNbyo6gY5g@mail.gmail.com>
References: <BANLkTikwxtkATFaK0jQJhpnsvgOhxJp2RQ@mail.gmail.com>
	<BANLkTinRmBORb9zO0yvt22sWW8JE2E7g3A@mail.gmail.com>
	<BANLkTimk_=m2GVZB=c7PtxqZoBJWR0ELMA@mail.gmail.com>
	<BANLkTikNrdU0aFHUxvGJ=sSfPD=_SzZgJg@mail.gmail.com>
	<BANLkTineb1_p9rd1mm2fism=aNbyo6gY5g@mail.gmail.com>
Message-ID: <BANLkTinr8XVEiFnzRhcSjM=w_DsP_zAtXg@mail.gmail.com>

Hello Luis,

The syntax used is explained here: http://wiki.freeswitch.org/wiki/TGML

I've not tested myself the tone detection, but I believe it should work in
FXO lines (the only type of lines where currently FreeTDM enables progress
tone detection).

Defaults are found in src/ftdm_io.c:828

Dial: %(1000,0,350,440)
Ring: %(2000,4000,440,480)
Busy: %(500,500,480,620)
Attn: %(100,100,1400,2060,2450,2600)


The first 2 numbers are always tone on/off values in milliseconds. The
remaining numbers are the frequencies to use, in the Attn tone there are 4
frequencies involved.

FreeTDM uses libteletone library created by Tony for tone detection and
generation.

Michael, is there any reason for you to think he did not complete the
cadence detection in libteletone?

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 Mon, Apr 25, 2011 at 4:30 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

> Hi all,
>
> I am updating the thread as per Moises Silva's request so he can answer for
> posterity. Not just to bump the thread.
>
> Thx.
>
> Hello Luis,
>
> Can you point to the mailing list conversation and please update the thread
> because if I deleted the thread in my mail already I have no way to reply
> unless someone else sends a new reply. I'd rather reply there to leave the
> response on the internet available for anyone. If you don't have it please
> create a new thread in freeswitch-users and I'll reply there.
>
> On Mon, Apr 18, 2011 at 5:38 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Generating a tone w/ cadence is a piece of cake. *Detecting* a tone and
>> cadence is a bit trickier. Nothing in FreeTDM (that I'm aware of) can detect
>> both tone *and* cadence, which is why it works great in the US (which uses a
>> combination of tones) as opposed to Mexico (which uses, I think 425Hz for
>> everything).
>>
>> I'd ask Moises Silva if there's any plans to add the cadence detection.
>>
>> -MC
>>
>>
>> On Mon, Apr 18, 2011 at 4:27 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>>
>>> Yes guru,
>>>
>>> The point is that using tone_detect I am not able to specify cadence
>>> either.
>>>
>>> I need to be able to detect a tone that plays 450Hz for 330ms and then
>>> silence for 330ms.
>>>
>>> Strange thing is that on tones.conf generate tones do seem to specify
>>> cadence such as US busy
>>>
>>> v=-7;(500,500,420,680)
>>>
>>> I would interpret 420Hz for .5s and 680Hz for .5s
>>>
>>> Is that how it is to be interpreted??
>>>
>>>
>>> TIA
>>>
>>>
>>> On Mon, Apr 18, 2011 at 4:48 PM, guru singh <grsingh750 at gmail.com>wrote:
>>>
>>>> Hi Luis,
>>>>
>>>> You could try the dialplan application tone_detect to detect the busy
>>>> tone and hangup.
>>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
>>>>
>>>> guru
>>>>
>>>> On Tue, Apr 19, 2011 at 3:34 AM, Luis F Urrea <lfurrea at gmail.com>
>>>> wrote:
>>>> > Hello,
>>>> > According to what I have found in regards the tones used for signaling
>>>> on
>>>> > FreeTDM, it seems that anything set through DAHDI is ignored and specs
>>>> from
>>>> > tones.conf are used instead.
>>>> > However I have not been able to properly detect a busy tone to be able
>>>> to
>>>> > set an FXO back on hook once a busy tone is sniffed.
>>>> > tones.conf has the following references for busy tones [us]:
>>>> > generate-busy => v=-7;%(500,500,480,620)
>>>> > detect-busy=> 480,620
>>>> > I would like to know if detect-busy takes into account the time frame
>>>> that
>>>> > the frequencies are on (played) or off (not being played) and if there
>>>> is
>>>> > somehow a way to specify this. Is there any default value for time on
>>>> and
>>>> > time off?
>>>> > I appreciate your help!
>>>> > 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
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gosha at inbox.ee  Mon Apr 25 23:55:05 2011
From: gosha at inbox.ee (Georgy Goshin)
Date: Mon, 25 Apr 2011 22:55:05 +0300
Subject: [Freeswitch-users] SIP router/switch only configuratoin advice?
Message-ID: <0f0501cc0382$ac845fd0$058d1f70$@inbox.ee>

Hello!

I need some advice from experienced FreeSWITCH users. Please review my needs
and advise me some configuration samples or adapted configuration  guides.
I'd like to use the FS in two roles - a SIP router for my PBXes and as a
wholesale switch. Both configurations are almost the same in my case. The
desired configuration should be allow me to do the following:

route calls based on A num, B num, SRC IP (Mera MVTS is an example of what I
want in simple calls routing).

If a more complex configuration that can do dynamic routing based on price,
acd/asr and prepaid/credit-limit billing is possible too, I'd be glad to see
some examples too.


Thanks in advance,
G. Goshin.



From grw.freeswitch at gmail.com  Tue Apr 26 00:55:46 2011
From: grw.freeswitch at gmail.com (Geovani Ricardo Wiedenhoft)
Date: Mon, 25 Apr 2011 17:55:46 -0300
Subject: [Freeswitch-users] Receiving SMS some alert?
In-Reply-To: <BANLkTimmcXeyaHacWZZEEUs+kFLbRSjUJw@mail.gmail.com>
References: <BANLkTimHvjOCLNUXUOSO9K=74=mno2ogTw@mail.gmail.com>
	<BANLkTim2HO4OhDMSo2yRPWxOfFyQWyCK6Q@mail.gmail.com>
	<BANLkTimmcXeyaHacWZZEEUs+kFLbRSjUJw@mail.gmail.com>
Message-ID: <BANLkTikKhcgh1VvugMzrzS3Hesv4BmPuGw@mail.gmail.com>

Hello.

 Enable logs to disk.

<param name="log-to-disk" value="debugging"/>

So, restart the Freeswitch or in this case o mod_khomp.


The logs always help us.


Probably the context was not set.

You need to configure sms incoming context in khomp.conf.xml

<param name="context-gsm-sms" value="khomp-sms-DD-CC"/>

So, restart the Freeswitch.


Regards,
Geovani Ricardo Wiedenhoft



2011/4/25 Rodrigo Ferrari <rodrigo.ferrari at cellmidia.com.br>:
> Obrigado Jo?o!
>
> Eu tentei o suporte mas n?o obtive a resposta do que perguntei, tentei
> novamente e estou aguardando devido ao feriado mesmo, por?m n?o consigo
> ficar parado, fico tentando buscar solu??o por todos os lugares, meu
> problema ? na quest?o de receber o SMS, pego meu celular e envio uma
> mensagem para um chip da placa, com o console do freeswitch aberto, n?o
> recebo nenhum evento de entrada e nem o dialplan ? executado, se ent?o eu
> com meu celular ligo, no momento o freeswitch apita no console a entrada de
> uma chamada e executa o dialplan, ent?o acredito estar esquecendo de
> configurar algo para lidar com as mensagens que entram na placa.
>
> Obrigado!
>
> Abra?os.
>
> Em 25 de abril de 2011 11:07, Jo?o Mesquita <jmesquita at freeswitch.org>
> escreveu:
>>
>> Rodrigo,
>> Have you tried their official support? They do provide tech support and
>> they officially support FreeSWITCH. I am sure if you did, they did not reply
>> because of the holiday.
>> suporte (at) khomp.com.br should be just fine.
>> Regards,
>> Jo?o Mesquita
>>
>>
>>
>> On Mon, Apr 25, 2011 at 9:45 AM, Rodrigo Ferrari
>> <rodrigo.ferrari at cellmidia.com.br> wrote:
>>>
>>> Hello friends,
>>>
>>> I just bought a Khomp board to send and receive SMS, the problem is, I'm
>>> not receiving the messages, it's not marking at the console a message
>>> incoming. Is this some configuration that I miss?
>>>
>>> Thanks, best regards.
>>> Rodrigo Ferrari.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From nboric at yx.cl  Tue Apr 26 01:00:33 2011
From: nboric at yx.cl (Neven Boric)
Date: Mon, 25 Apr 2011 18:00:33 -0300
Subject: [Freeswitch-users] mod_crd_sqlite entry limit and sqlite segfaults
	on triggers
Message-ID: <4DB5E0F1.4050602@yx.cl>

Hi,

By looking at the code, I believe there is no limit on the number of 
rows mod_cdr_sqlite will add to the cdr table. This could lead to 
(eventually) eating all RAM if you DB is in tmpfs or eating all disk 
space if it is stored on disk. Is this correct or am I missing something?

To correct this I tried to add a trigger to the table along the lines of:

CREATE TRIGGER IF NOT EXISTS CDR_LIMIT AFTER INSERT ON cdr
BEGIN
     DELETE FROM crd WHERE rowid <= (SELECT MAX(rowid) from cdr) - 100000;
END

and no matter how I try to add the trigger, I get a segmentation fault 
when the code calls sqlite3_exec in switch_core_db_exec.

Currently I'm blaiming SQLite, as there is a somewhat similar issue 
involving triggers reported for versions older than 3.5.4 (FS is using 
3.3.13)

http://sqlite.org/cvstrac/wiki?p=DatabaseCorruption

Of course I could just execute the delete statement manually after every 
insert, but it is uglier (more costly, non atomic).

Has anyone any experience using triggers with the sqlite version 
included in FS? Or has anyone tried other solution to limit the number 
on entries on the cdr table?

Thanks in advance
Neven Boric




From lists at telefaks.de  Tue Apr 26 02:37:14 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Tue, 26 Apr 2011 00:37:14 +0200
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
In-Reply-To: <BANLkTi=aKo+E1Dj1gC2Q+OZLeegr6V0-rQ@mail.gmail.com>
References: <4DAECCA8.1050203@gmx.net>	<BANLkTikH9QMuhvKXqfBu78fOzqxoME3X8A@mail.gmail.com>	<4DB328AF.8090107@telefaks.de>	<BANLkTikz_PSaE6T99jnrJSgEFJSWBpyxvw@mail.gmail.com>	<4DB3E075.1050202@telefaks.de>	<BANLkTi=_nMDp_kjCrDmUkfLM-tMXt+nZ1g@mail.gmail.com>
	<BANLkTi=aKo+E1Dj1gC2Q+OZLeegr6V0-rQ@mail.gmail.com>
Message-ID: <4DB5F79A.4000808@telefaks.de>

Thanks Guru and Moc,

I will be able to test it on Thursday.

Best
regards
peter

guru singh schrieb:
> Hi Peter,
>
> Please try latest git. Moc's commit has fixed the issue.
>
> Regards,
> guru
>
> On Sun, Apr 24, 2011 at 6:09 PM, guru singh <grsingh750 at gmail.com> wrote:
>   
>> Hi Peter,
>>
>> You're right. Please ignore my previous message, status 'Available (On
>> Demand)' should also be fine.
>> I've tried it and I see the same behavior as you. Reading the logs, I
>> can see that nothing after playback gets executed once the call is
>> hungup. It's not just transfer, any other application also is not
>> getting executing. Something is amiss.
>> Maybe moc or somebody else will point it out.
>>
>> Regards,
>> guru
>>
>> On Sun, Apr 24, 2011 at 2:03 PM, Peter Steinbach <lists at telefaks.de> wrote:
>>     
>>> Hello Guru,
>>>
>>> thanks for your hint, however this did not help.
>>> The point is that according to the dialplan the agent should be transferred
>>> to the same extension again an wait. In my case, when the call hanges up,
>>> there is no attempt to continue with the dialplan.
>>> So I expect it does not have to do with the agent's setings.
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>> guru singh schrieb:
>>>
>>> Hi Peter
>>>
>>> Try setting the status as 'Available' instead of 'Available (On Demand)'
>>> In case of 'Available (On Demand)' after the call ends, the agent's
>>> status is set to 'idle', so therefore no calls are given to the
>>> specific agent. I'm not too sure if this is the only change required
>>> to get the behavior you expect. I've not tried it on my box yet. I can
>>> only do it Monday and let you know.
>>>
>>> Regards,
>>> guru
>>>
>>> On Sun, Apr 24, 2011 at 12:59 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>>
>>>
>>> Thank you guru,
>>>
>>> I tried the example in the wiki. This worked.
>>> I wanted the agent also to wait again with MOH after the caller hung up.
>>> This did not work in my environment (Freeswitch git April 2011). The
>>> agent was always hungup after the caller hung up and was not transferred
>>> to the same dialplan extension again.
>>> Also hangup after bridge =false did not work.
>>>
>>> Does this work in your environment?
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>> guru singh schrieb:
>>>
>>>
>>> Hi Peter,
>>>
>>> I've been using mod_callcenter for a while and must say it works really
>>> well.
>>> I just tried the uuid-standby strategy and basically it's exactly what
>>> you say the asterisk thing does.
>>> See the dialplan example.
>>> http://wiki.freeswitch.org/wiki/Mod_callcenter#uuid-standby
>>> Agent is dialing 4099 and listening to MOH. When a call arrives, it's
>>> bridged directly to the agent.
>>>
>>> Regards
>>> guru
>>>
>>> On Wed, Apr 20, 2011 at 5:38 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>
>>>
>>> Hello,
>>>
>>> I am trying to use the mod_callcenter functionality. This works nicely
>>> so far so thank you to everybody involved for programming this nice module!
>>> But I am stuck somehow with uuid-standby. Can anybody explain how
>>> uuid-standby works?
>>>
>>> Another question: In the Asterisk based callcenter solution named
>>> "Vicidial", an agent can be held permanently in a conference, waiting
>>> for calls who are bridged to his uuid in the conference. Can this be
>>> haviour be done with mod_callcenter?
>>>
>>>
>>> Best regards
>>> Peter
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> With kind regards
>>> Peter Steinbach
>>>
>>> Telefaks Services GmbH
>>> mailto:lists (att) telefaks.de
>>> Internet: www.telefaks.de
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>       
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 elijah at crankenstein.com  Tue Apr 26 03:36:29 2011
From: elijah at crankenstein.com (elijah)
Date: Mon, 25 Apr 2011 16:36:29 -0700
Subject: [Freeswitch-users] attended transfer to gateway
In-Reply-To: <BANLkTin8Vdg=a3em_2O1bkX3C5qoGLCBBg@mail.gmail.com>
References: <BANLkTinB6uFAWDnq-Ot1tiSPi99pu0Oruw@mail.gmail.com>
	<BANLkTikkF95DCGgpQJ-kKCR-U_=X7fF5+g@mail.gmail.com>
	<BANLkTinAXp_u4+0OwNjY5dEDH5YHmFR-fw@mail.gmail.com>
	<65A0D45D-0666-493C-B53A-D9DC882EE77C@freeswitch.org>
	<BANLkTin8Vdg=a3em_2O1bkX3C5qoGLCBBg@mail.gmail.com>
Message-ID: <BANLkTi=u9d5NWeO=Z1Rtnn_7+qAqb1cuKQ@mail.gmail.com>

So I'm at the limit of my modest capabilities for troubleshooting this
further. There is a workaround I am thinking of by which the documented
3-way conferencing feature of att-xfer could be simulated using a full-on
conference bridge and if this is the end of any suggestions I receive on
this thread I will pursue it. I have implemented att-xfer in a manner nearly
identical to the online documentation and default config as is practical and
within my understanding. I have documented here my configuration and
corresponding logs in hope that someone else had encountered a similar
problem and could advise me of a solution. Hopefully my experience does not
indicate a larger issue within the att-xfer module, but if it does I hope
the following documentation is useful.

I understand FreeSwitch is an enormous commitment for everyone who has
contributed to source and I hope to be in a position to personally
contribute in the future whether with my own time or financially. If I can
help now to get this thread resolved and prevent someone else from having an
implementation issue please let me know.

thanks,
elijah

On Thu, Apr 21, 2011 at 10:15 AM, elijah <elijah at crankenstein.com> wrote:

>         <action application="bind_meta_app" data="1 b s
> execute_extension::ax XML callsdirect"/>
>
>     <extension name="bind_receiver">
>       <condition field="destination_number" expression="^ax$">
>         <action application="answer"/>
>         <action application="read" data="10 10
> 'tone_stream://%(10000,0,350,440)' attxfer_callthis 30000 *"/>
>         <action application="execute_extension" data="attended_xfer XML
> callsdirect"/>
>       </condition>
>     </extension>
>
>     <extension name="attended_xfer">
>       <condition field="destination_number" expression="^attended_xfer$"/>
>       <condition field="${attxfer_callthis}" expression="^(\d+)$">
>         <action application="set" data="continue_on_fail=true"/>
>         <action application="set" data="origination_cancel_key=#"/>
>          <action application="att_xfer"
> data="sofia/gateway/onesource/${attxfer_callthis}"/>
>         <anti-action application="eval" data="cancel transfer"/>
>       </condition>
>     </extension>
>
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From steveu at coppice.org  Tue Apr 26 03:50:01 2011
From: steveu at coppice.org (Steve Underwood)
Date: Tue, 26 Apr 2011 07:50:01 +0800
Subject: [Freeswitch-users] mod_spandsp fails to cross compile
In-Reply-To: <1303752081792-6303345.post@n2.nabble.com>
References: <1303752081792-6303345.post@n2.nabble.com>
Message-ID: <4DB608A9.8060105@coppice.org>

On 04/26/2011 01:21 AM, mazilo wrote:
> I don't remember when this problem started; however, I have been doing a git
> pull every time trying to re-compile my local FS source for the past two
> weeks and always ended up with the same error messages on mod_spandsp
> compilation when trying to cross compile FS as shown
> http://pastebin.com/7uKGPWxQ here  on line #13 (mod_spandsp_fax.c:1656:13:
> error: implicit declaration of function 't38_gateway_rx_fillin').
Perhaps something is messed up with your source tree, because when that 
call was added to mod_spandsp_fax.c, the corresponding function 
definition was added to the included spandsp, and the appropriate 
declaration of the function was added to 
libs/spandsp/src/spandsp/t38_gateway.h. If you can't find that 
declaration, your source tree is not really up to date. If you can find 
it, something is probably wrong with your cross-compile setup, such that 
you are not using in the version of spandsp that comes with Freeswitch.

Steve



From fieldpeak at gmail.com  Tue Apr 26 05:25:57 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 26 Apr 2011 09:25:57 +0800
Subject: [Freeswitch-users] config ramdisk on CentOS 5.5
In-Reply-To: <E826C94A3C6BE44198FAF4D66B5AB4D814A48A9240@VA3DIAXVS351.RED001.local>
References: <BANLkTim4SX5JTB0-h=JaMsoyBtyW_wbEwA@mail.gmail.com>
	<C9DA7C39.1453D%krice@freeswitch.org>
	<BANLkTi=aDMwRvUv6EdNG6Fx-ctaYPZ+nPA@mail.gmail.com>
	<E826C94A3C6BE44198FAF4D66B5AB4D814A48A9240@VA3DIAXVS351.RED001.local>
Message-ID: <BANLkTi=78LhWD7yTtNa2FLJwft25iRjoFw@mail.gmail.com>

thanks Robert, understood.

Regards,
Charles

? 2011-4-26 ??12:21?"Robert Hadley" <robert.hadley at teotech.com>???
>
> On CentOS the mount command executes the /etc/fstab file, to make the
tmpfs without rebooting try:
> mount -a
>
> -Robert
>
> -----Original Message-----
> From: fieldpeak [mailto:fieldpeak at gmail.com]
> Sent: Monday, April 25, 2011 12:04 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] config ramdisk on CentOS 5.5
>
> thanks Ken. I noticed the voice mail issue when using tmpfs, thanks again.
>
> Regards,
> Charles
>
> On 4/25/11, Ken Rice <krice at freeswitch.org> wrote:
>> That is correct
>>
>> Keep in mind that in using a ramdrive for freeswitch/db if you are
>> running voicemail you?ll need to look at mechanismd for backing up its
>> db
>>
>> K
>>
>>
>> On 4/25/11 12:42 AM, "fieldpeak" <fieldpeak at gmail.com> wrote:
>>
>>> Hi Ken,
>>>
>>> Thanks for your reply.
>>>
>>> if my below understanding correct?
>>> if i add below to the end of /etc/fstab, tmpfs
>>> /usr/local/freeswitch/db tmpfs defaults 0 0
>>>
>>> and then reboot machine, the system will auto mount the tmpfs.
>>> i don't need run mannually mount /usr/local/freeswitch/db before
>>> start freeswitch...
>>>
>>> Thanks.
>>>
>>> Regards,
>>> Charles
>>>
>>> 2011/4/23 Ken Rice <krice at freeswitch.org>
>>>> Tmpfs is not a program... Read that page a little closer... That?s
>>>> particular line is for your fstab...
>>>>
>>>> If you want to mount it from the command line its mount ?o tmpfs
>>>> tmpfs /usr/local/freeswitch/db
>>>>
>>>> K
>>>>
>>>>
>>>>
>>>> On 4/22/11 10:50 PM, "fieldpeak" <fieldpeak at gmail.com
>>>> <http://fieldpeak at gmail.com> > wrote:
>>>>
>>>>> i'm trying tuning the FS to max performance on centos 5.5, and
>>>>> referred to
>>>>> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurati
>>>>> ons#FreeS
>>>>> WITCH.27s_core.db_I.2FO_bottleneck
>>>>>
>>>>> #1,  i configure the DB of FS to ramdisk ,
>>>>>
>>>>> when i run "tmpfs /opt/freeswitch/db tmpfs defaults 0 0", it output:
>>>>> "-bash:
>>>>> tmpfs: command not found"
>>>>>
>>>>> #2,  i run "ethtool -g eth0", the output is below, what value i
>>>>> should config for RX and TX for max performance...
>>>>> Ring parameters for eth0:
>>>>> Pre-set maximums:
>>>>> RX:             4096
>>>>> RX Mini:        0
>>>>> RX Jumbo:       0
>>>>> TX:             4096
>>>>> Current hardware settings:
>>>>> RX:             256
>>>>> RX Mini:        0
>>>>> RX Jumbo:       0
>>>>> TX:             256
>>>>>
>>>>> Apprecited if anyone help how to configure it... thanks!
>>>>>
>>>>> Regards,
>>>>> Charles
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> 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-u
>>>> sers
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us
>>> ers
>>> http://www.freeswitch.org
>>
>>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 philippe at ppmt.org  Tue Apr 26 05:32:44 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Mon, 25 Apr 2011 21:32:44 -0400
Subject: [Freeswitch-users] call not connecting sometime
In-Reply-To: <2A5992B4-AABE-4041-A74E-4C7A956C4BD3@ipeva.fr>
References: <4DB47783.5090507@ppmt.org>
	<2A5992B4-AABE-4041-A74E-4C7A956C4BD3@ipeva.fr>
Message-ID: <4DB620BC.7020202@ppmt.org>

by provider you mean if I use an external company to forward the call?

Yes there is.

I am not sure if I can give name of company so I won't but it is one
of those company where you buy some credit and they give you 60 days
free before starting using the credit

Of course I can't rule out that it is on their side as most of the time
it works and other it won't but
when I check my call logs on their website it shows that the call was
process by them as successful

The only issue is that the call  doesn't connect fully (the other side
say they answer but can't hear anything
while on my side it is still ringing)

I guess that the trace is not showing anything? is there more trace I
could activate for the next time it occurs?

For the moment I have deactivated the DTMF action as recommended by
Vitalie as I don't think we will
need it for these type of calls ( I am sure my wife will tell if that is
not true :) )

regards

/Philippe

On 11-04-25 10:15 AM, David Ponzone wrote:
> Phil,
>
> is there a provider involved ?
> Perhaps it's on their side ?
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ? l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
> n'?tes pas destinataire de ce message, merci de le d?truire
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 24/04/2011 ? 21:18, Philippe Le Toquin a ?crit :
>
>> Hello,
>>
>> I had the problem a few weeks ago but after a reinstall (unrelated to
>> that) the problem had gone so I put it
>> done as my messing up my system at the time!
>>
>> But today I had the same issue again. When I call sometime the call
>> is not going through completely
>>
>> The symptom on my side are that nothing happens (no ring tone) on the
>> other they say that the phones
>> rings but when they pick up the phone they can't hear anything.
>>
>> Below is a siptrace where I changed the number and IP so I hope I
>> didn't mess it up too much.
>>
>> http://pastebin.freeswitch.org/16165
>>
>> Can someone let me know if they see something wrong? I tried to
>> understand but it is beyond me :(
>>
>> Regards
>>
>> /Philippe
>> <0x1A0BDC2B.asc>_______________________________________________
>> 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 jeff at jefflenk.com  Tue Apr 26 06:21:18 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Mon, 25 Apr 2011 19:21:18 -0700 (PDT)
Subject: [Freeswitch-users] mod_spandsp fails to cross compile
In-Reply-To: <4DB608A9.8060105@coppice.org>
References: <1303752081792-6303345.post@n2.nabble.com>
	<4DB608A9.8060105@coppice.org>
Message-ID: <1303784478758-6304409.post@n2.nabble.com>

Thanks Steve for your clarification on that I didnt look close enough.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-spandsp-fails-to-cross-compile-tp6303345p6304409.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rogelio.perez at gmail.com  Tue Apr 26 08:24:55 2011
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Tue, 26 Apr 2011 01:24:55 -0300
Subject: [Freeswitch-users] mod_nibblebill updating 3 times
In-Reply-To: <BANLkTikgTPekcDf3LtaN8DRPfoDq99dGNg@mail.gmail.com>
References: <C8A89431-F467-43B0-8D77-480DD2A1F4B4@gmail.com>
	<BANLkTikgTPekcDf3LtaN8DRPfoDq99dGNg@mail.gmail.com>
Message-ID: <93DC24C0-3D80-454C-945F-AC8EE790FAC1@gmail.com>

Hello, I'm using mod_nibblebill to bill my calls only for leg b and heartbeats set to 'off'.
After the hangup I see 3 different MySQL updates from nibblebill instead of just one.

This is my dial plan:

	<extension name="TEST">
        	<condition field="destination_number" expression="^(1\d+)$">
                	<action application="set" data="dialed_extension=$1"/>
                        <action application="export" data="dialed_extension=$1"/>
                        <action application="set" data="hangup_after_bridge=true"/>
                        <action application="set" data="continue_on_fail=true"/>
			<action application="bridge" data="{nibble_rate=5,nibble_account=13}sofia/gateway/xxxxxxx/$1"/>
                          <action application="hangup"/>
		</condition>
	</extension>

...this is the MySQL log:

		68080 Query	select 1
		68080 Query	UPDATE web_account SET balance=balance-0.343372 WHERE id='13'
		68080 Query	select 1
		68080 Query	SELECT balance AS nibble_balance FROM web_account WHERE id='13'
		68080 Query	select 1
		68080 Query	UPDATE web_account SET balance=balance-0.001177 WHERE id='13'
		68080 Query	select 1
		68080 Query	SELECT balance AS nibble_balance FROM web_account WHERE id='13'
		68080 Query	select 1
		68080 Query	UPDATE web_account SET balance=balance-0.001197 WHERE id='13'
		68080 Query	select 1
		68080 Query	SELECT balance AS nibble_balance FROM web_account WHERE id='13'


...and here's the FS log: http://pastebin.freeswitch.org/16175 

The xml_cdr is saved after the second update, so the nibble_total_billed is different than the real total value debited from the account balance.

Is this normal? 
Shouldn't nibblebill just do one update after the hangup?

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From rogelio.perez at gmail.com  Tue Apr 26 08:37:39 2011
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Tue, 26 Apr 2011 01:37:39 -0300
Subject: [Freeswitch-users] mod_nibblebill updates
In-Reply-To: <BANLkTikgTPekcDf3LtaN8DRPfoDq99dGNg@mail.gmail.com>
References: <C8A89431-F467-43B0-8D77-480DD2A1F4B4@gmail.com>
	<BANLkTikgTPekcDf3LtaN8DRPfoDq99dGNg@mail.gmail.com>
Message-ID: <A50B3793-9A5B-4F2F-A24F-3692D728B104@gmail.com>

Hello, I'm using mod_nibblebill to bill my calls only for leg b and heartbeats set to 'off'.
After the hangup I see 3 different MySQL updates from nibblebill instead of just one.

This is my dial plan:

	<extension name="TEST">
        	<condition field="destination_number" expression="^(1\d+)$">
                	<action application="set" data="dialed_extension=$1"/>
                        <action application="export" data="dialed_extension=$1"/>
                        <action application="set" data="hangup_after_bridge=true"/>
                        <action application="set" data="continue_on_fail=true"/>
			<action application="bridge" data="{nibble_rate=5,nibble_account=13}sofia/gateway/xxxxxxx/$1"/>
                          <action application="hangup"/>
		</condition>
	</extension>

...this is the MySQL log:

		68080 Query	select 1
		68080 Query	UPDATE web_account SET balance=balance-0.343372 WHERE id='13'
		68080 Query	select 1
		68080 Query	SELECT balance AS nibble_balance FROM web_account WHERE id='13'
		68080 Query	select 1
		68080 Query	UPDATE web_account SET balance=balance-0.001177 WHERE id='13'
		68080 Query	select 1
		68080 Query	SELECT balance AS nibble_balance FROM web_account WHERE id='13'
		68080 Query	select 1
		68080 Query	UPDATE web_account SET balance=balance-0.001197 WHERE id='13'
		68080 Query	select 1
		68080 Query	SELECT balance AS nibble_balance FROM web_account WHERE id='13'


...and here's the FS log: http://pastebin.freeswitch.org/16175 

The xml_cdr is saved after the second update, so the nibble_total_billed is different than the real total value debited from the account balance.

Is this normal? 
Shouldn't nibblebill just do one update after the hangup?

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From u2nsam at gmail.com  Tue Apr 26 09:05:38 2011
From: u2nsam at gmail.com (Sam)
Date: Tue, 26 Apr 2011 10:35:38 +0530
Subject: [Freeswitch-users] answering machine detection
Message-ID: <BANLkTinOdcB0haPwgf=E-oPX=_f-vOGvwA@mail.gmail.com>

Hello,



Currently i see there is a module for voicemail detection where in it
detects the beep,
is there a module for detection of answering machine ( automated voice IVRS
),
or a way to integrate to FS or a code to modify to detect automated voice or
IVRs ?




Regards
Sam
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From vetali100 at gmail.com  Tue Apr 26 09:29:00 2011
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Tue, 26 Apr 2011 08:29:00 +0300
Subject: [Freeswitch-users] call not connecting sometime
In-Reply-To: <4DB620BC.7020202@ppmt.org>
References: <4DB47783.5090507@ppmt.org>
	<2A5992B4-AABE-4041-A74E-4C7A956C4BD3@ipeva.fr>
	<4DB620BC.7020202@ppmt.org>
Message-ID: <BANLkTimkGJ2sqVcyAu7HMr9JS2=qToQwFQ@mail.gmail.com>

Hi Philippe,

So, you have deactivated the DTMF action, and ... did it help? :)



2011/4/26 Philippe Le Toquin <philippe at ppmt.org>

>  by provider you mean if I use an external company to forward the call?
>
> Yes there is.
>
> I am not sure if I can give name of company so I won't but it is one
> of those company where you buy some credit and they give you 60 days free
> before starting using the credit
>
> Of course I can't rule out that it is on their side as most of the time it
> works and other it won't but
> when I check my call logs on their website it shows that the call was
> process by them as successful
>
> The only issue is that the call  doesn't connect fully (the other side say
> they answer but can't hear anything
> while on my side it is still ringing)
>
> I guess that the trace is not showing anything? is there more trace I could
> activate for the next time it occurs?
>
> For the moment I have deactivated the DTMF action as recommended by Vitalie
> as I don't think we will
> need it for these type of calls ( I am sure my wife will tell if that is
> not true :) )
>
> regards
>
> /Philippe
>
>
> On 11-04-25 10:15 AM, David Ponzone wrote:
>
> Phil,
>
>  is there a provider involved ?
> Perhaps it's on their side ?
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
>  Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 24/04/2011 ? 21:18, Philippe Le Toquin a ?crit :
>
>  Hello,
>
> I had the problem a few weeks ago but after a reinstall (unrelated to that)
> the problem had gone so I put it
> done as my messing up my system at the time!
>
> But today I had the same issue again. When I call sometime the call is not
> going through completely
>
> The symptom on my side are that nothing happens (no ring tone) on the other
> they say that the phones
> rings but when they pick up the phone they can't hear anything.
>
> Below is a siptrace where I changed the number and IP so I hope I didn't
> mess it up too much.
>
> http://pastebin.freeswitch.org/16165
>
> Can someone let me know if they see something wrong? I tried to understand
> but it is beyond me :(
>
> Regards
>
> /Philippe
>  <0x1A0BDC2B.asc>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From michel at arneill-py.sacramento.ca.us  Tue Apr 26 09:36:56 2011
From: michel at arneill-py.sacramento.ca.us (Michel Py)
Date: Mon, 25 Apr 2011 22:36:56 -0700
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
	with productivity license.
In-Reply-To: <BANLkTi=o1nUa70X_PHU51M_TTGTfnSSdKg@mail.gmail.com>
References: <471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local><BANLkTi=XmNDV51=xoNsViHUYaUZeyMpz_A@mail.gmail.com><471D76419F9EF642962323D13DF1DF69011E58@newserver.arneill-py.local>
	<BANLkTi=o1nUa70X_PHU51M_TTGTfnSSdKg@mail.gmail.com>
Message-ID: <471D76419F9EF642962323D13DF1DF69011E59@newserver.arneill-py.local>

Christian,

> Christian Benke wrote:
> I don't think this is the cause of the problem - but
> what about attendant.resourceList.1.type?
> attendant.resourceList.1.type="automata"

Nice catch (I posted my latest try) but no cigar, sorry. And yes I did
try again.
You have sharp eyes.
 
> None of these attendant-parameters are listed in the 3.1.7
> sip admin guide though(In the 3.3.0-guide they are).
> I didn't look deeply into the docs - is this actually
> supposed to work in 3.1.7?

That is THE question, indeed. Well, supposedly, according to the wiki
that started this, regular 3.1.0 won't work but 3.10 with the
productivity license would.

I did not challenge the wiki earlier because I am both a freeswitch and
a polycom idiot, but I am starting to doubt it.

Oh, unrelated and FWIW, I have conducted another quick round of stupid
experiments this morning.
Note the "blah" line. I literally tried this.

<?xml version="1.0" encoding="utf-8" standalone="yes"?>
<userinfo>
 <reg
  reg.1.displayName="Test 2499"
  reg.1.blah="Blah"
  reg.1.address="2499"
  {<other commands snipped}
 />
</userinfo>

Well the "blah" line acts as expected: it does not do anything, but it
does not break anything either.

That part of the config file was working fine before, and it still is
after the introduction of the "blah" thing, so we can safely assume, as
expected, that the inclusion of a not-yet-implemented command (that
would be the case with a 3.2 command in a 3.1 config file) does not do
anything (heck that would be nice, btw: do what I expect future software
to do) but does not break anything either.


In a lousy effort to compensate for your time reading this,
I offer you the syntax of the new polycom software:
<?xml version="1.0" encoding="utf-8" standalone="yes"?>
<userinfo>
 <reg
  reg.1.MagicMindRead="read_the_user's_mind"
  reg.1.MagicAction="implement_thought"
/>
</userinfo>
:D

Michel.



From david.ponzone at ipeva.fr  Tue Apr 26 09:53:14 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 26 Apr 2011 07:53:14 +0200
Subject: [Freeswitch-users] SIP router/switch only configuratoin advice?
In-Reply-To: <0f0501cc0382$ac845fd0$058d1f70$@inbox.ee>
References: <0f0501cc0382$ac845fd0$058d1f70$@inbox.ee>
Message-ID: <FEA85ED4-6352-4D85-8C53-CC63879548F0@ipeva.fr>

Georgy,

the first thing you ask is trivial.
Basically, you may have one dialplan (one context) per client endpoint.
There is 2 ways to bind a context to a client:
-if the client registers, the context is configured in the user config (in directory/)
-if the client does not register, you fall back to IP-auth. So you can add an ACL to the SIP profile, for security, and then in the dialplan for this SIP profile, you should add rules testing the SRC IP (variable network_addr in FreeSWITCH XML dialplan) and transfering to the right context.

This way, you will have one specific context file per client, so you can have very specific routing rules per client.
Of course, you may use the same context for several clients.

I really recommend you dig a little bit for yourself in the wiki in order to setup this configuration.
This will provide you with the necessary understanding of FreeSWITCH to operate it.

For the second part of your question, you may need mod_lcr.
I never used, but a lot of people around use it.
I am not sure it does prepaid/credit-limit.
For that, you can use the "simple" nibblebill, but it needs some hacking to manage various rates per destination prefix.
Or use an external prepaid software.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 25/04/2011 ? 21:55, Georgy Goshin a ?crit :

> Hello!
> 
> I need some advice from experienced FreeSWITCH users. Please review my needs
> and advise me some configuration samples or adapted configuration  guides.
> I'd like to use the FS in two roles - a SIP router for my PBXes and as a
> wholesale switch. Both configurations are almost the same in my case. The
> desired configuration should be allow me to do the following:
> 
> route calls based on A num, B num, SRC IP (Mera MVTS is an example of what I
> want in simple calls routing).
> 
> If a more complex configuration that can do dynamic routing based on price,
> acd/asr and prepaid/credit-limit billing is possible too, I'd be glad to see
> some examples too.
> 
> 
> Thanks in advance,
> G. Goshin.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Tue Apr 26 10:40:15 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 26 Apr 2011 08:40:15 +0200
Subject: [Freeswitch-users] call not connecting sometime
In-Reply-To: <4DB47783.5090507@ppmt.org>
References: <4DB47783.5090507@ppmt.org>
Message-ID: <FA721E59-461F-4C90-BD3E-78BE60D7EDD9@ipeva.fr>

Phiippe,

I analyzed the trace a little bit and some things are odd.
Between lines 570 and 650, your provider (webcalldirect) sends you back twice a 183/SDP, but the IP in each SDP is different.
First:
o=ppmt 1303657356 1303657356 IN IP4 62.41.83.72
and then:
o=ppmt 1303657357 1303657357 IN IP4 208.167.230.118

I am not really sure if it's valid to REINVITE during early-media.
Anyway it seems, you don't receive early-media.
To check that, you would need to take a full trace (with tcpdump or wireshark/tshark) to see if you receive RTP traffic from them after they sent you back the 183/SDPs.
If they don't, it's quite normal you don't have ringback.
That's ugly.
But then, the worst part is that you never receive from them a 180 Ringing (meaning that the remote endpoint is ringing or that someone in the middle pretends it is) or a 200 OK (meaning the call was answered).
Again, not surprising the call fails.

We could spend hours trying to fix their issues, but I think the quick way is for you to get a test account from another reliable provider.
The one you use does not have a strong reputation. If I am not mistaken, it's one of those services with 30 different domain names and websites slightly different, all selling the same "service" with "free" calls.
I recommend you first validate your config with a good one, and then you'll go hunting for the cheap one (and remember that generally, with retail providers, cheap price = cheap quality).

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 24/04/2011 ? 21:18, Philippe Le Toquin a ?crit :

> Hello,
> 
> I had the problem a few weeks ago but after a reinstall (unrelated to that) the problem had gone so I put it
> done as my messing up my system at the time!
> 
> But today I had the same issue again. When I call sometime the call is not going through completely
> 
> The symptom on my side are that nothing happens (no ring tone) on the other they say that the phones
> rings but when they pick up the phone they can't hear anything.
> 
> Below is a siptrace where I changed the number and IP so I hope I didn't mess it up too much.
> 
> http://pastebin.freeswitch.org/16165
> 
> Can someone let me know if they see something wrong? I tried to understand but it is beyond me :(
> 
> Regards
> 
> /Philippe
> <0x1A0BDC2B.asc>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gosha at inbox.ee  Tue Apr 26 11:19:35 2011
From: gosha at inbox.ee (Georgy Goshin)
Date: Tue, 26 Apr 2011 10:19:35 +0300
Subject: [Freeswitch-users] SIP router/switch only configuratoin advice?
In-Reply-To: <FEA85ED4-6352-4D85-8C53-CC63879548F0@ipeva.fr>
References: <0f0501cc0382$ac845fd0$058d1f70$@inbox.ee>
	<FEA85ED4-6352-4D85-8C53-CC63879548F0@ipeva.fr>
Message-ID: <034201cc03e2$4c0b6850$e42238f0$@inbox.ee>

David, Thanks!

 

I see that routing/permission logic is little bit complicated for just call routing/proxing, I believe there will be simpler to do all job in the database. Is there a simple way to configure FS so it will ask from the database for a destination for a call and it?s parameters (ip parameters, media parameters, rewritten numbers A and B) based on src info (A num, SRC ip, SRC media pameters) and DST num?

 

 

Thanks,

Georgy

 

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Ponzone
Sent: Tuesday, April 26, 2011 8:53 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP router/switch only configuratoin advice?

 

Georgy,

 

the first thing you ask is trivial.

Basically, you may have one dialplan (one context) per client endpoint.

There is 2 ways to bind a context to a client:

-if the client registers, the context is configured in the user config (in directory/)

-if the client does not register, you fall back to IP-auth. So you can add an ACL to the SIP profile, for security, and then in the dialplan for this SIP profile, you should add rules testing the SRC IP (variable network_addr in FreeSWITCH XML dialplan) and transfering to the right context.

 

This way, you will have one specific context file per client, so you can have very specific routing rules per client.

Of course, you may use the same context for several clients.

 

I really recommend you dig a little bit for yourself in the wiki in order to setup this configuration.

This will provide you with the necessary understanding of FreeSWITCH to operate it.

 

For the second part of your question, you may need mod_lcr.

I never used, but a lot of people around use it.

I am not sure it does prepaid/credit-limit.

For that, you can use the "simple" nibblebill, but it needs some hacking to manage various rates per destination prefix.

Or use an external prepaid software.

 

David Ponzone  Direction Technique

email: david.ponzone at ipeva.fr

tel:      01 74 03 18 97

gsm:   06 66 98 76 34

 

Service Client IPeva

tel:      0811 46 26 26

www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>   -   www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/> 

 

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.

 





 

Le 25/04/2011 ? 21:55, Georgy Goshin a ?crit :





Hello!

I need some advice from experienced FreeSWITCH users. Please review my needs
and advise me some configuration samples or adapted configuration  guides.
I'd like to use the FS in two roles - a SIP router for my PBXes and as a
wholesale switch. Both configurations are almost the same in my case. The
desired configuration should be allow me to do the following:

route calls based on A num, B num, SRC IP (Mera MVTS is an example of what I
want in simple calls routing).

If a more complex configuration that can do dynamic routing based on price,
acd/asr and prepaid/credit-limit billing is possible too, I'd be glad to see
some examples too.


Thanks in advance,
G. Goshin.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Tue Apr 26 11:31:57 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 26 Apr 2011 09:31:57 +0200
Subject: [Freeswitch-users] SIP router/switch only configuratoin advice?
In-Reply-To: <034201cc03e2$4c0b6850$e42238f0$@inbox.ee>
References: <0f0501cc0382$ac845fd0$058d1f70$@inbox.ee>
	<FEA85ED4-6352-4D85-8C53-CC63879548F0@ipeva.fr>
	<034201cc03e2$4c0b6850$e42238f0$@inbox.ee>
Message-ID: <6E1355F9-14A6-44AB-A78B-0F3E4AD5F2B9@ipeva.fr>

Then check http://wiki.freeswitch.org/wiki/Xml_curl :)

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 26/04/2011 ? 09:19, Georgy Goshin a ?crit :

> David, Thanks!
>  
> I see that routing/permission logic is little bit complicated for just call routing/proxing, I believe there will be simpler to do all job in the database. Is there a simple way to configure FS so it will ask from the database for a destination for a call and it?s parameters (ip parameters, media parameters, rewritten numbers A and B) based on src info (A num, SRC ip, SRC media pameters) and DST num?
>  
>  
> Thanks,
> Georgy
>  
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of David Ponzone
> Sent: Tuesday, April 26, 2011 8:53 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] SIP router/switch only configuratoin advice?
>  
> Georgy,
>  
> the first thing you ask is trivial.
> Basically, you may have one dialplan (one context) per client endpoint.
> There is 2 ways to bind a context to a client:
> -if the client registers, the context is configured in the user config (in directory/)
> -if the client does not register, you fall back to IP-auth. So you can add an ACL to the SIP profile, for security, and then in the dialplan for this SIP profile, you should add rules testing the SRC IP (variable network_addr in FreeSWITCH XML dialplan) and transfering to the right context.
>  
> This way, you will have one specific context file per client, so you can have very specific routing rules per client.
> Of course, you may use the same context for several clients.
>  
> I really recommend you dig a little bit for yourself in the wiki in order to setup this configuration.
> This will provide you with the necessary understanding of FreeSWITCH to operate it.
>  
> For the second part of your question, you may need mod_lcr.
> I never used, but a lot of people around use it.
> I am not sure it does prepaid/credit-limit.
> For that, you can use the "simple" nibblebill, but it needs some hacking to manage various rates per destination prefix.
> Or use an external prepaid software.
>  
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>  
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>  
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>  
> 
> 
>  
> Le 25/04/2011 ? 21:55, Georgy Goshin a ?crit :
> 
> 
> Hello!
> 
> I need some advice from experienced FreeSWITCH users. Please review my needs
> and advise me some configuration samples or adapted configuration  guides.
> I'd like to use the FS in two roles - a SIP router for my PBXes and as a
> wholesale switch. Both configurations are almost the same in my case. The
> desired configuration should be allow me to do the following:
> 
> route calls based on A num, B num, SRC IP (Mera MVTS is an example of what I
> want in simple calls routing).
> 
> If a more complex configuration that can do dynamic routing based on price,
> acd/asr and prepaid/credit-limit billing is possible too, I'd be glad to see
> some examples too.
> 
> 
> Thanks in advance,
> G. Goshin.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>  
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 26 12:09:36 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 26 Apr 2011 09:09:36 +0100
Subject: [Freeswitch-users] presence in profiles
In-Reply-To: <84FA4FE9DD1F4E51BE172B2477CC96D6@e1705>
References: <84FA4FE9DD1F4E51BE172B2477CC96D6@e1705>
Message-ID: <BANLkTikcYOQwp2ojrDHD6xR1NwteW+P6GQ@mail.gmail.com>

<param name="manage-presence" value="false"/> on your sofia profile



On 25 April 2011 19:13, Madovsky <infos at madovsky.org> wrote:

>  Is it a problem to turn off presence in a FS cluster environment ?
>
> 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  Tue Apr 26 12:20:21 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 26 Apr 2011 09:20:21 +0100
Subject: [Freeswitch-users] absolute_codec_string question
In-Reply-To: <BANLkTik4mv7zWRndhM4_KxvvU-ybC-cLUQ@mail.gmail.com>
References: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>
	<4DB59C06.3090706@cupis.co.uk>
	<BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>
	<4DB5A53D.50201@cupis.co.uk>
	<BANLkTi=gh6AyV9C-fQaR6cY-x0=VX=XDeg@mail.gmail.com>
	<4DB5BB7B.4060409@cupis.co.uk>
	<BANLkTini7nLgyNstmf2GxzK9mcwOdG03Dw@mail.gmail.com>
	<BANLkTikDOK0YJoF8s0wnrz00Ah6MXF5-Gg@mail.gmail.com>
	<BANLkTik4mv7zWRndhM4_KxvvU-ybC-cLUQ@mail.gmail.com>
Message-ID: <BANLkTin32Ym=4TP4C_HOubGTCp0H_WOa_w@mail.gmail.com>

>
> {ignore_early_media,absolute_codec_string='G729,PCMU'}
>

The problem is that FS splits the variable names using commas. When there's
a comma in a value it gets confused. Using quotation marks doesn't make any
difference - they're treated as part of the value.

So FS is seeing this:
ignore_early_media
absolute_codec_string='G729
PCMU'

So the only codec that gets set is the G729.

There is a workaround for this. You want to use
{ignore_early_media,absolute_codec_string=^^:G729:PCMU}.

The ^^ at the start of a value tells FS 'treat the following character as a
comma'. Now FS sees:

ignore_early_media
absolute_codec_string=^^:G729:PCMU

And sets absolute_codec_string to G729,PCMU.

-Steve


On 25 April 2011 21:44, Nicolas Brenner <nico at clickfono.com> wrote:

> Oops, sorry you are right, I copied it wrong, doesn't fix it though.
>
>
> On Mon, Apr 25, 2011 at 3:35 PM, Vitalie Colosov <vetali100 at gmail.com>wrote:
>
>> Maybe I am wrong, but I see a typo in the following string, after "
>> ignore_early_media":
>>
>> originate {ignore_early_media,absolute_codec_string=G729}user/1001
>> &bridge(user/1000)
>>
>> It should be:
>>
>> originate {ignore_early_media=true,absolute_codec_string=G729}user/1001
>> &bridge(user/1000)
>>
>> Vitalie
>>
>>
>> 2011/4/25 Nicolas Brenner <nico at clickfono.com>
>>
>>> Thanks, yes, here you go:
>>>
>>> I'm making calls from the console to an X-Lite registered on extension
>>> 1001 which doesn't support G729. Calls 1 and 3 below fail because the codec
>>> options are not supported, and even though on the third call the
>>> absolute_codec_string variable should be 'G729,PCMU', it is not offering
>>> both codecs.
>>>
>>>
>>> On http://pastebin.freeswitch.org/16167 the SIP trace for:
>>>
>>> originate {ignore_early_media,absolute_codec_string=G729}user/1001
>>> &bridge(user/1000)
>>>
>>> This is the SDP:
>>>
>>>    v=0
>>>    o=FreeSWITCH 1303734153 1303734154 IN IP4 127.0.0.1
>>>    s=FreeSWITCH
>>>    c=IN IP4 127.0.0.1
>>>    t=0 0
>>>     m=audio 24036 RTP/AVP 18 101 13
>>>    a=rtpmap:101 telephone-event/8000
>>>    a=fmtp:101 0-16
>>>    a=ptime:20
>>>
>>>
>>> On http://pastebin.freeswitch.org/16169 the SIP trace for:
>>>
>>> originate {ignore_early_media,absolute_codec_string=PCMU}user/1001
>>> &bridge(user/1000)
>>>
>>> This is the SDP:
>>>
>>>    v=0
>>>    o=FreeSWITCH 1303733515 1303733516 IN IP4 127.0.0.1
>>>     s=FreeSWITCH
>>>    c=IN IP4 127.0.0.1
>>>    t=0 0
>>>    m=audio 24896 RTP/AVP 0 101 13
>>>    a=rtpmap:101 telephone-event/8000
>>>    a=fmtp:101 0-16
>>>    a=ptime:20
>>>
>>>
>>> On http://pastebin.freeswitch.org/16170 the SIP trace for:
>>>
>>> originate {ignore_early_media,absolute_codec_string='G729,PCMU'}user/1001
>>> &bridge(user/1000)
>>>
>>> This is the SDP:
>>>
>>>    v=0
>>>    o=FreeSWITCH 1303733520 1303733521 IN IP4 127.0.0.1
>>>    s=FreeSWITCH
>>>    c=IN IP4 127.0.0.1
>>>    t=0 0
>>>     m=audio 25138 RTP/AVP 18 101 13
>>>    a=rtpmap:101 telephone-event/8000
>>>    a=fmtp:101 0-16
>>>    a=ptime:20
>>>
>>>
>>>
>>>
>>> On Mon, Apr 25, 2011 at 2:20 PM, Paul Cupis <paul at cupis.co.uk> wrote:
>>>
>>>> On 25/04/11 18:08, Nicolas Brenner wrote:
>>>> > I'm trying to use the
>>>> > absolute_codec_string with originate from the console, like so:
>>>> >
>>>> > originate
>>>> >
>>>> {ignore_early_media,verbose_sdp=true,absolute_codec_string='G729,PMCU'}sofia/gateway/mygateway/444444
>>>> > &bridge(user/1001)
>>>> >
>>>> > Paul, I am using {absolute_codec_string='G729,PCMU'}, and I get the
>>>> same as
>>>> > if I don't quote the string, or if I just specify one codec:
>>>>
>>>> Can you provide (on pastebin.freeswitch.org) a complete log of a call,
>>>> please?
>>>>
>>>> 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
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 26 12:21:21 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 26 Apr 2011 09:21:21 +0100
Subject: [Freeswitch-users] absolute_codec_string question
In-Reply-To: <BANLkTin32Ym=4TP4C_HOubGTCp0H_WOa_w@mail.gmail.com>
References: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>
	<4DB59C06.3090706@cupis.co.uk>
	<BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>
	<4DB5A53D.50201@cupis.co.uk>
	<BANLkTi=gh6AyV9C-fQaR6cY-x0=VX=XDeg@mail.gmail.com>
	<4DB5BB7B.4060409@cupis.co.uk>
	<BANLkTini7nLgyNstmf2GxzK9mcwOdG03Dw@mail.gmail.com>
	<BANLkTikDOK0YJoF8s0wnrz00Ah6MXF5-Gg@mail.gmail.com>
	<BANLkTik4mv7zWRndhM4_KxvvU-ybC-cLUQ@mail.gmail.com>
	<BANLkTin32Ym=4TP4C_HOubGTCp0H_WOa_w@mail.gmail.com>
Message-ID: <BANLkTim7W_BhEMQKXuKk3WbVxvnrBnxmYw@mail.gmail.com>

Oh, and you'll need a version since 14 April 2011 for it to work in {}.
It'll work in [] since September 2010.

-Steve


On 26 April 2011 09:20, Steven Ayre <steveayre at gmail.com> wrote:

> {ignore_early_media,absolute_codec_string='G729,PCMU'}
>>
>
> The problem is that FS splits the variable names using commas. When there's
> a comma in a value it gets confused. Using quotation marks doesn't make any
> difference - they're treated as part of the value.
>
> So FS is seeing this:
>
> ignore_early_media
> absolute_codec_string='G729
> PCMU'
>
> So the only codec that gets set is the G729.
>
> There is a workaround for this. You want to use
> {ignore_early_media,absolute_codec_string=^^:G729:PCMU}.
>
> The ^^ at the start of a value tells FS 'treat the following character as a
> comma'. Now FS sees:
>
>
> ignore_early_media
> absolute_codec_string=^^:G729:PCMU
>
> And sets absolute_codec_string to G729,PCMU.
>
> -Steve
>
>
>
> On 25 April 2011 21:44, Nicolas Brenner <nico at clickfono.com> wrote:
>
>> Oops, sorry you are right, I copied it wrong, doesn't fix it though.
>>
>>
>> On Mon, Apr 25, 2011 at 3:35 PM, Vitalie Colosov <vetali100 at gmail.com>wrote:
>>
>>> Maybe I am wrong, but I see a typo in the following string, after "
>>> ignore_early_media":
>>>
>>> originate {ignore_early_media,absolute_codec_string=G729}user/1001
>>> &bridge(user/1000)
>>>
>>> It should be:
>>>
>>> originate {ignore_early_media=true,absolute_codec_string=G729}user/1001
>>> &bridge(user/1000)
>>>
>>> Vitalie
>>>
>>>
>>> 2011/4/25 Nicolas Brenner <nico at clickfono.com>
>>>
>>>> Thanks, yes, here you go:
>>>>
>>>> I'm making calls from the console to an X-Lite registered on extension
>>>> 1001 which doesn't support G729. Calls 1 and 3 below fail because the codec
>>>> options are not supported, and even though on the third call the
>>>> absolute_codec_string variable should be 'G729,PCMU', it is not offering
>>>> both codecs.
>>>>
>>>>
>>>> On http://pastebin.freeswitch.org/16167 the SIP trace for:
>>>>
>>>> originate {ignore_early_media,absolute_codec_string=G729}user/1001
>>>> &bridge(user/1000)
>>>>
>>>> This is the SDP:
>>>>
>>>>    v=0
>>>>    o=FreeSWITCH 1303734153 1303734154 IN IP4 127.0.0.1
>>>>    s=FreeSWITCH
>>>>    c=IN IP4 127.0.0.1
>>>>    t=0 0
>>>>     m=audio 24036 RTP/AVP 18 101 13
>>>>    a=rtpmap:101 telephone-event/8000
>>>>    a=fmtp:101 0-16
>>>>    a=ptime:20
>>>>
>>>>
>>>> On http://pastebin.freeswitch.org/16169 the SIP trace for:
>>>>
>>>> originate {ignore_early_media,absolute_codec_string=PCMU}user/1001
>>>> &bridge(user/1000)
>>>>
>>>> This is the SDP:
>>>>
>>>>    v=0
>>>>    o=FreeSWITCH 1303733515 1303733516 IN IP4 127.0.0.1
>>>>     s=FreeSWITCH
>>>>    c=IN IP4 127.0.0.1
>>>>    t=0 0
>>>>    m=audio 24896 RTP/AVP 0 101 13
>>>>    a=rtpmap:101 telephone-event/8000
>>>>    a=fmtp:101 0-16
>>>>    a=ptime:20
>>>>
>>>>
>>>> On http://pastebin.freeswitch.org/16170 the SIP trace for:
>>>>
>>>> originate
>>>> {ignore_early_media,absolute_codec_string='G729,PCMU'}user/1001
>>>> &bridge(user/1000)
>>>>
>>>> This is the SDP:
>>>>
>>>>    v=0
>>>>    o=FreeSWITCH 1303733520 1303733521 IN IP4 127.0.0.1
>>>>    s=FreeSWITCH
>>>>    c=IN IP4 127.0.0.1
>>>>    t=0 0
>>>>     m=audio 25138 RTP/AVP 18 101 13
>>>>    a=rtpmap:101 telephone-event/8000
>>>>    a=fmtp:101 0-16
>>>>    a=ptime:20
>>>>
>>>>
>>>>
>>>>
>>>> On Mon, Apr 25, 2011 at 2:20 PM, Paul Cupis <paul at cupis.co.uk> wrote:
>>>>
>>>>> On 25/04/11 18:08, Nicolas Brenner wrote:
>>>>> > I'm trying to use the
>>>>> > absolute_codec_string with originate from the console, like so:
>>>>> >
>>>>> > originate
>>>>> >
>>>>> {ignore_early_media,verbose_sdp=true,absolute_codec_string='G729,PMCU'}sofia/gateway/mygateway/444444
>>>>> > &bridge(user/1001)
>>>>> >
>>>>> > Paul, I am using {absolute_codec_string='G729,PCMU'}, and I get the
>>>>> same as
>>>>> > if I don't quote the string, or if I just specify one codec:
>>>>>
>>>>> Can you provide (on pastebin.freeswitch.org) a complete log of a call,
>>>>> please?
>>>>>
>>>>> 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
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 26 12:23:08 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 26 Apr 2011 09:23:08 +0100
Subject: [Freeswitch-users] variable direction
In-Reply-To: <2961A5A6-C60E-498B-899E-19B5BC24535F@ipeva.fr>
References: <4DB0E180.8020201@tagnet.ru>
	<BANLkTik3qeuVzmV5Ab5uYrTzZHjKLbvtxg@mail.gmail.com>
	<4DB11B7B.2090401@tagnet.ru>
	<2961A5A6-C60E-498B-899E-19B5BC24535F@ipeva.fr>
Message-ID: <BANLkTi=poA99jQPgFNTM7+EerwG4wa_yXQ@mail.gmail.com>

David,

He isn't talking about the Direction channel variable, he is talking about
the direction attribute available on gateway variables. :)

-Steve


On 22 April 2011 09:09, David Ponzone <david.ponzone at ipeva.fr> wrote:

> Boris,
>
> you don't set direction, it is set automatically.
> If you call from the user you was talking about, the A-leg (from the user
> to FS) will be inbound, and the B-leg (from FS to the callee party) will be
> outbound.
> If you call the user, it's the other way around, of course.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 22/04/2011 ? 08:08, Boris Kovalenko a ?crit :
>
>  Hello!
>
>     Yes, but this form of variable is available only to gateways? I've
> tried to set direction for user variable too without success. Variable is
> always present. My test enviroment:
>
> <include>
>   <user id="73435230024">
>     <params>
>       <param name="password" value="012345"/>
>       <param name="vm-enabled" value="false"/>
>       <param name="vm-password" value="326699"/>
>     </params>
>     <variables>
>       <variable name="toll_allow"
> value="internal,local,domestic,international"/>
>       <variable name="v_test1" value="test1" direction="inbound"/>
>       <variable name="v_test2" value="test2" direction="outbound"/>
>       <!-- <variable name="v_tagnet_context" value="ho"/>-->
>     </variables>
>   </user>
> </include>
>
> I do a call from this user to external gateway and see both variables are
> set with application=info.
>
> The direction variable is set at the time of the call based upon whether or
> not the user is the caller (outbound) or callee (inbound).
>
>  -MC
>
> On Thu, Apr 21, 2011 at 7:01 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
>
>> Hello!
>>
>>     I found that with gateways I may use <variable ....
>> direction="..."/>. Is this possible with regular (directory) users too?
>> If not, may somebody explain why?
>>
>> --
>> ? ?????????,
>>   ????? ?????????
>>   ??? "??????"
>>   ???. +7 (3435) 230001
>>   ???? +7 (3435) 230005
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> 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
>
>
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435) 230001
>   ???? +7 (3435) 230005
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 26 12:25:33 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 26 Apr 2011 09:25:33 +0100
Subject: [Freeswitch-users] variable direction
In-Reply-To: <4DB0E180.8020201@tagnet.ru>
References: <4DB0E180.8020201@tagnet.ru>
Message-ID: <BANLkTinhXSmht6VbJtD3r91VLGLwjwkhUA@mail.gmail.com>

Checking the source code, sorry no this attribute only appears to be
available on gateways.

-Steve



On 22 April 2011 03:01, Boris Kovalenko <boris at tagnet.ru> wrote:

> Hello!
>
>     I found that with gateways I may use <variable ....
> direction="..."/>. Is this possible with regular (directory) users too?
> If not, may somebody explain why?
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435) 230001
>   ???? +7 (3435) 230005
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 26 12:26:38 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 26 Apr 2011 09:26:38 +0100
Subject: [Freeswitch-users] variable direction
In-Reply-To: <4DB13A0C.3080008@tagnet.ru>
References: <4DB0E180.8020201@tagnet.ru>
	<BANLkTik3qeuVzmV5Ab5uYrTzZHjKLbvtxg@mail.gmail.com>
	<4DB11B7B.2090401@tagnet.ru>
	<2961A5A6-C60E-498B-899E-19B5BC24535F@ipeva.fr>
	<4DB13A0C.3080008@tagnet.ru>
Message-ID: <BANLkTin2r8D=mN-sUKzk0YBcBoEmS_t=pA@mail.gmail.com>

For users that attribute is unknown and therefore will be ignored, so
they'll both be set.

-Steve


On 22 April 2011 09:19, Boris Kovalenko <boris at tagnet.ru> wrote:

>  Hello!
>
>     Yes, so, with my configuration, when I do a call only the variable
> test1 should be set? Isn't?
>
>
> Boris,
>
>  you don't set direction, it is set automatically.
> If you call from the user you was talking about, the A-leg (from the user
> to FS) will be inbound, and the B-leg (from FS to the callee party) will be
> outbound.
> If you call the user, it's the other way around, of course.
>
>   David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
>  Service Client IPeva
>  tel:      0811 46 26 26
>  www.ipeva.fr  -   www.ipeva-studio.com
>
>  *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
>  Le 22/04/2011 ? 08:08, Boris Kovalenko a ?crit :
>
>  Hello!
>
>     Yes, but this form of variable is available only to gateways? I've
> tried to set direction for user variable too without success. Variable is
> always present. My test enviroment:
>
> <include>
>   <user id="73435230024">
>     <params>
>       <param name="password" value="012345"/>
>       <param name="vm-enabled" value="false"/>
>       <param name="vm-password" value="326699"/>
>     </params>
>     <variables>
>       <variable name="toll_allow"
> value="internal,local,domestic,international"/>
>       <variable name="v_test1" value="test1" direction="inbound"/>
>       <variable name="v_test2" value="test2" direction="outbound"/>
>       <!-- <variable name="v_tagnet_context" value="ho"/>-->
>     </variables>
>   </user>
> </include>
>
> I do a call from this user to external gateway and see both variables are
> set with application=info.
>
> The direction variable is set at the time of the call based upon whether or
> not the user is the caller (outbound) or callee (inbound).
>
>  -MC
>
> On Thu, Apr 21, 2011 at 7:01 PM, Boris Kovalenko <boris at tagnet.ru> wrote:
>
>> Hello!
>>
>>     I found that with gateways I may use <variable ....
>> direction="..."/>. Is this possible with regular (directory) users too?
>> If not, may somebody explain why?
>>
>> --
>> ? ?????????,
>>   ????? ?????????
>>   ??? "??????"
>>   ???. +7 (3435) 230001
>>   ???? +7 (3435) 230005
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> 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
>
>
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435) 230001
>   ???? +7 (3435) 230005
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> 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
>
>
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435) 230001
>   ???? +7 (3435) 230005
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Tue Apr 26 12:27:36 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 26 Apr 2011 10:27:36 +0200
Subject: [Freeswitch-users] continue_on_fail for condition only where
	100 not received?
In-Reply-To: <BANLkTin2g_5=RJusFR2bvJMYSQumnQOztg@mail.gmail.com>
References: <1303324320582-6291782.post@n2.nabble.com>
	<245144A7-1F53-400D-A23A-C83DEA96C400@freeswitch.org>
	<BANLkTi=dDmN4s0t3SaLt2ig_cDFXo9sgkQ@mail.gmail.com>
	<BANLkTi=nuCxO2CHM0nOzpgMG8MS5i0P_tw@mail.gmail.com>
	<BANLkTin2g_5=RJusFR2bvJMYSQumnQOztg@mail.gmail.com>
Message-ID: <C0525DDD-2238-47F9-A665-420F56CB0A15@ipeva.fr>

I think the original question was: is there a way to failover if the remote gateway does not send back anything (no 1XX, no 2XX), with a configurable INVITE timeout (5 seconds would probably be better than 60).

To answer to Mike:
I think we should consider the possibility that the remote gateway answers correctly to SIP OPTIONS, but only fails to respond to INVITEs. I saw weirder things :)

David Ponzone  Direction Technique,
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 21/04/2011 ? 20:06, Kristian Kielhofner a ?crit :

> Of course it can be skipped but nothing, and I mean nothing, can fire
> faster than a 100.  To my knowledge Sofia fires a 100 before the
> dialplan even begins hunting, as an example (very good behavior).  In
> cases where the far end has a low T1 timer you should fire a 100
> before you even take the time to execute dialplan, etc.  Especially in
> cases where you may encounter high call volume and varying call
> setup/hunting times.
> 
> SER config 101 is to fire a 100 before you do any DB logic, etc as it
> buys you time beyond the first T1 timer (500ms or less) before the far
> end (that you have no control over) retries.  It's best practice and
> HIGHLY, HIGHLY recommended.
> 
> This is such a strange case I'm not sure what the best recommendation
> might be as the original question doesn't contain enough specific
> detail.
> 
> On Thu, Apr 21, 2011 at 11:55 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> Yes, but it can be skipped - for instance if you're autoanswering you could
>> skip the 100 and jump straight to the 200. Or similarly if you're generating
>> ringback jump straight to the 183. There's no reason that 100 is needed
>> first.
>> 
>> -Steve
>> 
> 
> -- 
> 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 david.ponzone at ipeva.fr  Tue Apr 26 12:35:51 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 26 Apr 2011 10:35:51 +0200
Subject: [Freeswitch-users] FS -route to next GW if the first GW full
In-Reply-To: <BANLkTimSG_M+q8G05W7V2SwdUgV0w938xQ@mail.gmail.com>
References: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>
	<4dae8ae9.823d2b0a.02b3.155f@mx.google.com>
	<BANLkTino_bR7=BvhvMdhOWmbMVXcwFQJ5Q@mail.gmail.com>
	<BANLkTi=Pi9_gkv-WbkyijJH6-Kj1VgEuvg@mail.gmail.com>
	<BANLkTin0fO=4N81-1iVsLotcR9F2-Ftzww@mail.gmail.com>
	<BANLkTimSG_M+q8G05W7V2SwdUgV0w938xQ@mail.gmail.com>
Message-ID: <1BCD5E86-6436-4F8D-88BB-05A9B0AE00B4@ipeva.fr>

Charles,

if you may, I would recommend you change the behaviour of this GW sending back a 404 when it's full...
That's odd.
It should rather use: 503 Service unavailable.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 21/04/2011 ? 04:45, fieldpeak a ?crit :

> Thanks All, it is resolved now with below config.
> 
> 2011/4/20, Kristian Kielhofner <kris at kriskinc.com>:
>> Try this:
>> 
>> <extension name="9_GW" >
>>  <condition field="destination_number" expression="^(9\d+)$">
>>    <action application="set" data="continue_on_fail=true"/>
>>    <action application="set" data="failure_causes=UNALLOCATED_NUMBER"/>
>>    <action application="set" data="hangup_after_bridge=true"/>
>>    <action application="bridge" data="sofia/internal/$1 at 192.168.200.201"/>
>>    <action application="bridge" data="sofia/internal/$1 at 192.168.200.202"/>
>> </condition>
>> </extension>
>> 
>> On Wed, Apr 20, 2011 at 10:58 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>> Hi Steve,
>>> 
>>> Thanks for your so rich stuff provided.
>>> 
>>> however, i tried to use error code to route as below, it failed (did
>>> not route the next GW when recevied UNALLOCATED_NUMBER), can you
>>> please point out is there any place wrong in the dailplan? attached is
>>> the log, thanks.
>>> 
>>> <extension name="9_GW" >
>>> 
>>>  <condition field="destination_number" expression="^(9\d+)$">
>>>     <action application="set" data="continue_on_fail=true"/>
>>>    <action application="set" data="hangup_after_bridge=false"/>
>>>     <action application="bridge"
>>> data="sofia/internal/$1 at 192.168.200.201"/>
>>> </condition>
>>> 
>>>  <condition field="brige_hangup_cause" expression="UNALLOCATED_NUMBER">
>>>   <action application="bridge" data="sofia/internal/$1 at 192.168.200.202"/>
>>>  </condition>
>>> 
>>> </extension>
>>> 
>>> Regards,
>>> Charles
>>> 
>>> 2011/4/20, Steven Ayre <steveayre at gmail.com>:
>>>> If you know the number of calls the GW can handle in advance, you can
>>>> limit
>>>> the call attempts before sending the call to the gateway:
>>>> http://wiki.freeswitch.org/wiki/Limit
>>>> There are several ways to use that.
>>>> 
>>>> You can reroute calls to a 2nd gateway on error:
>>>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover
>>>> 
>>>> There are channel variables that let you control what to consider a
>>>> reroutable error and what is a give-up error:
>>>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>>> http://wiki.freeswitch.org/wiki/Channel_Variables#failure_causes
>>>> This might also be useful, particularly with mod_limit:
>>>> http://wiki.freeswitch.org/wiki/Channel_Variables#transfer_on_fail
>>>> 
>>>> You could use mod_lcr to get a list of all the GWs, but in a random
>>>> order.
>>>> That'd let you load balance (randomly) but reroute when required without
>>>> duplicates unlike mod_distributor.
>>>> 
>>>> Hopefully that's enough building blocks to give you somewhere to start...
>>>> 
>>>> -Steve
>>>> 
>>>> 
>>>> 
>>>> On 20 April 2011 08:27, Charles <fieldpeak at gmail.com> wrote:
>>>> 
>>>>> 
>>>>> i have two media GWs connected with FS for PSTN calls, FS route the
>>>>> first
>>>>> one in normal case, once the first GW get full (all of channels were
>>>>> occupied), i need FS route to the second GW.
>>>>> i found once the first GW get full, it will reply '404 not found' to FS,
>>>>> can FS route to the second one according to this condition or any other
>>>>> alternative?
>>>>> 
>>>>> although i know there is mod_distributor for load balancing, however, i
>>>>> need if only first one full then route to second one, so it looks this
>>>>> module not suitable for this senario...
>>>>> 
>>>>> 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
>>> 
>>> 
>> 
>> 
>> 
>> --
>> 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
>> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 26 13:16:11 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Tue, 26 Apr 2011 15:16:11 +0600
Subject: [Freeswitch-users] variable direction
In-Reply-To: <BANLkTin2r8D=mN-sUKzk0YBcBoEmS_t=pA@mail.gmail.com>
References: <4DB0E180.8020201@tagnet.ru>	<BANLkTik3qeuVzmV5Ab5uYrTzZHjKLbvtxg@mail.gmail.com>	<4DB11B7B.2090401@tagnet.ru>	<2961A5A6-C60E-498B-899E-19B5BC24535F@ipeva.fr>	<4DB13A0C.3080008@tagnet.ru>
	<BANLkTin2r8D=mN-sUKzk0YBcBoEmS_t=pA@mail.gmail.com>
Message-ID: <4DB68D5B.3050102@tagnet.ru>

Hello!

     Thank You, Steve!
> For users that attribute is unknown and therefore will be ignored, so 
> they'll both be set.
>
> -Steve
>
>
> On 22 April 2011 09:19, Boris Kovalenko <boris at tagnet.ru 
> <mailto:boris at tagnet.ru>> wrote:
>
>     Hello!
>
>         Yes, so, with my configuration, when I do a call only the
>     variable test1 should be set? Isn't?
>
>
>>     Boris,
>>
>>     you don't set direction, it is set automatically.
>>     If you call from the user you was talking about, the A-leg (from
>>     the user to FS) will be inbound, and the B-leg (from FS to the
>>     callee party) will be outbound.
>>     If you call the user, it's the other way around, of course.
>>
>>     David Ponzone Direction Technique
>>     email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>     tel:      01 74 03 18 97
>>     gsm:   06 66 98 76 34
>>
>>     Service ClientIPeva
>>     tel:      0811 46 26 26
>>     www.ipeva.fr  - www.ipeva-studio.com
>>
>>     /Ce message et toutes les pi?ces jointes sont confidentiels et
>>     ?tablis ? l'intention exclusive de ses destinataires. Toute
>>     utilisation ou diffusion non autoris?e est interdite. Tout
>>     message ?lectronique est susceptible d'alt?ration.
>>     /*/IPeva/*/ d?cline toute responsabilit? au titre de ce message
>>     s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>>     destinataire de ce message, merci de le d?truire imm?diatement et
>>     d'avertir l'exp?diteur./
>>     /
>>     /
>>
>>
>>
>>     Le 22/04/2011 ? 08:08, Boris Kovalenko a ?crit :
>>
>>>     Hello!
>>>
>>>         Yes, but this form of variable is available only to
>>>     gateways? I've tried to set direction for user variable too
>>>     without success. Variable is always present. My test enviroment:
>>>
>>>     <include>
>>>     <user id="73435230024">
>>>     <params>
>>>     <param name="password" value="012345"/>
>>>     <param name="vm-enabled" value="false"/>
>>>     <param name="vm-password" value="326699"/>
>>>     </params>
>>>     <variables>
>>>     <variable name="toll_allow"
>>>     value="internal,local,domestic,international"/>
>>>     <variable name="v_test1" value="test1" direction="inbound"/>
>>>     <variable name="v_test2" value="test2" direction="outbound"/>
>>>     <!-- <variable name="v_tagnet_context" value="ho"/>-->
>>>     </variables>
>>>     </user>
>>>     </include>
>>>
>>>     I do a call from this user to external gateway and see both
>>>     variables are set with application=info.
>>>
>>>>     The direction variable is set at the time of the call based
>>>>     upon whether or not the user is the caller (outbound) or callee
>>>>     (inbound).
>>>>
>>>>     -MC
>>>>
>>>>     On Thu, Apr 21, 2011 at 7:01 PM, Boris Kovalenko
>>>>     <boris at tagnet.ru <mailto:boris at tagnet.ru>> wrote:
>>>>
>>>>         Hello!
>>>>
>>>>             I found that with gateways I may use <variable ....
>>>>         direction="..."/>. Is this possible with regular
>>>>         (directory) users too?
>>>>         If not, may somebody explain why?
>>>>
>>>>         --
>>>>         ? ?????????,
>>>>           ????? ?????????
>>>>           ??? "??????"
>>>>           ???. +7 (3435) 230001
>>>>           ???? +7 (3435) 230005
>>>>
>>>>
>>>>         _______________________________________________
>>>>         FreeSWITCH-users mailing list
>>>>         FreeSWITCH-users at lists.freeswitch.org
>>>>         <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>         http://www.freeswitch.org <http://www.freeswitch.org/>
>>>>
>>>>
>>>>
>>>>     _______________________________________________
>>>>     FreeSWITCH-users mailing list
>>>>     FreeSWITCH-users at lists.freeswitch.org  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>     http://www.freeswitch.org  <http://www.freeswitch.org/>
>>>
>>>
>>>     -- 
>>>     ? ?????????,
>>>        ????? ?????????
>>>        ??? "??????"
>>>        ???. +7 (3435) 230001
>>>        ???? +7 (3435) 230005
>>>     _______________________________________________
>>>     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
>
>
>     -- 
>     ? ?????????,
>        ????? ?????????
>        ??? "??????"
>        ???. +7 (3435) 230001
>        ???? +7 (3435) 230005
>
>
>     _______________________________________________
>     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


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005

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From steveayre at gmail.com  Tue Apr 26 13:17:17 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 26 Apr 2011 10:17:17 +0100
Subject: [Freeswitch-users] FS -route to next GW if the first GW full
In-Reply-To: <1BCD5E86-6436-4F8D-88BB-05A9B0AE00B4@ipeva.fr>
References: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>
	<4dae8ae9.823d2b0a.02b3.155f@mx.google.com>
	<BANLkTino_bR7=BvhvMdhOWmbMVXcwFQJ5Q@mail.gmail.com>
	<BANLkTi=Pi9_gkv-WbkyijJH6-Kj1VgEuvg@mail.gmail.com>
	<BANLkTin0fO=4N81-1iVsLotcR9F2-Ftzww@mail.gmail.com>
	<BANLkTimSG_M+q8G05W7V2SwdUgV0w938xQ@mail.gmail.com>
	<1BCD5E86-6436-4F8D-88BB-05A9B0AE00B4@ipeva.fr>
Message-ID: <BANLkTik52YThrDPU252n-nT=X=6YvHsSBw@mail.gmail.com>

Indeed... there's a specific ISDN clearing cause for it - 34 "no
circuit/channel available" (NORMAL_CIRCUIT_CONGESTION). That maps to SIP 503
Service Unavailable (RFC3398).

Unallocated number implies it's not allocated anywhere and therefore
shouldn't reroute, while 503 suggests you should try another route.

-Steve



On 26 April 2011 09:35, David Ponzone <david.ponzone at ipeva.fr> wrote:

> Charles,
>
> if you may, I would recommend you change the behaviour of this GW sending
> back a 404 when it's full...
> That's odd.
> It should rather use: 503 Service unavailable.
>
>  David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr  -   www.ipeva-studio.com
>
> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
> destinataire de ce message, merci de le d?truire imm?diatement et d'avertir
> l'exp?diteur.*
> *
> *
>
>
>
> Le 21/04/2011 ? 04:45, fieldpeak a ?crit :
>
> Thanks All, it is resolved now with below config.
>
> 2011/4/20, Kristian Kielhofner <kris at kriskinc.com>:
>
> Try this:
>
>
> <extension name="9_GW" >
>
>  <condition field="destination_number" expression="^(9\d+)$">
>
>    <action application="set" data="continue_on_fail=true"/>
>
>    <action application="set" data="failure_causes=UNALLOCATED_NUMBER"/>
>
>    <action application="set" data="hangup_after_bridge=true"/>
>
>    <action application="bridge" data="sofia/internal/$1 at 192.168.200.201"/>
>
>    <action application="bridge" data="sofia/internal/$1 at 192.168.200.202"/>
>
> </condition>
>
> </extension>
>
>
> On Wed, Apr 20, 2011 at 10:58 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>
> Hi Steve,
>
>
> Thanks for your so rich stuff provided.
>
>
> however, i tried to use error code to route as below, it failed (did
>
> not route the next GW when recevied UNALLOCATED_NUMBER), can you
>
> please point out is there any place wrong in the dailplan? attached is
>
> the log, thanks.
>
>
> <extension name="9_GW" >
>
>
>  <condition field="destination_number" expression="^(9\d+)$">
>
>     <action application="set" data="continue_on_fail=true"/>
>
>    <action application="set" data="hangup_after_bridge=false"/>
>
>     <action application="bridge"
>
> data="sofia/internal/$1 at 192.168.200.201"/>
>
>  </condition>
>
>
>  <condition field="brige_hangup_cause" expression="UNALLOCATED_NUMBER">
>
>   <action application="bridge" data="sofia/internal/$1 at 192.168.200.202"/>
>
>  </condition>
>
>
>  </extension>
>
>
> Regards,
>
> Charles
>
>
> 2011/4/20, Steven Ayre <steveayre at gmail.com>:
>
> If you know the number of calls the GW can handle in advance, you can
>
> limit
>
> the call attempts before sending the call to the gateway:
>
> http://wiki.freeswitch.org/wiki/Limit
>
> There are several ways to use that.
>
>
> You can reroute calls to a 2nd gateway on error:
>
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover
>
>
> There are channel variables that let you control what to consider a
>
> reroutable error and what is a give-up error:
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#failure_causes
>
> This might also be useful, particularly with mod_limit:
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#transfer_on_fail
>
>
> You could use mod_lcr to get a list of all the GWs, but in a random
>
> order.
>
> That'd let you load balance (randomly) but reroute when required without
>
> duplicates unlike mod_distributor.
>
>
> Hopefully that's enough building blocks to give you somewhere to start...
>
>
> -Steve
>
>
>
>
> On 20 April 2011 08:27, Charles <fieldpeak at gmail.com> wrote:
>
>
>
> i have two media GWs connected with FS for PSTN calls, FS route the
>
> first
>
> one in normal case, once the first GW get full (all of channels were
>
> occupied), i need FS route to the second GW.
>
> i found once the first GW get full, it will reply '404 not found' to FS,
>
> can FS route to the second one according to this condition or any other
>
> alternative?
>
>
> although i know there is mod_distributor for load balancing, however, i
>
> need if only first one full then route to second one, so it looks this
>
> module not suitable for this senario...
>
>
> 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
>
>
>
>
>
>
> --
>
> 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
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Tue Apr 26 13:28:07 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Tue, 26 Apr 2011 17:28:07 +0800
Subject: [Freeswitch-users] FS -route to next GW if the first GW full
In-Reply-To: <BANLkTik52YThrDPU252n-nT=X=6YvHsSBw@mail.gmail.com>
References: <94FE8C418F344DA5A07CBE1D9913DAEB@e1705>
	<4dae8ae9.823d2b0a.02b3.155f@mx.google.com>
	<BANLkTino_bR7=BvhvMdhOWmbMVXcwFQJ5Q@mail.gmail.com>
	<BANLkTi=Pi9_gkv-WbkyijJH6-Kj1VgEuvg@mail.gmail.com>
	<BANLkTin0fO=4N81-1iVsLotcR9F2-Ftzww@mail.gmail.com>
	<BANLkTimSG_M+q8G05W7V2SwdUgV0w938xQ@mail.gmail.com>
	<1BCD5E86-6436-4F8D-88BB-05A9B0AE00B4@ipeva.fr>
	<BANLkTik52YThrDPU252n-nT=X=6YvHsSBw@mail.gmail.com>
Message-ID: <BANLkTinpoUX-FiUTxgt0ByN20fX5AybDCQ@mail.gmail.com>

yes...agree with you , 503 better than 404(maybe also reply 404 in some case
that GW is not full...), thanks.

Regards,
Charles

? 2011-4-26 ??5:19?"Steven Ayre" <steveayre at gmail.com>???
> Indeed... there's a specific ISDN clearing cause for it - 34 "no
> circuit/channel available" (NORMAL_CIRCUIT_CONGESTION). That maps to SIP
503
> Service Unavailable (RFC3398).
>
> Unallocated number implies it's not allocated anywhere and therefore
> shouldn't reroute, while 503 suggests you should try another route.
>
> -Steve
>
>
>
> On 26 April 2011 09:35, David Ponzone <david.ponzone at ipeva.fr> wrote:
>
>> Charles,
>>
>> if you may, I would recommend you change the behaviour of this GW sending
>> back a 404 when it's full...
>> That's odd.
>> It should rather use: 503 Service unavailable.
>>
>> David Ponzone Direction Technique
>> email: david.ponzone at ipeva.fr
>> tel: 01 74 03 18 97
>> gsm: 06 66 98 76 34
>>
>> Service Client IPeva
>> tel: 0811 46 26 26
>> www.ipeva.fr - www.ipeva-studio.com
>>
>> *Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
>> l'intention exclusive de ses destinataires. Toute utilisation ou
diffusion
>> non autoris?e est interdite. Tout message ?lectronique est susceptible
>> d'alt?ration. **IPeva** d?cline toute responsabilit? au titre de ce
>> message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas
>> destinataire de ce message, merci de le d?truire imm?diatement et
d'avertir
>> l'exp?diteur.*
>> *
>> *
>>
>>
>>
>> Le 21/04/2011 ? 04:45, fieldpeak a ?crit :
>>
>> Thanks All, it is resolved now with below config.
>>
>> 2011/4/20, Kristian Kielhofner <kris at kriskinc.com>:
>>
>> Try this:
>>
>>
>> <extension name="9_GW" >
>>
>> <condition field="destination_number" expression="^(9\d+)$">
>>
>> <action application="set" data="continue_on_fail=true"/>
>>
>> <action application="set" data="failure_causes=UNALLOCATED_NUMBER"/>
>>
>> <action application="set" data="hangup_after_bridge=true"/>
>>
>> <action application="bridge" data="sofia/internal/$1 at 192.168.200.201"/>
>>
>> <action application="bridge" data="sofia/internal/$1 at 192.168.200.202"/>
>>
>> </condition>
>>
>> </extension>
>>
>>
>> On Wed, Apr 20, 2011 at 10:58 AM, fieldpeak <fieldpeak at gmail.com> wrote:
>>
>> Hi Steve,
>>
>>
>> Thanks for your so rich stuff provided.
>>
>>
>> however, i tried to use error code to route as below, it failed (did
>>
>> not route the next GW when recevied UNALLOCATED_NUMBER), can you
>>
>> please point out is there any place wrong in the dailplan? attached is
>>
>> the log, thanks.
>>
>>
>> <extension name="9_GW" >
>>
>>
>> <condition field="destination_number" expression="^(9\d+)$">
>>
>> <action application="set" data="continue_on_fail=true"/>
>>
>> <action application="set" data="hangup_after_bridge=false"/>
>>
>> <action application="bridge"
>>
>> data="sofia/internal/$1 at 192.168.200.201"/>
>>
>> </condition>
>>
>>
>> <condition field="brige_hangup_cause" expression="UNALLOCATED_NUMBER">
>>
>> <action application="bridge" data="sofia/internal/$1 at 192.168.200.202"/>
>>
>> </condition>
>>
>>
>> </extension>
>>
>>
>> Regards,
>>
>> Charles
>>
>>
>> 2011/4/20, Steven Ayre <steveayre at gmail.com>:
>>
>> If you know the number of calls the GW can handle in advance, you can
>>
>> limit
>>
>> the call attempts before sending the call to the gateway:
>>
>> http://wiki.freeswitch.org/wiki/Limit
>>
>> There are several ways to use that.
>>
>>
>> You can reroute calls to a 2nd gateway on error:
>>
>>
>>
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_bridge#Implementing_Failover
>>
>>
>> There are channel variables that let you control what to consider a
>>
>> reroutable error and what is a give-up error:
>>
>> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
>>
>> http://wiki.freeswitch.org/wiki/Channel_Variables#failure_causes
>>
>> This might also be useful, particularly with mod_limit:
>>
>> http://wiki.freeswitch.org/wiki/Channel_Variables#transfer_on_fail
>>
>>
>> You could use mod_lcr to get a list of all the GWs, but in a random
>>
>> order.
>>
>> That'd let you load balance (randomly) but reroute when required without
>>
>> duplicates unlike mod_distributor.
>>
>>
>> Hopefully that's enough building blocks to give you somewhere to start...
>>
>>
>> -Steve
>>
>>
>>
>>
>> On 20 April 2011 08:27, Charles <fieldpeak at gmail.com> wrote:
>>
>>
>>
>> i have two media GWs connected with FS for PSTN calls, FS route the
>>
>> first
>>
>> one in normal case, once the first GW get full (all of channels were
>>
>> occupied), i need FS route to the second GW.
>>
>> i found once the first GW get full, it will reply '404 not found' to FS,
>>
>> can FS route to the second one according to this condition or any other
>>
>> alternative?
>>
>>
>> although i know there is mod_distributor for load balancing, however, i
>>
>> need if only first one full then route to second one, so it looks this
>>
>> module not suitable for this senario...
>>
>>
>> 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
>>
>>
>>
>>
>>
>>
>> --
>>
>> 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
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Tue Apr 26 13:48:05 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 26 Apr 2011 11:48:05 +0200
Subject: [Freeswitch-users] how to pass arbitrary headers from A leg to
	B leg when bridging
In-Reply-To: <047B4BA6-39DE-4BB4-BFED-7353A6EBBB8B@dchorton.com>
References: <047B4BA6-39DE-4BB4-BFED-7353A6EBBB8B@dchorton.com>
Message-ID: <6483B4EF-899F-4D60-9553-9815396CB398@ipeva.fr>

Wouldn't it be simpler to first deal with auth, and to put your transcoding FS after that in your network path ?

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 12/04/2011 ? 18:49, Dave Horton a ?crit :

> So I'm guessing this isn't possible without hacking the source code, which I've already done to solve my problem for now.
> 
> But I'd like to make sure there isn't a better way of doing things, and thus I'd like to revise and restate my question for clarity.  First, though, let me describe what I am doing, because I think it's a not-uncommon scenario that I think would be something that others may want to do.  I basically want to use FS as a simple transcoding server between two endpoints, call them A and B.  Calls coming in from A will be using speex codec and I want to send them out to B using PCMU; calls coming in from B will be PCMU and I want to send them to A using speex.  The FS server will be a B2BUA and will be doing transcoding only -- no authentication (and no registration).  Simple, right?  The only fly in the ointment is that A is authenticating calls with B by providing a Proxy-Authorization header.  So I need to take the Proxy-Authorization header received on the A leg and include it on the B leg.
> 
> So far, the only way I have found to do that is to hack the code to create a new channel variable.  I've done this, and it works.  However, this leads me to the following questions
> 
> 1) Is there a better way to do this?  If there is no way to do it as a dialplan out of the box, can it be done as a script?
> 
> 2) sofia has parsed all of the sip headers on the incoming invite for us, and they're all available from mod_sofia.  Shouldn't those all be available to us (i.e., application developers) by some means (i.e., channel variables)?
> 
> Dave
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From philippe at ppmt.org  Tue Apr 26 14:47:02 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Tue, 26 Apr 2011 06:47:02 -0400
Subject: [Freeswitch-users] call not connecting sometime
In-Reply-To: <BANLkTimkGJ2sqVcyAu7HMr9JS2=qToQwFQ@mail.gmail.com>
References: <4DB47783.5090507@ppmt.org>	<2A5992B4-AABE-4041-A74E-4C7A956C4BD3@ipeva.fr>	<4DB620BC.7020202@ppmt.org>
	<BANLkTimkGJ2sqVcyAu7HMr9JS2=qToQwFQ@mail.gmail.com>
Message-ID: <4DB6A2A6.5060402@ppmt.org>

I don't know :(

All I can say is that calls are working again. Only time will tell if
they start failing again

By the time I deactivated DTMF it was too late for me to try the failing
call (I live in Canada and
the number I want to call are in Europe). They love me but no that much 
that I can call them
 at 3 in the morning :D



On 11-04-26 01:29 AM, Vitalie Colosov wrote:
> Hi Philippe,
>
> So, you have deactivated the DTMF action, and ... did it help? :)
>
>
>
> 2011/4/26 Philippe Le Toquin <philippe at ppmt.org
> <mailto:philippe at ppmt.org>>
>
>     by provider you mean if I use an external company to forward the call?
>
>     Yes there is.
>
>     I am not sure if I can give name of company so I won't but it is one
>     of those company where you buy some credit and they give you 60
>     days free before starting using the credit
>
>     Of course I can't rule out that it is on their side as most of the
>     time it works and other it won't but
>     when I check my call logs on their website it shows that the call
>     was process by them as successful
>
>     The only issue is that the call  doesn't connect fully (the other
>     side say they answer but can't hear anything
>     while on my side it is still ringing)
>
>     I guess that the trace is not showing anything? is there more
>     trace I could activate for the next time it occurs?
>
>     For the moment I have deactivated the DTMF action as recommended
>     by Vitalie as I don't think we will
>     need it for these type of calls ( I am sure my wife will tell if
>     that is not true :) )
>
>     regards
>
>     /Philippe
>
>
>     On 11-04-25 10:15 AM, David Ponzone wrote:
>>     Phil,
>>
>>     is there a provider involved ?
>>     Perhaps it's on their side ?
>>
>>     David Ponzone  Direction Technique
>>     email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>>     tel:      01 74 03 18 97
>>     gsm:   06 66 98 76 34
>>
>>     Service Client IPeva
>>     tel:      0811 46 26 26
>>     www.ipeva.fr  -   www.ipeva-studio.com
>>
>>     /Ce message et toutes les pi?ces jointes sont confidentiels et
>>     ?tablis ? l'intention exclusive de ses destinataires. Toute
>>     utilisation ou diffusion non autoris?e est interdite. Tout
>>     message ?lectronique est susceptible
>>     d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au titre
>>     de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
>>     n'?tes pas destinataire de ce message, merci de le d?truire
>>     imm?diatement et d'avertir l'exp?diteur./
>>     /
>>     /
>>
>>
>>
>>     Le 24/04/2011 ? 21:18, Philippe Le Toquin a ?crit :
>>
>>>     Hello,
>>>
>>>     I had the problem a few weeks ago but after a reinstall
>>>     (unrelated to that) the problem had gone so I put it
>>>     done as my messing up my system at the time!
>>>
>>>     But today I had the same issue again. When I call sometime the
>>>     call is not going through completely
>>>
>>>     The symptom on my side are that nothing happens (no ring tone)
>>>     on the other they say that the phones
>>>     rings but when they pick up the phone they can't hear anything.
>>>
>>>     Below is a siptrace where I changed the number and IP so I hope
>>>     I didn't mess it up too much.
>>>
>>>     http://pastebin.freeswitch.org/16165
>>>
>>>     Can someone let me know if they see something wrong? I tried to
>>>     understand but it is beyond me :(
>>>
>>>     Regards
>>>
>>>     /Philippe
>>>     <0x1A0BDC2B.asc>_______________________________________________
>>>     FreeSWITCH-users mailing list
>>>     FreeSWITCH-users at lists.freeswitch.org
>>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>     http://www.freeswitch.org
>>
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From philippe at ppmt.org  Tue Apr 26 14:53:31 2011
From: philippe at ppmt.org (Philippe Le Toquin)
Date: Tue, 26 Apr 2011 06:53:31 -0400
Subject: [Freeswitch-users] call not connecting sometime
In-Reply-To: <FA721E59-461F-4C90-BD3E-78BE60D7EDD9@ipeva.fr>
References: <4DB47783.5090507@ppmt.org>
	<FA721E59-461F-4C90-BD3E-78BE60D7EDD9@ipeva.fr>
Message-ID: <4DB6A42B.9030100@ppmt.org>

Thanks for looking into it David!

I totally agree with you cheap rarely mean quality but sometime it has
to do :(

Next time it happens again I will take a wireshark trace

As far as fixing their issue I doubt it will help. They tend to royally
ignore you in case of problem

I only stick with them because in most cases (ie 98%) it works

I just wanted to make sure that the issue was not on FS side. If it is
not then I am not going to pursue it!

I guess I will be on the hunt for another cheap provider (if anyone
knows one ! )

Thanks a gain

/Philippe

On 11-04-26 02:40 AM, David Ponzone wrote:
> Phiippe,
>
> I analyzed the trace a little bit and some things are odd.
> Between lines 570 and 650, your provider (webcalldirect) sends you
> back twice a 183/SDP, but the IP in each SDP is different.
> First:
> o=ppmt 1303657356 1303657356 IN IP4 62.41.83.72
> and then:
> o=ppmt 1303657357 1303657357 IN IP4 208.167.230.118
>
> I am not really sure if it's valid to REINVITE during early-media.
> Anyway it seems, you don't receive early-media.
> To check that, you would need to take a full trace (with tcpdump or
> wireshark/tshark) to see if you receive RTP traffic from them after
> they sent you back the 183/SDPs.
> If they don't, it's quite normal you don't have ringback.
> That's ugly.
> But then, the worst part is that you never receive from them a 180
> Ringing (meaning that the remote endpoint is ringing or that someone
> in the middle pretends it is) or a 200 OK (meaning the call was answered).
> Again, not surprising the call fails.
>
> We could spend hours trying to fix their issues, but I think the quick
> way is for you to get a test account from another reliable provider.
> The one you use does not have a strong reputation. If I am not
> mistaken, it's one of those services with 30 different domain names
> and websites slightly different, all selling the same "service" with
> "free" calls.
> I recommend you first validate your config with a good one, and then
> you'll go hunting for the cheap one (and remember that generally, with
> retail providers, cheap price = cheap quality).
>
> David Ponzone  Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel:      01 74 03 18 97
> gsm:   06 66 98 76 34
>
> Service Client IPeva
> tel:      0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>  -   www.ipeva-studio.com
> <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis
> ? l'intention exclusive de ses destinataires. Toute utilisation ou
> diffusion non autoris?e est interdite. Tout message ?lectronique est
> susceptible d'alt?ration. /*/IPeva/*/ d?cline toute responsabilit? au
> titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous
> n'?tes pas destinataire de ce message, merci de le d?truire
> imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
> Le 24/04/2011 ? 21:18, Philippe Le Toquin a ?crit :
>
>> Hello,
>>
>> I had the problem a few weeks ago but after a reinstall (unrelated to
>> that) the problem had gone so I put it
>> done as my messing up my system at the time!
>>
>> But today I had the same issue again. When I call sometime the call
>> is not going through completely
>>
>> The symptom on my side are that nothing happens (no ring tone) on the
>> other they say that the phones
>> rings but when they pick up the phone they can't hear anything.
>>
>> Below is a siptrace where I changed the number and IP so I hope I
>> didn't mess it up too much.
>>
>> http://pastebin.freeswitch.org/16165
>>
>> Can someone let me know if they see something wrong? I tried to
>> understand but it is beyond me :(
>>
>> Regards
>>
>> /Philippe
>> <0x1A0BDC2B.asc>_______________________________________________
>> 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 rogelio.perez at gmail.com  Tue Apr 26 15:58:04 2011
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Tue, 26 Apr 2011 08:58:04 -0300
Subject: [Freeswitch-users] mod_nibblebill updates
Message-ID: <F2AD9D2B-681F-4B5D-991D-8B3379DB97CD@gmail.com>

Hello, I'm using mod_nibblebill to bill my calls only for leg b and heartbeats set to 'off'.
After the hangup I see 3 different MySQL updates from nibblebill instead of just one.

This is my dial plan:

	<extension name="TEST">
        	<condition field="destination_number" expression="^(1\d+)$">
                	<action application="set" data="dialed_extension=$1"/>
                        <action application="export" data="dialed_extension=$1"/>
                        <action application="set" data="hangup_after_bridge=true"/>
                        <action application="set" data="continue_on_fail=true"/>
			<action application="bridge" data="{nibble_rate=5,nibble_account=13}sofia/gateway/xxxxxxx/$1"/>
                          <action application="hangup"/>
		</condition>
	</extension>

...this is the MySQL log:

		68080 Query	select 1
		68080 Query	UPDATE web_account SET balance=balance-0.343372 WHERE id='13'
		68080 Query	select 1
		68080 Query	SELECT balance AS nibble_balance FROM web_account WHERE id='13'
		68080 Query	select 1
		68080 Query	UPDATE web_account SET balance=balance-0.001177 WHERE id='13'
		68080 Query	select 1
		68080 Query	SELECT balance AS nibble_balance FROM web_account WHERE id='13'
		68080 Query	select 1
		68080 Query	UPDATE web_account SET balance=balance-0.001197 WHERE id='13'
		68080 Query	select 1
		68080 Query	SELECT balance AS nibble_balance FROM web_account WHERE id='13'


...and here's the FS log: http://pastebin.freeswitch.org/16175 

The xml_cdr is saved after the second update, so the nibble_total_billed is different than the real total value debited from the account balance.

Is this normal? 
Shouldn't nibblebill just do one update after the hangup?

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From Nabble at slickdeals.endjunk.com  Tue Apr 26 16:06:01 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Tue, 26 Apr 2011 05:06:01 -0700 (PDT)
Subject: [Freeswitch-users] mod_spandsp fails to cross compile
In-Reply-To: <4DB608A9.8060105@coppice.org>
References: <1303752081792-6303345.post@n2.nabble.com>
	<4DB608A9.8060105@coppice.org>
Message-ID: <1303819561747-6305469.post@n2.nabble.com>

Thanks Steve and you nailed it to the point.

In my OpenWRT configuration, I also included  http://yate.null.ro YATE 
package that depends on an external SpanDSP package. As such, this breaks
the FS mod_spandsp compilation. I wish there is a way to solve this problem.
For now, I just disable to build with  http://yate.null.ro YATE  package. 

-----
FreeSWITCH hosted on a Seagate DockStar with OpenWRT.
--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-spandsp-fails-to-cross-compile-tp6303345p6305469.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From rodrigo.ferrari at cellmidia.com.br  Tue Apr 26 17:03:47 2011
From: rodrigo.ferrari at cellmidia.com.br (Rodrigo Ferrari)
Date: Tue, 26 Apr 2011 10:03:47 -0300
Subject: [Freeswitch-users] Receiving SMS some alert?
In-Reply-To: <BANLkTikKhcgh1VvugMzrzS3Hesv4BmPuGw@mail.gmail.com>
References: <BANLkTimHvjOCLNUXUOSO9K=74=mno2ogTw@mail.gmail.com>
	<BANLkTim2HO4OhDMSo2yRPWxOfFyQWyCK6Q@mail.gmail.com>
	<BANLkTimmcXeyaHacWZZEEUs+kFLbRSjUJw@mail.gmail.com>
	<BANLkTikKhcgh1VvugMzrzS3Hesv4BmPuGw@mail.gmail.com>
Message-ID: <BANLkTin87g5m3=395kHn3vYNtydf_90zmQ@mail.gmail.com>

Great one,

I will try this log to disk, may this help.

Thanks.

Best regard's.

Rodrigo Ferrari.

Em 25 de abril de 2011 17:55, Geovani Ricardo Wiedenhoft <
grw.freeswitch at gmail.com> escreveu:

> Hello.
>
>  Enable logs to disk.
>
> <param name="log-to-disk" value="debugging"/>
>
> So, restart the Freeswitch or in this case o mod_khomp.
>
>
> The logs always help us.
>
>
> Probably the context was not set.
>
> You need to configure sms incoming context in khomp.conf.xml
>
> <param name="context-gsm-sms" value="khomp-sms-DD-CC"/>
>
> So, restart the Freeswitch.
>
>
> Regards,
> Geovani Ricardo Wiedenhoft
>
>
>
> 2011/4/25 Rodrigo Ferrari <rodrigo.ferrari at cellmidia.com.br>:
> > Obrigado Jo?o!
> >
> > Eu tentei o suporte mas n?o obtive a resposta do que perguntei, tentei
> > novamente e estou aguardando devido ao feriado mesmo, por?m n?o consigo
> > ficar parado, fico tentando buscar solu??o por todos os lugares, meu
> > problema ? na quest?o de receber o SMS, pego meu celular e envio uma
> > mensagem para um chip da placa, com o console do freeswitch aberto, n?o
> > recebo nenhum evento de entrada e nem o dialplan ? executado, se ent?o eu
> > com meu celular ligo, no momento o freeswitch apita no console a entrada
> de
> > uma chamada e executa o dialplan, ent?o acredito estar esquecendo de
> > configurar algo para lidar com as mensagens que entram na placa.
> >
> > Obrigado!
> >
> > Abra?os.
> >
> > Em 25 de abril de 2011 11:07, Jo?o Mesquita <jmesquita at freeswitch.org>
> > escreveu:
> >>
> >> Rodrigo,
> >> Have you tried their official support? They do provide tech support and
> >> they officially support FreeSWITCH. I am sure if you did, they did not
> reply
> >> because of the holiday.
> >> suporte (at) khomp.com.br should be just fine.
> >> Regards,
> >> Jo?o Mesquita
> >>
> >>
> >>
> >> On Mon, Apr 25, 2011 at 9:45 AM, Rodrigo Ferrari
> >> <rodrigo.ferrari at cellmidia.com.br> wrote:
> >>>
> >>> Hello friends,
> >>>
> >>> I just bought a Khomp board to send and receive SMS, the problem is,
> I'm
> >>> not receiving the messages, it's not marking at the console a message
> >>> incoming. Is this some configuration that I miss?
> >>>
> >>> Thanks, best regards.
> >>> Rodrigo Ferrari.
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 nico at clickfono.com  Tue Apr 26 17:26:14 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Tue, 26 Apr 2011 09:26:14 -0400
Subject: [Freeswitch-users] absolute_codec_string question
In-Reply-To: <BANLkTim7W_BhEMQKXuKk3WbVxvnrBnxmYw@mail.gmail.com>
References: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>
	<4DB59C06.3090706@cupis.co.uk>
	<BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>
	<4DB5A53D.50201@cupis.co.uk>
	<BANLkTi=gh6AyV9C-fQaR6cY-x0=VX=XDeg@mail.gmail.com>
	<4DB5BB7B.4060409@cupis.co.uk>
	<BANLkTini7nLgyNstmf2GxzK9mcwOdG03Dw@mail.gmail.com>
	<BANLkTikDOK0YJoF8s0wnrz00Ah6MXF5-Gg@mail.gmail.com>
	<BANLkTik4mv7zWRndhM4_KxvvU-ybC-cLUQ@mail.gmail.com>
	<BANLkTin32Ym=4TP4C_HOubGTCp0H_WOa_w@mail.gmail.com>
	<BANLkTim7W_BhEMQKXuKk3WbVxvnrBnxmYw@mail.gmail.com>
Message-ID: <BANLkTimo-kqZyN0aMqyXTkiGaa-smAo=zA@mail.gmail.com>

Awesome! that worked perfectly. Thanks Steve.


On Tue, Apr 26, 2011 at 4:21 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Oh, and you'll need a version since 14 April 2011 for it to work in {}.
> It'll work in [] since September 2010.
>
> -Steve
>
>
>
> On 26 April 2011 09:20, Steven Ayre <steveayre at gmail.com> wrote:
>
>> {ignore_early_media,absolute_codec_string='G729,PCMU'}
>>>
>>
>> The problem is that FS splits the variable names using commas. When
>> there's a comma in a value it gets confused. Using quotation marks doesn't
>> make any difference - they're treated as part of the value.
>>
>> So FS is seeing this:
>>
>> ignore_early_media
>> absolute_codec_string='G729
>> PCMU'
>>
>> So the only codec that gets set is the G729.
>>
>> There is a workaround for this. You want to use
>> {ignore_early_media,absolute_codec_string=^^:G729:PCMU}.
>>
>> The ^^ at the start of a value tells FS 'treat the following character as
>> a comma'. Now FS sees:
>>
>>
>> ignore_early_media
>> absolute_codec_string=^^:G729:PCMU
>>
>> And sets absolute_codec_string to G729,PCMU.
>>
>> -Steve
>>
>>
>>
>> On 25 April 2011 21:44, Nicolas Brenner <nico at clickfono.com> wrote:
>>
>>> Oops, sorry you are right, I copied it wrong, doesn't fix it though.
>>>
>>>
>>> On Mon, Apr 25, 2011 at 3:35 PM, Vitalie Colosov <vetali100 at gmail.com>wrote:
>>>
>>>> Maybe I am wrong, but I see a typo in the following string, after "
>>>> ignore_early_media":
>>>>
>>>> originate {ignore_early_media,absolute_codec_string=G729}user/1001
>>>> &bridge(user/1000)
>>>>
>>>> It should be:
>>>>
>>>> originate {ignore_early_media=true,absolute_codec_string=G729}user/1001
>>>> &bridge(user/1000)
>>>>
>>>> Vitalie
>>>>
>>>>
>>>> 2011/4/25 Nicolas Brenner <nico at clickfono.com>
>>>>
>>>>> Thanks, yes, here you go:
>>>>>
>>>>> I'm making calls from the console to an X-Lite registered on extension
>>>>> 1001 which doesn't support G729. Calls 1 and 3 below fail because the codec
>>>>> options are not supported, and even though on the third call the
>>>>> absolute_codec_string variable should be 'G729,PCMU', it is not offering
>>>>> both codecs.
>>>>>
>>>>>
>>>>> On http://pastebin.freeswitch.org/16167 the SIP trace for:
>>>>>
>>>>> originate {ignore_early_media,absolute_codec_string=G729}user/1001
>>>>> &bridge(user/1000)
>>>>>
>>>>> This is the SDP:
>>>>>
>>>>>    v=0
>>>>>    o=FreeSWITCH 1303734153 1303734154 IN IP4 127.0.0.1
>>>>>    s=FreeSWITCH
>>>>>    c=IN IP4 127.0.0.1
>>>>>    t=0 0
>>>>>     m=audio 24036 RTP/AVP 18 101 13
>>>>>    a=rtpmap:101 telephone-event/8000
>>>>>    a=fmtp:101 0-16
>>>>>    a=ptime:20
>>>>>
>>>>>
>>>>> On http://pastebin.freeswitch.org/16169 the SIP trace for:
>>>>>
>>>>> originate {ignore_early_media,absolute_codec_string=PCMU}user/1001
>>>>> &bridge(user/1000)
>>>>>
>>>>> This is the SDP:
>>>>>
>>>>>    v=0
>>>>>    o=FreeSWITCH 1303733515 1303733516 IN IP4 127.0.0.1
>>>>>     s=FreeSWITCH
>>>>>    c=IN IP4 127.0.0.1
>>>>>    t=0 0
>>>>>    m=audio 24896 RTP/AVP 0 101 13
>>>>>    a=rtpmap:101 telephone-event/8000
>>>>>    a=fmtp:101 0-16
>>>>>    a=ptime:20
>>>>>
>>>>>
>>>>> On http://pastebin.freeswitch.org/16170 the SIP trace for:
>>>>>
>>>>> originate
>>>>> {ignore_early_media,absolute_codec_string='G729,PCMU'}user/1001
>>>>> &bridge(user/1000)
>>>>>
>>>>> This is the SDP:
>>>>>
>>>>>    v=0
>>>>>    o=FreeSWITCH 1303733520 1303733521 IN IP4 127.0.0.1
>>>>>    s=FreeSWITCH
>>>>>    c=IN IP4 127.0.0.1
>>>>>    t=0 0
>>>>>     m=audio 25138 RTP/AVP 18 101 13
>>>>>    a=rtpmap:101 telephone-event/8000
>>>>>    a=fmtp:101 0-16
>>>>>    a=ptime:20
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Mon, Apr 25, 2011 at 2:20 PM, Paul Cupis <paul at cupis.co.uk> wrote:
>>>>>
>>>>>> On 25/04/11 18:08, Nicolas Brenner wrote:
>>>>>> > I'm trying to use the
>>>>>> > absolute_codec_string with originate from the console, like so:
>>>>>> >
>>>>>> > originate
>>>>>> >
>>>>>> {ignore_early_media,verbose_sdp=true,absolute_codec_string='G729,PMCU'}sofia/gateway/mygateway/444444
>>>>>> > &bridge(user/1001)
>>>>>> >
>>>>>> > Paul, I am using {absolute_codec_string='G729,PCMU'}, and I get the
>>>>>> same as
>>>>>> > if I don't quote the string, or if I just specify one codec:
>>>>>>
>>>>>> Can you provide (on pastebin.freeswitch.org) a complete log of a
>>>>>> call,
>>>>>> please?
>>>>>>
>>>>>> 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
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 benkokakao at gmail.com  Tue Apr 26 17:56:34 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Tue, 26 Apr 2011 15:56:34 +0200
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
 with productivity license.
In-Reply-To: <471D76419F9EF642962323D13DF1DF69011E59@newserver.arneill-py.local>
References: <471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local>
	<BANLkTi=XmNDV51=xoNsViHUYaUZeyMpz_A@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E58@newserver.arneill-py.local>
	<BANLkTi=o1nUa70X_PHU51M_TTGTfnSSdKg@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E59@newserver.arneill-py.local>
Message-ID: <BANLkTi=9Gu1vOD-KVdCSs3MbNxng+Tqv2w@mail.gmail.com>

> Well the "blah" line acts as expected: it does not do anything, but it
> does not break anything either.

Haha, so you are not too frustrated, that's good :-)

I've taken a look at
http://downloads.polycom.com/voice/voip/uc/UC_Software_Release_Notes_3_3_1F.pdf
and on page 44 you will find that the BLF-features including the
enhanced attendant-features where introduced with firmware version 3.2
- sorry to say, but you're most probably out of luck.

I've updated the wiki accordingly.

Best regards
Christian


From psilvao at gmail.com  Tue Apr 26 18:06:04 2011
From: psilvao at gmail.com (Pablo Silva)
Date: Tue, 26 Apr 2011 11:06:04 -0300
Subject: [Freeswitch-users] No Dial Tone ... Zaptel + FreeSwitch +
	TDM400P
In-Reply-To: <BANLkTinYAVo136Y2YgYftRikGC5sDdUrGQ@mail.gmail.com>
References: <BANLkTimyKOrBq8TCDA3imnYS79b74=KgXA@mail.gmail.com>
	<4DB57C37.4020203@gmail.com>
	<BANLkTi=p1xTTgJyAsJgO1YGY=1C3hp=hOw@mail.gmail.com>
	<BANLkTinYAVo136Y2YgYftRikGC5sDdUrGQ@mail.gmail.com>
Message-ID: <BANLkTi=WVdxr88D18Db737R3d4KfQ5zcEA@mail.gmail.com>

Opss forget pastebin ...

http://pastebin.com/87WRcsE2

On Tue, Apr 26, 2011 at 11:05 AM, Pablo Silva <psilvao at gmail.com> wrote:
> well.., now i ?get freetdm installed, the last version, but dosen't
> works.., i can't get dial tone and don't know why.. :-(, only i can
> see the message on my sip phone
>
> Service Unavailable
>
>
>
> That's my logs and configuration files, i'm using TDM400P with one FXO module ..
>
> Thanks in advance for any hint
> -Pablo
>
> Pd: I'm Chilean for these reason you can see in tone zone cl (Chile).
>
>
> On Mon, Apr 25, 2011 at 10:50 AM, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:
>> openzap have bean renamed to freetdm
>> so please try mod_freetdm
>> thank you.
>> On 25/04/2011 14:56, Pablo Silva wrote:
>>>
>>> Hi!
>>>
>>>
>>> ? ?I've tryed for severals hours to make a phone call from freeswitch
>>> server, using TDM400P.
>>>
>>> ? ?I don't know why i can't hear the dial tone, and it's impossible
>>> make a phone call.., i read freeswitch wiki page
>>> about TDM400P configuration, as you can see that's my configuration an log
>>>
>>> http://pastebin.com/Y94UqN5R
>>>
>>> ? if I focus on information from the log, you can see that loads
>>> openzap, the server recognizes the TDM400P card, the call is routed to
>>> the PSTN but no concrete achievements.
>>>
>>> When you call, I hear the tone, not the dialing, I guess I'm near the
>>> solution to my problem, I request your cooperation.
>>>
>>> In advance, thanks.
>>> -Pablo
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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 yehavi.bourvine at gmail.com  Tue Apr 26 18:42:35 2011
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 26 Apr 2011 17:42:35 +0300
Subject: [Freeswitch-users] Audiocodes phones
Message-ID: <BANLkTinQnMC6mxRZQfCZAEGHDx9rNAZgxw@mail.gmail.com>

Audiocodes has announced recently two IP phones. (single line and 4 lines).
I've updated the wiki at
http://wiki.freeswitch.org/wiki/Interop_List#AudioCodes with more pointers
from there.

The phones are targetted toward Asterisk (as this is the PBX they use) but
they are doing some changes to suit FreeSwitch as well. I am in touch with
the product managers there, asking them for more features...

                           Regards, __Yehavi:
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From psilvao at gmail.com  Tue Apr 26 18:54:06 2011
From: psilvao at gmail.com (Pablo Silva)
Date: Tue, 26 Apr 2011 11:54:06 -0300
Subject: [Freeswitch-users] was --> No Dial Tone ... Zaptel + FreeSwitch
	+ TDM400P
Message-ID: <BANLkTi=+0HCfZMVS4sy_DSHTxNGj6Fr-EQ@mail.gmail.com>

Dear Friends:


   This article is out? http://wiki.freeswitch.org/wiki/OpenZap_Dahdi,
or is exactly to do for my case..?

Greetings,
Pablo


On Tue, Apr 26, 2011 at 11:06 AM, Pablo Silva <psilvao at gmail.com> wrote:
> Opss forget pastebin ...
>
> http://pastebin.com/87WRcsE2
>
> On Tue, Apr 26, 2011 at 11:05 AM, Pablo Silva <psilvao at gmail.com> wrote:
>> well.., now i ?get freetdm installed, the last version, but dosen't
>> works.., i can't get dial tone and don't know why.. :-(, only i can
>> see the message on my sip phone
>>
>> Service Unavailable
>>
>>
>>
>> That's my logs and configuration files, i'm using TDM400P with one FXO module ..
>>
>> Thanks in advance for any hint
>> -Pablo
>>
>> Pd: I'm Chilean for these reason you can see in tone zone cl (Chile).
>>
>>
>> On Mon, Apr 25, 2011 at 10:50 AM, Meftah Tayeb <tayeb.meftah at gmail.com> wrote:
>>> openzap have bean renamed to freetdm
>>> so please try mod_freetdm
>>> thank you.
>>> On 25/04/2011 14:56, Pablo Silva wrote:
>>>>
>>>> Hi!
>>>>
>>>>
>>>> ? ?I've tryed for severals hours to make a phone call from freeswitch
>>>> server, using TDM400P.
>>>>
>>>> ? ?I don't know why i can't hear the dial tone, and it's impossible
>>>> make a phone call.., i read freeswitch wiki page
>>>> about TDM400P configuration, as you can see that's my configuration an log
>>>>
>>>> http://pastebin.com/Y94UqN5R
>>>>
>>>> ? if I focus on information from the log, you can see that loads
>>>> openzap, the server recognizes the TDM400P card, the call is routed to
>>>> the PSTN but no concrete achievements.
>>>>
>>>> When you call, I hear the tone, not the dialing, I guess I'm near the
>>>> solution to my problem, I request your cooperation.
>>>>
>>>> In advance, thanks.
>>>> -Pablo
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at 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 michel at arneill-py.sacramento.ca.us  Tue Apr 26 18:59:11 2011
From: michel at arneill-py.sacramento.ca.us (Michel Py)
Date: Tue, 26 Apr 2011 07:59:11 -0700
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
	with productivity license.
In-Reply-To: <BANLkTi=9Gu1vOD-KVdCSs3MbNxng+Tqv2w@mail.gmail.com>
References: <471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local><BANLkTi=XmNDV51=xoNsViHUYaUZeyMpz_A@mail.gmail.com><471D76419F9EF642962323D13DF1DF69011E58@newserver.arneill-py.local><BANLkTi=o1nUa70X_PHU51M_TTGTfnSSdKg@mail.gmail.com><471D76419F9EF642962323D13DF1DF69011E59@newserver.arneill-py.local>
	<BANLkTi=9Gu1vOD-KVdCSs3MbNxng+Tqv2w@mail.gmail.com>
Message-ID: <471D76419F9EF642962323D13DF1DF69011E5A@newserver.arneill-py.local>

> Christian Benke wrote:
> you will find that the BLF-features including the enhanced attendant
> features where introduced with firmware version 3.2

I found some other references that mentioned something with 3.1, though.

http://support.polycom.com/global/documents/support/technical/products/v
oice/SoundPoint_IP_Enhanced_BLF_QT37381.pdf

http://www.polycomnz.co.nz/software/spip_ssip_3_1_0RevB_relnotes.pdf

http://blog.voipsupply.com/polycom-sip-firmware-31-released-whats-in-it-
for-me

Are you the author of the original wiki page?


From freeswitch at peely.com  Tue Apr 26 19:50:35 2011
From: freeswitch at peely.com (peely)
Date: Tue, 26 Apr 2011 08:50:35 -0700 (PDT)
Subject: [Freeswitch-users] continue_on_fail for condition only where
	100 not received?
In-Reply-To: <BANLkTinXAijFk8GxH4u3AL8DNS7Dd-pTVg@mail.gmail.com>
References: <1303324320582-6291782.post@n2.nabble.com>
	<245144A7-1F53-400D-A23A-C83DEA96C400@freeswitch.org>
	<BANLkTi=dDmN4s0t3SaLt2ig_cDFXo9sgkQ@mail.gmail.com>
	<BANLkTi=nuCxO2CHM0nOzpgMG8MS5i0P_tw@mail.gmail.com>
	<BANLkTinXAijFk8GxH4u3AL8DNS7Dd-pTVg@mail.gmail.com>
Message-ID: <1303833035795-6306349.post@n2.nabble.com>

Hi,

I can guarantee a 100 as the equipment is within my network!

In this condition I'm only concerned about a dead box scenario, in fact the
boxes are configured as separate Gateways so I guess I could rely on the
gateway being available through OPTIONS pings and being taken out of service
automatically.

So either a CONTINUE_ON_FAIL which acts on no 100 response within x seconds
or will only attempt a second gateway if the first gateway is disabled would
do.

Any ideas?

On the other topics, it's quite common for User Agents to offer an "always
send 100 trying" option for this specific reason, although you need to know
it is enabled I guess.


Thanks,



Neil.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/continue-on-fail-for-condition-only-where-100-not-received-tp6291782p6306349.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From bwibowo at gmail.com  Tue Apr 26 19:55:20 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Tue, 26 Apr 2011 22:55:20 +0700
Subject: [Freeswitch-users] fs prepaid wholesale
Message-ID: <BANLkTimw3J_M6tPPQ1q-XPmF7q8=64XdPA@mail.gmail.com>

hi
any idea to implement prepaid wholesale for FS?


thx

budi
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From juanito1982 at gmail.com  Tue Apr 26 20:57:18 2011
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Tue, 26 Apr 2011 18:57:18 +0200
Subject: [Freeswitch-users] fs prepaid wholesale
In-Reply-To: <BANLkTimw3J_M6tPPQ1q-XPmF7q8=64XdPA@mail.gmail.com>
References: <BANLkTimw3J_M6tPPQ1q-XPmF7q8=64XdPA@mail.gmail.com>
Message-ID: <BANLkTiknes7esHBSZTMmhiwWEQQ3D-fMmw@mail.gmail.com>

mod_lcr? mod_nible? a LUA script? a custom C module?

I think you have several options. Look at the wiki for more info.

Regards

2011/4/26 budi wibowo <bwibowo at gmail.com>

> hi
> any idea to implement prepaid wholesale for FS?
>
>
> thx
>
> budi
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 benkokakao at gmail.com  Tue Apr 26 20:59:07 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Tue, 26 Apr 2011 18:59:07 +0200
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
 with productivity license.
In-Reply-To: <471D76419F9EF642962323D13DF1DF69011E5A@newserver.arneill-py.local>
References: <471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local>
	<BANLkTi=XmNDV51=xoNsViHUYaUZeyMpz_A@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E58@newserver.arneill-py.local>
	<BANLkTi=o1nUa70X_PHU51M_TTGTfnSSdKg@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E59@newserver.arneill-py.local>
	<BANLkTi=9Gu1vOD-KVdCSs3MbNxng+Tqv2w@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E5A@newserver.arneill-py.local>
Message-ID: <BANLkTimzyrNQOLKs_CRRtFKRpw6isCg+AQ@mail.gmail.com>

> Are you the author of the original wiki page?

nope.

Interesting links - have you tried to remove the attendant-settings
from the FS-wiki and to only apply the settings listed in
spip_ssip_3_1_0RevB_relnotes.pdf resp.
http://supportdocs.polycom.com/PolycomService/support/global/documents/support/setup_maintenance/products/voice/spip_ssip_Admin_Guide_SIP_3_1.pdf

So the BLF-configuration is probably working, but not as
straightforward as documented in the wiki(attendant.uri and stuff).


I've updated the wiki again to reflect your findings.

Regards
Christian


From gosha at inbox.ee  Tue Apr 26 21:02:23 2011
From: gosha at inbox.ee (Georgy Goshin)
Date: Tue, 26 Apr 2011 20:02:23 +0300
Subject: [Freeswitch-users] SIP router/switch only configuratoin advice?
References: <0f0501cc0382$ac845fd0$058d1f70$@inbox.ee>
	<FEA85ED4-6352-4D85-8C53-CC63879548F0@ipeva.fr> 
Message-ID: <00a501cc0433$b614c740$223e55c0$@inbox.ee>

David wrote:

> Then check http://wiki.freeswitch.org/wiki/Xml_curl :)

It's interesting module, I think that with this module we can support and use ANY routing scheme, why to use Web/PHP layer? Why not to work directly with the database?





From vetali100 at gmail.com  Tue Apr 26 21:20:41 2011
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Tue, 26 Apr 2011 20:20:41 +0300
Subject: [Freeswitch-users] SIP router/switch only configuratoin advice?
In-Reply-To: <00a501cc0433$b614c740$223e55c0$@inbox.ee>
References: <0f0501cc0382$ac845fd0$058d1f70$@inbox.ee>
	<FEA85ED4-6352-4D85-8C53-CC63879548F0@ipeva.fr>
	<00a501cc0433$b614c740$223e55c0$@inbox.ee>
Message-ID: <BANLkTik_YON=EyqT6fOKx4ERf+JJpCKgPA@mail.gmail.com>

You can work directly on the database from the Lua scripts.

Example of Lua script:

-----------------------------------------------------------------------------------------------------
called_number = argv[1];
local dbh = assert(freeswitch.Dbh("my_db","my_user","my_password"));

dbh:query("select price_buy,price_sell,whatever_else from my_numbers
where called_number like 'xxx%'",
function(row)

price_buy=row.price_buy;
price_sell=row.price_sell;
whatever_else=row.whatever_else;

end);

dbh:release();

--use your variables, execute bridge app, etc

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

Just call the Lua script from your dialplan and you can implement whatever
logic you want.

Example for default dialplan:

<extension name="international">
      <condition field="destination_number" expression="^(00|\+)(\d+)">
       <action application="lua" data="process_call.lua $2"/>
      </condition>
    </extension>

See this page for more info:
http://wiki.freeswitch.org/wiki/Lua_freeswitch_dbh


Vitalie



2011/4/26 Georgy Goshin <gosha at inbox.ee>

> David wrote:
>
> > Then check http://wiki.freeswitch.org/wiki/Xml_curl :)
>
> It's interesting module, I think that with this module we can support and
> use ANY routing scheme, why to use Web/PHP layer? Why not to work directly
> with the database?
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 nico at clickfono.com  Tue Apr 26 21:26:37 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Tue, 26 Apr 2011 13:26:37 -0400
Subject: [Freeswitch-users] webphone app
In-Reply-To: <1481137344-1303084287-cardhu_decombobulator_blackberry.rim.net-323628945-@b26.c2.bise3.blackberry>
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
	<32EF1A658EFC4E5393D8D1A3A486DA31@e1705>
	<BANLkTikdW-5-hDz9A+0mDid_KNEQNhzf6Q@mail.gmail.com>
	<BANLkTi=73yyZ4Chh8OwfiJhNXAVXnrZ8MA@mail.gmail.com>
	<BANLkTi=_tM67uu0MJBsWq5NLnVOEbPHm0w@mail.gmail.com>
	<1481137344-1303084287-cardhu_decombobulator_blackberry.rim.net-323628945-@b26.c2.bise3.blackberry>
Message-ID: <BANLkTikr8stm=8ay7yprHKn2gp5kKQ0UYA@mail.gmail.com>

I'd recommend using rpmtlite/siprtmp, which seems simpler and a lot lighter
than red5:

http://code.google.com/p/siprtmp/

Hope it helps.


On Sun, Apr 17, 2011 at 7:51 PM, Budi wibowo <bwibowo at gmail.com> wrote:

> The phone is working now thx
> ------------------------------
> *From: * budi wibowo <bwibowo at gmail.com>
> *Date: *Sun, 17 Apr 2011 05:11:08 +0700
> *To: *FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> *Subject: *Re: [Freeswitch-users] webphone app
>
> i try that, but got confussion on  installing the application.
> http://code.google.com/p/bigbluebutton/wiki/Red5Phone also not work.
> please give detail installation method
>
>
> On Thu, Apr 7, 2011 at 8:17 AM, Moe Navid <manavid at gmail.com> wrote:
>
>> 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
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 26 21:34:58 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 26 Apr 2011 10:34:58 -0700
Subject: [Freeswitch-users] answering machine detection
In-Reply-To: <BANLkTinOdcB0haPwgf=E-oPX=_f-vOGvwA@mail.gmail.com>
References: <BANLkTinOdcB0haPwgf=E-oPX=_f-vOGvwA@mail.gmail.com>
Message-ID: <BANLkTima7jx=LUZfUrLkB8_wy4HLpGejUg@mail.gmail.com>

Contact consulting at freeswitch.org.
-MC

On Mon, Apr 25, 2011 at 10:05 PM, Sam <u2nsam at gmail.com> wrote:

> Hello,
>
>
>
> Currently i see there is a module for voicemail detection where in it
> detects the beep,
> is there a module for detection of answering machine ( automated voice IVRS
> ),
> or a way to integrate to FS or a code to modify to detect automated voice
> or IVRs ?
>
>
>
>
> 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 infos at madovsky.org  Tue Apr 26 21:38:18 2011
From: infos at madovsky.org (Madovsky)
Date: Tue, 26 Apr 2011 13:38:18 -0400
Subject: [Freeswitch-users] webphone app
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com><32EF1A658EFC4E5393D8D1A3A486DA31@e1705><BANLkTikdW-5-hDz9A+0mDid_KNEQNhzf6Q@mail.gmail.com><BANLkTi=73yyZ4Chh8OwfiJhNXAVXnrZ8MA@mail.gmail.com><BANLkTi=_tM67uu0MJBsWq5NLnVOEbPHm0w@mail.gmail.com><1481137344-1303084287-cardhu_decombobulator_blackberry.rim.net-323628945-@b26.c2.bise3.blackberry>
	<BANLkTikr8stm=8ay7yprHKn2gp5kKQ0UYA@mail.gmail.com>
Message-ID: <B702922119A848E99B8617497D34914B@e1705>

sirtmp is not really stable and use a lot of CPU
  ----- Original Message ----- 
  From: Nicolas Brenner 
  To: FreeSWITCH Users Help 
  Sent: Tuesday, April 26, 2011 1:26 PM
  Subject: Re: [Freeswitch-users] webphone app


  I'd recommend using rpmtlite/siprtmp, which seems simpler and a lot lighter than red5:


  http://code.google.com/p/siprtmp/



  Hope it helps.



  On Sun, Apr 17, 2011 at 7:51 PM, Budi wibowo <bwibowo at gmail.com> wrote:

    The phone is working now thx

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

    From: budi wibowo <bwibowo at gmail.com> 
    Date: Sun, 17 Apr 2011 05:11:08 +0700
    To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
    Subject: Re: [Freeswitch-users] webphone app


    i try that, but got confussion on  installing the application.
    http://code.google.com/p/bigbluebutton/wiki/Red5Phone also not work.
    please give detail installation method 



    On Thu, Apr 7, 2011 at 8:17 AM, Moe Navid <manavid at gmail.com> wrote:

      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 
            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


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


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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



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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





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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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      _______________________________________________
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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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    _______________________________________________
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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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From krice at freeswitch.org  Tue Apr 26 21:38:09 2011
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 26 Apr 2011 12:38:09 -0500
Subject: [Freeswitch-users] webphone app
In-Reply-To: <BANLkTikr8stm=8ay7yprHKn2gp5kKQ0UYA@mail.gmail.com>
Message-ID: <C9DC6D32.1482B%krice@freeswitch.org>

Its Also GPLv3... IANAL So carefully read the GPLv3 license before deploying
such software.

K


On 4/26/11 12:26 PM, "Nicolas Brenner" <nico at clickfono.com> wrote:

> I'd recommend using rpmtlite/siprtmp, which seems simpler and a lot lighter
> than red5:
> 
> http://code.google.com/p/siprtmp/
> 
> Hope it helps.
> 
> 
> On Sun, Apr 17, 2011 at 7:51 PM, Budi wibowo <bwibowo at gmail.com> wrote:
>>  The phone is working now thx
>> 
>> From:  budi wibowo <bwibowo at gmail.com>
>> Date: Sun, 17 Apr 2011 05:11:08 +0700
>> To: FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] webphone app
>> 
>> i try that, but got confussion on ?installing the application.
>> http://code.google.com/p/bigbluebutton/wiki/Red5Phone?also not work.
>> please give detail installation method?
>> 
>> 
>> On Thu, Apr 7, 2011 at 8:17 AM, Moe Navid <manavid at gmail.com> wrote:
>>> 
>>> I tried this about a year ago, it was ok
>>> 
>>> http://code.google.com/p/red5phone <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 <http://boophone.com>
>>>>>>  
>>>>>> ----- Original Message -----
>>>>>>  
>>>>>> From:  budi wibowo <mailto:bwibowo at gmail.com>
>>>>>>  
>>>>>> To: FreeSWITCH Users Help <mailto: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
>>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 nico at clickfono.com  Tue Apr 26 21:45:59 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Tue, 26 Apr 2011 13:45:59 -0400
Subject: [Freeswitch-users] webphone app
In-Reply-To: <B702922119A848E99B8617497D34914B@e1705>
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
	<32EF1A658EFC4E5393D8D1A3A486DA31@e1705>
	<BANLkTikdW-5-hDz9A+0mDid_KNEQNhzf6Q@mail.gmail.com>
	<BANLkTi=73yyZ4Chh8OwfiJhNXAVXnrZ8MA@mail.gmail.com>
	<BANLkTi=_tM67uu0MJBsWq5NLnVOEbPHm0w@mail.gmail.com>
	<1481137344-1303084287-cardhu_decombobulator_blackberry.rim.net-323628945-@b26.c2.bise3.blackberry>
	<BANLkTikr8stm=8ay7yprHKn2gp5kKQ0UYA@mail.gmail.com>
	<B702922119A848E99B8617497D34914B@e1705>
Message-ID: <BANLkTim8syC-twhPGMubZQ_MCVZu00csdA@mail.gmail.com>

It's worked a lot better than red5 for me, using a lot less resources, but
that's my experience.


On Tue, Apr 26, 2011 at 1:38 PM, Madovsky <infos at madovsky.org> wrote:

>  sirtmp is not really stable and use a lot of CPU
>
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From nico at clickfono.com  Tue Apr 26 21:46:52 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Tue, 26 Apr 2011 13:46:52 -0400
Subject: [Freeswitch-users] webphone app
In-Reply-To: <C9DC6D32.1482B%krice@freeswitch.org>
References: <BANLkTikr8stm=8ay7yprHKn2gp5kKQ0UYA@mail.gmail.com>
	<C9DC6D32.1482B%krice@freeswitch.org>
Message-ID: <BANLkTikigEs4FC3kGQbuusCGkdNQ+xpuZw@mail.gmail.com>

I'll definitely have to check that, thanks Ken!


On Tue, Apr 26, 2011 at 1:38 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Its Also GPLv3... IANAL So carefully read the GPLv3 license before
> deploying such software.
>
> K
>
>
>
> On 4/26/11 12:26 PM, "Nicolas Brenner" <nico at clickfono.com> wrote:
>
> I'd recommend using rpmtlite/siprtmp, which seems simpler and a lot lighter
> than red5:
>
> http://code.google.com/p/siprtmp/
>
> Hope it helps.
>
>
> On Sun, Apr 17, 2011 at 7:51 PM, Budi wibowo <bwibowo at gmail.com> wrote:
>
> The phone is working now thx
> ------------------------------
> *From: * budi wibowo <bwibowo at gmail.com>
> *Date: *Sun, 17 Apr 2011 05:11:08 +0700
> *To: *FreeSWITCH Users Help<freeswitch-users at lists.freeswitch.org>
> *Subject: *Re: [Freeswitch-users] webphone app
>
> i try that, but got confussion on  installing the application.
> http://code.google.com/p/bigbluebutton/wiki/Red5Phone also not work.
> please give detail installation method
>
>
> On Thu, Apr 7, 2011 at 8:17 AM, Moe Navid <manavid at gmail.com> wrote:
>
>
> I tried this about a year ago, it was ok
>
> http://code.google.com/p/red5phone <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 <http://boophone.com>
>
>
> ----- Original Message -----
>
> *From:*  budi wibowo <mailto:bwibowo at gmail.com <bwibowo at gmail.com>>
>
> *To:* FreeSWITCH Users Help <mailto:freeswitch-users at lists.freeswitch.org<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
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 david.ponzone at ipeva.fr  Tue Apr 26 23:15:21 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Tue, 26 Apr 2011 21:15:21 +0200
Subject: [Freeswitch-users] continue_on_fail for condition only where
	100 not received?
In-Reply-To: <1303833035795-6306349.post@n2.nabble.com>
References: <1303324320582-6291782.post@n2.nabble.com>
	<245144A7-1F53-400D-A23A-C83DEA96C400@freeswitch.org>
	<BANLkTi=dDmN4s0t3SaLt2ig_cDFXo9sgkQ@mail.gmail.com>
	<BANLkTi=nuCxO2CHM0nOzpgMG8MS5i0P_tw@mail.gmail.com>
	<BANLkTinXAijFk8GxH4u3AL8DNS7Dd-pTVg@mail.gmail.com>
	<1303833035795-6306349.post@n2.nabble.com>
Message-ID: <FDC630B1-3432-45AF-9406-3CDD0040DA3E@ipeva.fr>

Well if you can rely on SIP OPTIONS, you should do that then :)

You can easily check gateway status with:

<condition field="${sofia profile external gwlist up}" expression="\bname_of_your_gateway\b">
</condition>

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 26/04/2011 ? 17:50, peely a ?crit :

> Hi,
> 
> I can guarantee a 100 as the equipment is within my network!
> 
> In this condition I'm only concerned about a dead box scenario, in fact the
> boxes are configured as separate Gateways so I guess I could rely on the
> gateway being available through OPTIONS pings and being taken out of service
> automatically.
> 
> So either a CONTINUE_ON_FAIL which acts on no 100 response within x seconds
> or will only attempt a second gateway if the first gateway is disabled would
> do.
> 
> Any ideas?
> 
> On the other topics, it's quite common for User Agents to offer an "always
> send 100 trying" option for this specific reason, although you need to know
> it is enabled I guess.
> 
> 
> Thanks,
> 
> 
> 
> Neil.
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/continue-on-fail-for-condition-only-where-100-not-received-tp6291782p6306349.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 davidwaf at gmail.com  Tue Apr 26 23:25:29 2011
From: davidwaf at gmail.com (David Wafula)
Date: Tue, 26 Apr 2011 21:25:29 +0200
Subject: [Freeswitch-users] fs prepaid wholesale
In-Reply-To: <BANLkTiknes7esHBSZTMmhiwWEQQ3D-fMmw@mail.gmail.com>
References: <BANLkTimw3J_M6tPPQ1q-XPmF7q8=64XdPA@mail.gmail.com>
	<BANLkTiknes7esHBSZTMmhiwWEQQ3D-fMmw@mail.gmail.com>
Message-ID: <BANLkTin_4aYS4poLF=s_18MLyU5cCMD8bw@mail.gmail.com>

2011/4/26 Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com>

> mod_lcr? mod_nible? a LUA script? a custom C module?
>
> I think you have several options. Look at the wiki for more info.
>
>
>

http://wiki.freeswitch.org/wiki/Mod_nibblebill#Billing_.28Pre-pay.29
<http://wiki.freeswitch.org/wiki/Mod_nibblebill#Billing_.28Pre-pay.29>
-- 
David Wafula
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From msc at freeswitch.org  Tue Apr 26 23:53:51 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 26 Apr 2011 12:53:51 -0700
Subject: [Freeswitch-users] absolute_codec_string question
In-Reply-To: <BANLkTin32Ym=4TP4C_HOubGTCp0H_WOa_w@mail.gmail.com>
References: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>
	<4DB59C06.3090706@cupis.co.uk>
	<BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>
	<4DB5A53D.50201@cupis.co.uk>
	<BANLkTi=gh6AyV9C-fQaR6cY-x0=VX=XDeg@mail.gmail.com>
	<4DB5BB7B.4060409@cupis.co.uk>
	<BANLkTini7nLgyNstmf2GxzK9mcwOdG03Dw@mail.gmail.com>
	<BANLkTikDOK0YJoF8s0wnrz00Ah6MXF5-Gg@mail.gmail.com>
	<BANLkTik4mv7zWRndhM4_KxvvU-ybC-cLUQ@mail.gmail.com>
	<BANLkTin32Ym=4TP4C_HOubGTCp0H_WOa_w@mail.gmail.com>
Message-ID: <BANLkTi=ZpWZeLjzbMBee44xf0LEY2t-yiQ@mail.gmail.com>

This email is a perfect example of community members stepping up! Thanks
Steven!!
-MC

On Tue, Apr 26, 2011 at 1:20 AM, Steven Ayre <steveayre at gmail.com> wrote:

> {ignore_early_media,absolute_codec_string='G729,PCMU'}
>>
>
> The problem is that FS splits the variable names using commas. When there's
> a comma in a value it gets confused. Using quotation marks doesn't make any
> difference - they're treated as part of the value.
>
> So FS is seeing this:
>
> ignore_early_media
> absolute_codec_string='G729
> PCMU'
>
> So the only codec that gets set is the G729.
>
> There is a workaround for this. You want to use
> {ignore_early_media,absolute_codec_string=^^:G729:PCMU}.
>
> The ^^ at the start of a value tells FS 'treat the following character as a
> comma'. Now FS sees:
>
>
> ignore_early_media
> absolute_codec_string=^^:G729:PCMU
>
> And sets absolute_codec_string to G729,PCMU.
>
> -Steve
>
>
>
> On 25 April 2011 21:44, Nicolas Brenner <nico at clickfono.com> wrote:
>
>> Oops, sorry you are right, I copied it wrong, doesn't fix it though.
>>
>>
>> On Mon, Apr 25, 2011 at 3:35 PM, Vitalie Colosov <vetali100 at gmail.com>wrote:
>>
>>> Maybe I am wrong, but I see a typo in the following string, after "
>>> ignore_early_media":
>>>
>>> originate {ignore_early_media,absolute_codec_string=G729}user/1001
>>> &bridge(user/1000)
>>>
>>> It should be:
>>>
>>> originate {ignore_early_media=true,absolute_codec_string=G729}user/1001
>>> &bridge(user/1000)
>>>
>>> Vitalie
>>>
>>>
>>> 2011/4/25 Nicolas Brenner <nico at clickfono.com>
>>>
>>>> Thanks, yes, here you go:
>>>>
>>>> I'm making calls from the console to an X-Lite registered on extension
>>>> 1001 which doesn't support G729. Calls 1 and 3 below fail because the codec
>>>> options are not supported, and even though on the third call the
>>>> absolute_codec_string variable should be 'G729,PCMU', it is not offering
>>>> both codecs.
>>>>
>>>>
>>>> On http://pastebin.freeswitch.org/16167 the SIP trace for:
>>>>
>>>> originate {ignore_early_media,absolute_codec_string=G729}user/1001
>>>> &bridge(user/1000)
>>>>
>>>> This is the SDP:
>>>>
>>>>    v=0
>>>>    o=FreeSWITCH 1303734153 1303734154 IN IP4 127.0.0.1
>>>>    s=FreeSWITCH
>>>>    c=IN IP4 127.0.0.1
>>>>    t=0 0
>>>>     m=audio 24036 RTP/AVP 18 101 13
>>>>    a=rtpmap:101 telephone-event/8000
>>>>    a=fmtp:101 0-16
>>>>    a=ptime:20
>>>>
>>>>
>>>> On http://pastebin.freeswitch.org/16169 the SIP trace for:
>>>>
>>>> originate {ignore_early_media,absolute_codec_string=PCMU}user/1001
>>>> &bridge(user/1000)
>>>>
>>>> This is the SDP:
>>>>
>>>>    v=0
>>>>    o=FreeSWITCH 1303733515 1303733516 IN IP4 127.0.0.1
>>>>     s=FreeSWITCH
>>>>    c=IN IP4 127.0.0.1
>>>>    t=0 0
>>>>    m=audio 24896 RTP/AVP 0 101 13
>>>>    a=rtpmap:101 telephone-event/8000
>>>>    a=fmtp:101 0-16
>>>>    a=ptime:20
>>>>
>>>>
>>>> On http://pastebin.freeswitch.org/16170 the SIP trace for:
>>>>
>>>> originate
>>>> {ignore_early_media,absolute_codec_string='G729,PCMU'}user/1001
>>>> &bridge(user/1000)
>>>>
>>>> This is the SDP:
>>>>
>>>>    v=0
>>>>    o=FreeSWITCH 1303733520 1303733521 IN IP4 127.0.0.1
>>>>    s=FreeSWITCH
>>>>    c=IN IP4 127.0.0.1
>>>>    t=0 0
>>>>     m=audio 25138 RTP/AVP 18 101 13
>>>>    a=rtpmap:101 telephone-event/8000
>>>>    a=fmtp:101 0-16
>>>>    a=ptime:20
>>>>
>>>>
>>>>
>>>>
>>>> On Mon, Apr 25, 2011 at 2:20 PM, Paul Cupis <paul at cupis.co.uk> wrote:
>>>>
>>>>> On 25/04/11 18:08, Nicolas Brenner wrote:
>>>>> > I'm trying to use the
>>>>> > absolute_codec_string with originate from the console, like so:
>>>>> >
>>>>> > originate
>>>>> >
>>>>> {ignore_early_media,verbose_sdp=true,absolute_codec_string='G729,PMCU'}sofia/gateway/mygateway/444444
>>>>> > &bridge(user/1001)
>>>>> >
>>>>> > Paul, I am using {absolute_codec_string='G729,PCMU'}, and I get the
>>>>> same as
>>>>> > if I don't quote the string, or if I just specify one codec:
>>>>>
>>>>> Can you provide (on pastebin.freeswitch.org) a complete log of a call,
>>>>> please?
>>>>>
>>>>> 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
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Apr 27 02:42:59 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 26 Apr 2011 23:42:59 +0100
Subject: [Freeswitch-users] absolute_codec_string question
In-Reply-To: <BANLkTi=ZpWZeLjzbMBee44xf0LEY2t-yiQ@mail.gmail.com>
References: <BANLkTimpVj9LUabzWxv8HkpAdf5++b1e-g@mail.gmail.com>
	<4DB59C06.3090706@cupis.co.uk>
	<BANLkTintKcd08ZT6cyiehnYzDiUtJh1dww@mail.gmail.com>
	<4DB5A53D.50201@cupis.co.uk>
	<BANLkTi=gh6AyV9C-fQaR6cY-x0=VX=XDeg@mail.gmail.com>
	<4DB5BB7B.4060409@cupis.co.uk>
	<BANLkTini7nLgyNstmf2GxzK9mcwOdG03Dw@mail.gmail.com>
	<BANLkTikDOK0YJoF8s0wnrz00Ah6MXF5-Gg@mail.gmail.com>
	<BANLkTik4mv7zWRndhM4_KxvvU-ybC-cLUQ@mail.gmail.com>
	<BANLkTin32Ym=4TP4C_HOubGTCp0H_WOa_w@mail.gmail.com>
	<BANLkTi=ZpWZeLjzbMBee44xf0LEY2t-yiQ@mail.gmail.com>
Message-ID: <C1FB4F67-D5CB-45BE-AF4D-B9806D7518B5@gmail.com>

No problem! :)

Steve on iPhone

On 26 Apr 2011, at 20:53, Michael Collins <msc at freeswitch.org> wrote:

> This email is a perfect example of community members stepping up! Thanks Steven!!
> -MC
> 
> On Tue, Apr 26, 2011 at 1:20 AM, Steven Ayre <steveayre at gmail.com> wrote:
> {ignore_early_media,absolute_codec_string='G729,PCMU'}
> 
> The problem is that FS splits the variable names using commas. When there's a comma in a value it gets confused. Using quotation marks doesn't make any difference - they're treated as part of the value.
> 
> So FS is seeing this:
> 
> ignore_early_media
> absolute_codec_string='G729
> PCMU'
> 
> So the only codec that gets set is the G729.
> 
> There is a workaround for this. You want to use {ignore_early_media,absolute_codec_string=^^:G729:PCMU}.
> 
> The ^^ at the start of a value tells FS 'treat the following character as a comma'. Now FS sees:
> 
> 
> ignore_early_media
> absolute_codec_string=^^:G729:PCMU
> 
> And sets absolute_codec_string to G729,PCMU.
> 
> -Steve
> 
> 
> 
> On 25 April 2011 21:44, Nicolas Brenner <nico at clickfono.com> wrote:
> Oops, sorry you are right, I copied it wrong, doesn't fix it though.
> 
> 
> On Mon, Apr 25, 2011 at 3:35 PM, Vitalie Colosov <vetali100 at gmail.com> wrote:
> Maybe I am wrong, but I see a typo in the following string, after "ignore_early_media":
> 
> originate {ignore_early_media,absolute_codec_string=G729}user/1001 &bridge(user/1000)
> 
> It should be:
> 
> originate {ignore_early_media=true,absolute_codec_string=G729}user/1001 &bridge(user/1000)
> 
> Vitalie
> 
> 
> 2011/4/25 Nicolas Brenner <nico at clickfono.com>
> Thanks, yes, here you go:
> 
> I'm making calls from the console to an X-Lite registered on extension 1001 which doesn't support G729. Calls 1 and 3 below fail because the codec options are not supported, and even though on the third call the absolute_codec_string variable should be 'G729,PCMU', it is not offering both codecs.
> 
> 
> On http://pastebin.freeswitch.org/16167 the SIP trace for:
> 
> originate {ignore_early_media,absolute_codec_string=G729}user/1001 &bridge(user/1000)
> 
> This is the SDP:
> 
>    v=0
>    o=FreeSWITCH 1303734153 1303734154 IN IP4 127.0.0.1
>    s=FreeSWITCH
>    c=IN IP4 127.0.0.1
>    t=0 0
>    m=audio 24036 RTP/AVP 18 101 13
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
> 
> 
> On http://pastebin.freeswitch.org/16169 the SIP trace for:
> 
> originate {ignore_early_media,absolute_codec_string=PCMU}user/1001 &bridge(user/1000)
> 
> This is the SDP:
> 
>    v=0
>    o=FreeSWITCH 1303733515 1303733516 IN IP4 127.0.0.1
>    s=FreeSWITCH
>    c=IN IP4 127.0.0.1
>    t=0 0
>    m=audio 24896 RTP/AVP 0 101 13
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
> 
> 
> On http://pastebin.freeswitch.org/16170 the SIP trace for:
> 
> originate {ignore_early_media,absolute_codec_string='G729,PCMU'}user/1001 &bridge(user/1000)
> 
> This is the SDP:
> 
>    v=0
>    o=FreeSWITCH 1303733520 1303733521 IN IP4 127.0.0.1
>    s=FreeSWITCH
>    c=IN IP4 127.0.0.1
>    t=0 0
>    m=audio 25138 RTP/AVP 18 101 13
>    a=rtpmap:101 telephone-event/8000
>    a=fmtp:101 0-16
>    a=ptime:20
> 
> 
> 
> 
> On Mon, Apr 25, 2011 at 2:20 PM, Paul Cupis <paul at cupis.co.uk> wrote:
> On 25/04/11 18:08, Nicolas Brenner wrote:
> > I'm trying to use the
> > absolute_codec_string with originate from the console, like so:
> >
> > originate
> > {ignore_early_media,verbose_sdp=true,absolute_codec_string='G729,PMCU'}sofia/gateway/mygateway/444444
> > &bridge(user/1001)
> >
> > Paul, I am using {absolute_codec_string='G729,PCMU'}, and I get the same as
> > if I don't quote the string, or if I just specify one codec:
> 
> Can you provide (on pastebin.freeswitch.org) a complete log of a call,
> please?
> 
> 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
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Apr 27 03:00:50 2011
From: bwibowo at gmail.com (budi wibowo)
Date: Wed, 27 Apr 2011 06:00:50 +0700
Subject: [Freeswitch-users] fs prepaid wholesale
In-Reply-To: <BANLkTin_4aYS4poLF=s_18MLyU5cCMD8bw@mail.gmail.com>
References: <BANLkTimw3J_M6tPPQ1q-XPmF7q8=64XdPA@mail.gmail.com>
	<BANLkTiknes7esHBSZTMmhiwWEQQ3D-fMmw@mail.gmail.com>
	<BANLkTin_4aYS4poLF=s_18MLyU5cCMD8bw@mail.gmail.com>
Message-ID: <BANLkTimH35Bzb_owuVFSn5HHeuFTPTTizw@mail.gmail.com>

i have implemented nibblebill but i oonly use it user for my user defined
in /usr/local/freeswitch/conf/directory/default and it works.
but implement for wholesale i have no idea, in this case i will receive
incoming sip connection from my partner and nibble it. i finished reading
the Fs book from packt publisihing but still no clue

any advice is highly appreciated


br

budi


On Wed, Apr 27, 2011 at 2:25 AM, David Wafula <davidwaf at gmail.com> wrote:

>
>
> 2011/4/26 Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com>
>
>> mod_lcr? mod_nible? a LUA script? a custom C module?
>>
>> I think you have several options. Look at the wiki for more info.
>>
>>
>>
>
> http://wiki.freeswitch.org/wiki/Mod_nibblebill#Billing_.28Pre-pay.29
> <http://wiki.freeswitch.org/wiki/Mod_nibblebill#Billing_.28Pre-pay.29>
> --
> David Wafula
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 27 03:21:31 2011
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 26 Apr 2011 16:21:31 -0700
Subject: [Freeswitch-users] fs prepaid wholesale
In-Reply-To: <BANLkTimH35Bzb_owuVFSn5HHeuFTPTTizw@mail.gmail.com>
References: <BANLkTimw3J_M6tPPQ1q-XPmF7q8=64XdPA@mail.gmail.com>
	<BANLkTiknes7esHBSZTMmhiwWEQQ3D-fMmw@mail.gmail.com>
	<BANLkTin_4aYS4poLF=s_18MLyU5cCMD8bw@mail.gmail.com>
	<BANLkTimH35Bzb_owuVFSn5HHeuFTPTTizw@mail.gmail.com>
Message-ID: <BANLkTikpKvam6_PuYPtdgZgxWywymZPdoQ@mail.gmail.com>

On Tue, Apr 26, 2011 at 4:00 PM, budi wibowo <bwibowo at gmail.com> wrote:

> i have implemented nibblebill but i oonly use it user for my user defined
> in /usr/local/freeswitch/conf/directory/default and it works.
> but implement for wholesale i have no idea, in this case i will receive
> incoming sip connection from my partner and nibble it. i finished reading
> the Fs book from packt publisihing but still no clue
>
> any advice is highly appreciated
>

If you don't know how to do it then you need to hire someone who does. There
is *no way* that you're going to gain the knowledge you need just by
googling around and asking questions here on the mailing list. I know that's
not what you want to hear but it's the honest truth.

-MC
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From Dennis.Young at supportkids.com  Wed Apr 27 00:45:08 2011
From: Dennis.Young at supportkids.com (Dennis Young)
Date: Tue, 26 Apr 2011 15:45:08 -0500
Subject: [Freeswitch-users] Freetdm - Wanpipe - ftmod_sangoma_isdn
Message-ID: <32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A467@jehuty.supportkids.com>

All,

Last month I upgraded from using ftmod_sangoma_boost to ftmod_sangoma_isdn, as a result, I'm no longer getting caller_name on inbound calls.  I can see the caller_number but the caller_name is blank.

I'm running windows 32-bit, sangoma A102 card, wanpipe_6_42_1, libsng_isdn-7.1.0, GIT-Head from 4/14/11.

I'm sure it's a configuration issue because it was working with boost.  Can someone point me in the right direction?

------- wanpipe.conf -----------
[defaults]
codec_ms => 20
wink_ms => 150
flash_ms => 750

------ freetdm.conf -----------
[span wanpipe wp1]
trunk_type => T1
group => grp1
b-channel => 1:1-23
d-channel => 1:24

[span wanpipe wp2]
trunk_type => T1
group => grp1
b-channel => 2:1-23
d-channel => 2:24

--------- freetdm.conf.xml --------
<configuration name="freetdm.conf" description="FreeTDM Configuration">
  <settings>
    <param name="debug" value="0"/>
  </settings>
  <config_profiles>
     <profile name="twtelecom">
<param name="facility" value="no" />
     </profile>
   </config_profiles>

  <sangoma_pri_spans>
    <span name="wp1" cfgprofile="twtelecom">
                                <param name="switchtype" value="national" />
                                <param name="interface" value="cpe"/>
                                <param name="dialplan" value="XML"/>
                                <param name="context" value="public" />
    </span>
    <span name="wp2" cfgprofile="twtelecom">
                                <param name="switchtype" value="national" />
                                <param name="interface" value="cpe"/>
                                <param name="dialplan" value="XML"/>
                                <param name="context" value="public" />
    </span>
  </sangoma_pri_spans>
</configuration>


...dly

Notice: This electronic transmission and its attachments are confidential and protected by applicable state and/or federal law.  Any use, reading, dissemination, distribution, copying or storage of this information by anyone other than the intended recipient is strictly prohibited.  If you are not the intended recipient, please immediately notify the sender by return email or telephone and delete this message and its attachments from your system.
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From brian at freeswitch.org  Wed Apr 27 03:46:54 2011
From: brian at freeswitch.org (Brian West)
Date: Tue, 26 Apr 2011 18:46:54 -0500
Subject: [Freeswitch-users] Freetdm - Wanpipe - ftmod_sangoma_isdn
In-Reply-To: <32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A467@jehuty.supportkids.com>
References: <32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A467@jehuty.supportkids.com>
Message-ID: <6404824F-841B-4491-8CCD-CD0D65E8D8BB@freeswitch.org>

Set this to YES

/b

On Apr 26, 2011, at 3:45 PM, Dennis Young wrote:

> <param name="facility" value="no" />



From b_ball_henry at hotmail.com  Wed Apr 27 05:13:57 2011
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Wed, 27 Apr 2011 09:13:57 +0800
Subject: [Freeswitch-users] webphone app
In-Reply-To: <BANLkTim8syC-twhPGMubZQ_MCVZu00csdA@mail.gmail.com>
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
	<32EF1A658EFC4E5393D8D1A3A486DA31@e1705>
	<BANLkTikdW-5-hDz9A+0mDid_KNEQNhzf6Q@mail.gmail.com>
	<BANLkTi=73yyZ4Chh8OwfiJhNXAVXnrZ8MA@mail.gmail.com>
	<BANLkTi=_tM67uu0MJBsWq5NLnVOEbPHm0w@mail.gmail.com>
	<1481137344-1303084287-cardhu_decombobulator_blackberry.rim.net-323628945-@b26.c2.bise3.blackberry>
	<BANLkTikr8stm=8ay7yprHKn2gp5kKQ0UYA@mail.gmail.com>
	<B702922119A848E99B8617497D34914B@e1705>
	<BANLkTim8syC-twhPGMubZQ_MCVZu00csdA@mail.gmail.com>
Message-ID: <BANLkTikL+JKNsbaEXbdaEBzK4hdhrk0YkA@mail.gmail.com>

Do you have to develop a flash client from scratch or this project has a
sample client to use?

Henry

On Wed, Apr 27, 2011 at 1:45 AM, Nicolas Brenner <nico at clickfono.com> wrote:

> It's worked a lot better than red5 for me, using a lot less resources, but
> that's my experience.
>
>
> On Tue, Apr 26, 2011 at 1:38 PM, Madovsky <infos at madovsky.org> wrote:
>
>>  sirtmp is not really stable and use a lot of CPU
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 anton.vazir at gmail.com  Wed Apr 27 10:35:53 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 27 Apr 2011 11:35:53 +0500
Subject: [Freeswitch-users] Python vs LUA
Message-ID: <BANLkTinHmV=GTEQKoGuzvmcjBxM02Nd4xg@mail.gmail.com>

Hello!
Cannot get answer for myself for languages processing overhead/preference,

I personally would like to use Python for scripting under Freeswitch,
but considering some recent list activity about low CPS while using
python scripts, etc
querstioning, maybe it's better to use LUA instead? WIKI or list does
not answer this questions in satisfying manner unfortunatelly. For DB
intensive scripts - DB connections pool is a very important stuff,
having direct influence performance and CPS specifically.

It looks that lua might be lighter and more tightly integrated to FS,
but if processing overhead difference is just a few times python is
still preferred for me (and many others) since much more
powerfull/libraryreach/etc.
I'ts would be just nice to clear this question.

1. Is there any comparisions on plain processing overhead?
2. Any data on memory consumtption per active python process?
2. Is there a freeswitch DB pool connection interface like for LUA in
mod_python?
3. If not - what would be the known technique to use external DB
connection pool from inside the FS python script?
4. Seems logical, but please confirm/reject - once executed (compiled
to the bytecode) - the second time the python script under FS will be
executed as precompiled bytecode?

Regards,
Anton.


From vetali100 at gmail.com  Wed Apr 27 10:47:45 2011
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Wed, 27 Apr 2011 09:47:45 +0300
Subject: [Freeswitch-users] attended transfer to gateway
In-Reply-To: <BANLkTi=u9d5NWeO=Z1Rtnn_7+qAqb1cuKQ@mail.gmail.com>
References: <BANLkTinB6uFAWDnq-Ot1tiSPi99pu0Oruw@mail.gmail.com>
	<BANLkTikkF95DCGgpQJ-kKCR-U_=X7fF5+g@mail.gmail.com>
	<BANLkTinAXp_u4+0OwNjY5dEDH5YHmFR-fw@mail.gmail.com>
	<65A0D45D-0666-493C-B53A-D9DC882EE77C@freeswitch.org>
	<BANLkTin8Vdg=a3em_2O1bkX3C5qoGLCBBg@mail.gmail.com>
	<BANLkTi=u9d5NWeO=Z1Rtnn_7+qAqb1cuKQ@mail.gmail.com>
Message-ID: <BANLkTin=3jrY0kzjw-CsOZM07O4XtVBtWQ@mail.gmail.com>

I had the same problem when I was testing attended transfer last year -
pressing "0" just hung up the channel.
But I was thinking that it is because I messed with some configuration
settings, and decided to lower priority on this problem, until I will really
need this.

Now it looks like it is indeed a general problem.

Is anybody on the list have this functionality working?
Meaning pressing "0" should convert the attended transfer to the 3-way call
as described here:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer


Thank you,
Vitalie




2011/4/26 elijah <elijah at crankenstein.com>

> So I'm at the limit of my modest capabilities for troubleshooting this
> further. There is a workaround I am thinking of by which the documented
> 3-way conferencing feature of att-xfer could be simulated using a full-on
> conference bridge and if this is the end of any suggestions I receive on
> this thread I will pursue it. I have implemented att-xfer in a manner nearly
> identical to the online documentation and default config as is practical and
> within my understanding. I have documented here my configuration and
> corresponding logs in hope that someone else had encountered a similar
> problem and could advise me of a solution. Hopefully my experience does not
> indicate a larger issue within the att-xfer module, but if it does I hope
> the following documentation is useful.
>
> I understand FreeSwitch is an enormous commitment for everyone who has
> contributed to source and I hope to be in a position to personally
> contribute in the future whether with my own time or financially. If I can
> help now to get this thread resolved and prevent someone else from having an
> implementation issue please let me know.
>
> thanks,
> elijah
>
>
> On Thu, Apr 21, 2011 at 10:15 AM, elijah <elijah at crankenstein.com> wrote:
>
>>         <action application="bind_meta_app" data="1 b s
>> execute_extension::ax XML callsdirect"/>
>>
>>     <extension name="bind_receiver">
>>       <condition field="destination_number" expression="^ax$">
>>         <action application="answer"/>
>>         <action application="read" data="10 10
>> 'tone_stream://%(10000,0,350,440)' attxfer_callthis 30000 *"/>
>>         <action application="execute_extension" data="attended_xfer XML
>> callsdirect"/>
>>       </condition>
>>     </extension>
>>
>>     <extension name="attended_xfer">
>>       <condition field="destination_number" expression="^attended_xfer$"/>
>>       <condition field="${attxfer_callthis}" expression="^(\d+)$">
>>         <action application="set" data="continue_on_fail=true"/>
>>         <action application="set" data="origination_cancel_key=#"/>
>>          <action application="att_xfer"
>> data="sofia/gateway/onesource/${attxfer_callthis}"/>
>>         <anti-action application="eval" data="cancel transfer"/>
>>       </condition>
>>     </extension>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 arielmonaco at flylabs.com  Wed Apr 27 04:24:59 2011
From: arielmonaco at flylabs.com (Ariel Monaco)
Date: Tue, 26 Apr 2011 21:24:59 -0300
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
Message-ID: <C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>

We had high CPU utilization peaks in the past, which lead to some audio issues (clipping). We were using debian at that time, which was a customer-side requirement.

I'm not a kernel guru but I remember this had something to do with kernel timer cycles and the issue was address by adding "divider=10" or "divider=100" as a kernel's boot loader option.

My 2 cents,
Ariel

On Apr 21, 2011, at 13:24 , Antonio Teixeira wrote:

> Hello List.
> 
> I'm currently integrating an IVR in python together with freeswitch using mod_python and ESL and my life has been well until ...
> The flow of calls went over 80 simultaneous calls.
> Now freeswitch starts sending packets with huge delays ( even when establishing the call , mainly the 200 ) and firing up the IVR with tons of delay up to 20 seconds.
> 
> So i searched the wiki forums and mailing list:
> 
> Put freeswitch on a diet , trimmed modules.conf
> Played with the ulimit stuff.
> Played with the IVRS to reduce load to a minimum and i was able to squeeze more 5 calls of performance.
> 
> The problem is :
> 
> Top shows 
> top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92, 1.76, 1.78
> Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0 zombie
> Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,  0.5%si,  0.0%st
> Mem:   8193336k total,  1639156k used,  6554180k free,   177208k buffers
> Swap: 19534904k total,        0k used, 19534904k free,  1062272k cached
> 
>   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND                                                                                         
> 31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85 freeswitch   
> 
> Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
> 
> 
> Some basic System Info :
> Debian 6.0 ( i heard the timming module is affected by Debian , but if the CPU % gets lower than 95% everything will be more stable)
> Python 2.5
> 
> 2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
> 8 GB of Ram
> 
> as you can see 94 % of the "Cpu Power" is sleeping :\
> 
> 
> It appears freeswitch is only capable of using let's say "one cpu"/thread ??
> Do you guys recommend simply starting more instances or redoing the IVR stuff.
> 
> 
> Hope you guys can help me out.
> 
> Thanks
> Ant?nio Teixeira
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Ariel Monaco ? Systems Engineer
Flylabs - Open Source Telecommunications and  IT Consultants

Address: Potos? 4456 C1199ACP - Buenos Aires - Argentina
Web: http://flylabs.com
E-Mail: arielmonaco at flylabs.com
Tel. +54 (11) 4982-2689, +1 (315) 688-7333



From paul at cupis.co.uk  Wed Apr 27 11:50:13 2011
From: paul at cupis.co.uk (Paul Cupis)
Date: Wed, 27 Apr 2011 08:50:13 +0100
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
Message-ID: <4DB7CAB5.9080801@cupis.co.uk>

On 27/04/11 01:24, Ariel Monaco wrote:
> We had high CPU utilization peaks in the past, which lead to some 
> audio issues (clipping). We were using debian at that time, which
> was a customer-side requirement.
> 
> I'm not a kernel guru but I remember this had something to do with 
> kernel timer cycles and the issue was address by adding "divider=10" 
> or "divider=100" as a kernel's boot loader option.

Interesting. This kernel option is designed to reduce the valy of
CONFIG_HZ at run=time, so if you have a 1000Hz kernel, setting
divider=10 at boot time will effectively give you a 100Hz kernel.

This seems to fly in the face of other recommendations on this kernel
setting!

Debian default kernel sets CONFIG_HZ=100
CentOS default kernel sets CONFIG_HZ=1000

Regards,



From freeswitch at priv.de  Wed Apr 27 12:27:38 2011
From: freeswitch at priv.de (Markus Mueller)
Date: Wed, 27 Apr 2011 10:27:38 +0200
Subject: [Freeswitch-users] Out-of-memory problem (dmesg: [1950219.067153]
 Killed process 12772 (freeswitch))
Message-ID: <4DB7D37A.9070404@priv.de>

Hello Freeswitch,

after running Freeswitch for some weeks the system resulted in a 
out-of-memory problem. No ssh-login is possible, only on console. Please see

http://projekte.priv.de/freeswitch/dmesg

for the kernel messages. The memory statistics is interesting: In the

http://projekte.priv.de/freeswitch/memory

, wich is /proc/meminfo, you can see that SUnreclaim is very big...

I am using Debian Squezze with the dahdi-source package to access the, 
libpri-1.4.12-beta3 and the git fetched with "git clone git:/ 
/git.freeswitch.org/freeswitch.git " from "Wed Mar 2 20:50:33 2011 
-0600" (last entry in git log).

I am glad for any hits or suggestions!

Regards,
Markus Mueller



From freeswitch at priv.de  Wed Apr 27 12:44:30 2011
From: freeswitch at priv.de (Markus Mueller)
Date: Wed, 27 Apr 2011 10:44:30 +0200
Subject: [Freeswitch-users] FreeTDM: ISDN-Port (D-Channel?) useable only one
	time
Message-ID: <4DB7D76E.4020900@priv.de>

Hi Freeswitch users,

if I get an incomming call (FAX), the call is answered and successfully 
processed, this channel don't get any new calls in until the channel has 
been reseted by "ftdm stop" and "ftdm start".

Please see the log file on

http://projekte.priv.de/freeswitch/log

You can see there that a fax is comming in on three ISDN lines, but only 
accepted on one of them. On the first fax it comes in on all three ISDN 
lines, on the second fax only on two ISDN lines, the third fax is only 
comming in on one line and all further faxes can't be accepted. After 
resetet with "ftdm stop" and "ftdm start" it is comming in again.

Further config files can be found on

http://projekte.priv.de/freeswitch/freetdm.conf.xml
http://projekte.priv.de/freeswitch/dialplan.xml
http://projekte.priv.de/freeswitch/freetdm.conf
http://projekte.priv.de/freeswitch/dahdi_system.conf

I am using Debian Squezze with the dahdi-source package to access the, 
libpri-1.4.12-beta3 and the git fetched with "git clone git:/ 
/git.freeswitch.org/freeswitch.git " from "Wed Mar 2 20:50:33 2011 
-0600" (last entry in git log). Als hardware I have three singleport HFC 
Cards:

04:00.0 Network controller: Cologne Chip Designs GmbH ISDN network 
controller [HFC-PCI] (rev 02)
04:01.0 Network controller: Cologne Chip Designs GmbH ISDN network 
controller [HFC-PCI] (rev 02)
04:02.0 Network controller: Cologne Chip Designs GmbH ISDN network 
controller [HFC-PCI] (rev 02)

root at sip:/usr/local/freeswitch/conf# dahdi_scan
[1]
active=yes
alarms=OK
description=HFC-S PCI A ISDN card 0 [TE]
name=ZTHFC1
manufacturer=Cologne Chips
devicetype=HFC-S PCI-A ISDN
location=PCI Bus 04 Slot 01
basechan=1
totchans=3
irq=16
type=digital-TE
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=AMI
framing_opts=CCS
coding=AMI
framing=CCS
[2]
active=yes
alarms=OK
description=HFC-S PCI A ISDN card 1 [TE]
name=ZTHFC2
manufacturer=Cologne Chips
devicetype=HFC-S PCI-A ISDN
location=PCI Bus 04 Slot 02
basechan=4
totchans=3
irq=17
type=digital-TE
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=AMI
framing_opts=CCS
coding=AMI
framing=CCS
[3]
active=yes
alarms=OK
description=HFC-S PCI A ISDN card 2 [TE]
name=ZTHFC3
manufacturer=Cologne Chips
devicetype=HFC-S PCI-A ISDN
location=PCI Bus 04 Slot 03
basechan=7
totchans=3
irq=18
type=digital-TE
syncsrc=0
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=AMI
framing_opts=CCS
coding=AMI
framing=CCS
root at sip:/usr/local/freeswitch/conf#

I am glad for any hits or suggestions!

Regards,
Markus Mueller



From brian at microcomaustralia.com.au  Wed Apr 27 13:37:03 2011
From: brian at microcomaustralia.com.au (Brian May)
Date: Wed, 27 Apr 2011 19:37:03 +1000
Subject: [Freeswitch-users] Out-of-memory problem (dmesg:
 [1950219.067153] Killed process 12772 (freeswitch))
In-Reply-To: <4DB7D37A.9070404@priv.de>
References: <4DB7D37A.9070404@priv.de>
Message-ID: <BANLkTin4oAehA9Kz3PcqgvS9jQ+954YNrw@mail.gmail.com>

On 27 April 2011 18:27, Markus Mueller <freeswitch at priv.de> wrote:

> I am glad for any hits or suggestions!
>

I have also also had the oom-killer kill my freeswitch. Trouble is by the
time I get to the system, the problem has gone away, and I can restart
freeswitch without any problems.

On the last occasion I was getting flooded with incoming SIP REGISTER
packets, so I suspect that may have been the culprit at least this last time
(see archives).

(Not absolutely convinced this was an attack, it seem to stop immediately
after I restarted freeswitch, so maybe somebodies mis-configured system was
trying desperately to get a response, which wasn't going to happen while
freeswitch was dead? Still seems strange though that somebody would be
trying to contact my server...)

Haven't had any recent problems.
-- 
Brian May <brian at microcomaustralia.com.au>
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From nico at clickfono.com  Wed Apr 27 13:52:19 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Wed, 27 Apr 2011 05:52:19 -0400
Subject: [Freeswitch-users] webphone app
In-Reply-To: <BANLkTikL+JKNsbaEXbdaEBzK4hdhrk0YkA@mail.gmail.com>
References: <BANLkTi=p8Osyho+aXV5yXQa991e5Xe8n9Q@mail.gmail.com>
	<32EF1A658EFC4E5393D8D1A3A486DA31@e1705>
	<BANLkTikdW-5-hDz9A+0mDid_KNEQNhzf6Q@mail.gmail.com>
	<BANLkTi=73yyZ4Chh8OwfiJhNXAVXnrZ8MA@mail.gmail.com>
	<BANLkTi=_tM67uu0MJBsWq5NLnVOEbPHm0w@mail.gmail.com>
	<1481137344-1303084287-cardhu_decombobulator_blackberry.rim.net-323628945-@b26.c2.bise3.blackberry>
	<BANLkTikr8stm=8ay7yprHKn2gp5kKQ0UYA@mail.gmail.com>
	<B702922119A848E99B8617497D34914B@e1705>
	<BANLkTim8syC-twhPGMubZQ_MCVZu00csdA@mail.gmail.com>
	<BANLkTikL+JKNsbaEXbdaEBzK4hdhrk0YkA@mail.gmail.com>
Message-ID: <BANLkTimL8jAcRCzg8eDFnDwszyVpdZd8-w@mail.gmail.com>

There's a sample client included, you can check it out here:

http://myprojectguide.org/p/siprtmp/

You can use that to test it for connecting to your rtpm/sip gateway and sip
server. The source code is included in a folder of the rtmplite project,
which is locate here:

http://code.google.com/p/rtmplite/


On Tue, Apr 26, 2011 at 9:13 PM, Henry Huang <b_ball_henry at hotmail.com>wrote:

> Do you have to develop a flash client from scratch or this project has a
> sample client to use?
>
> Henry
>
> On Wed, Apr 27, 2011 at 1:45 AM, Nicolas Brenner <nico at clickfono.com>wrote:
>
>> It's worked a lot better than red5 for me, using a lot less resources, but
>> that's my experience.
>>
>>
>> On Tue, Apr 26, 2011 at 1:38 PM, Madovsky <infos at madovsky.org> wrote:
>>
>>>  sirtmp is not really stable and use a lot of CPU
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 nico at clickfono.com  Wed Apr 27 13:58:18 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Wed, 27 Apr 2011 05:58:18 -0400
Subject: [Freeswitch-users] Python vs LUA
In-Reply-To: <BANLkTinHmV=GTEQKoGuzvmcjBxM02Nd4xg@mail.gmail.com>
References: <BANLkTinHmV=GTEQKoGuzvmcjBxM02Nd4xg@mail.gmail.com>
Message-ID: <BANLkTinEdZgBSCxFz4XiWXoy-jLBiJ9uNg@mail.gmail.com>

After having used Javascript and Lua for scripting in FreeSWITCH, I would
recommend using ESL for anything serious. There are many advantages to using
ESL instead of a script inside FreeSWITCH, but the main one for me was
easier call control, especially when doing asynchronous stuff. I believe
there's a Python version of the ESL library, so you shouldn't have much
trouble getting started.


On Wed, Apr 27, 2011 at 2:35 AM, Anton VG <anton.vazir at gmail.com> wrote:

> Hello!
> Cannot get answer for myself for languages processing overhead/preference,
>
> I personally would like to use Python for scripting under Freeswitch,
> but considering some recent list activity about low CPS while using
> python scripts, etc
> querstioning, maybe it's better to use LUA instead? WIKI or list does
> not answer this questions in satisfying manner unfortunatelly. For DB
> intensive scripts - DB connections pool is a very important stuff,
> having direct influence performance and CPS specifically.
>
> It looks that lua might be lighter and more tightly integrated to FS,
> but if processing overhead difference is just a few times python is
> still preferred for me (and many others) since much more
> powerfull/libraryreach/etc.
> I'ts would be just nice to clear this question.
>
> 1. Is there any comparisions on plain processing overhead?
> 2. Any data on memory consumtption per active python process?
> 2. Is there a freeswitch DB pool connection interface like for LUA in
> mod_python?
> 3. If not - what would be the known technique to use external DB
> connection pool from inside the FS python script?
> 4. Seems logical, but please confirm/reject - once executed (compiled
> to the bytecode) - the second time the python script under FS will be
> executed as precompiled bytecode?
>
> Regards,
> Anton.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 nico at clickfono.com  Wed Apr 27 14:10:20 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Wed, 27 Apr 2011 06:10:20 -0400
Subject: [Freeswitch-users] Outbound caller id for a specific gateway?
In-Reply-To: <BANLkTikYBgua9M3ya8wvJ8UjFMU5EEjQHg@mail.gmail.com>
References: <AANLkTin=M3MJKB5QA86iL+hkhkZncTQ8eiKQJM2t6ap5@mail.gmail.com>
	<AANLkTimqg7TRBULWR2__kxu32K4QuHQi1ChgsAZGiW+G@mail.gmail.com>
	<AANLkTimnorNHZa_3KqeqgecKjgHo4PLQWsm8+xH_FJMQ@mail.gmail.com>
	<BANLkTikYBgua9M3ya8wvJ8UjFMU5EEjQHg@mail.gmail.com>
Message-ID: <BANLkTimKmC1ErpTJ20ofmt64v7CNw_xGeA@mail.gmail.com>

Wow, that's really helpful, I'm going to try it for setting specific codecs
(absolute_codec_string).


On Wed, Apr 20, 2011 at 4:14 AM, Steven Ayre <steveayre at gmail.com> wrote:

> How about you set the channel variable in the gateway definition for the
> outbound direction:
> http://wiki.freeswitch.org/wiki/Sofia.conf.xml#Variables
>
> That way it will be automatically set on all calls going through that
> gateway without you needing to change your dialplan.
>
> -Steve
>
>
>
>
>
> On 22 March 2011 07:49, Dmitry Bely <dmitry.bely at gmail.com> wrote:
>
>> On Tue, Mar 22, 2011 at 8:02 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > How about set the caller id in vars.xml:
>> > <X-PRE-PROCESS cmd="set" data="outbound_caller_id=0000000000"/>
>> > Set it to the most common value and then you only have to do something
>> in
>> > bridges and originates that need a CID different from what you set in
>> > vars.xml...
>>
>> I have a number of gateways that require different CIDs. So there is
>> no "most common" value. I have managed to make a proper dialplan but
>> originate is still tedious... I just wonder if there is more clean way
>> (something like caller-id-in-from gateway parameter)
>>
>> > -MC
>> >
>> > On Sat, Mar 19, 2011 at 4:27 AM, Dmitry Bely <dmitry.bely at gmail.com>
>> wrote:
>> >>
>> >> My VoIP provider requires a specific caller ID set for an outbound
>> >> call otherwise the call is rejected. Currently I set it just before
>> >> bridge
>> >>
>> >>      <action application="set" data="effective_caller_id_name=..."/>
>> >>      <action application="bridge" data="..."/>
>> >>
>> >> But it's tedious as there is a number of bridge commands in the
>> >> dialplan and I still have to explicitly specify the caller id for
>> >> "originate" command in the FreeSWITCH console. Is it possible to force
>> >> an outbound caller id on a gateway basis?
>>
>> - Dmitry Bely
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Apr 27 14:16:01 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Wed, 27 Apr 2011 12:16:01 +0200
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
In-Reply-To: <BANLkTi=aKo+E1Dj1gC2Q+OZLeegr6V0-rQ@mail.gmail.com>
References: <4DAECCA8.1050203@gmx.net>	<BANLkTikH9QMuhvKXqfBu78fOzqxoME3X8A@mail.gmail.com>	<4DB328AF.8090107@telefaks.de>	<BANLkTikz_PSaE6T99jnrJSgEFJSWBpyxvw@mail.gmail.com>	<4DB3E075.1050202@telefaks.de>	<BANLkTi=_nMDp_kjCrDmUkfLM-tMXt+nZ1g@mail.gmail.com>
	<BANLkTi=aKo+E1Dj1gC2Q+OZLeegr6V0-rQ@mail.gmail.com>
Message-ID: <4DB7ECE1.2060603@telefaks.de>

Hello Guru,

I upgraded to latest GIT, but behaviour is still the same.

Best regards
Peter

guru singh schrieb:
> Hi Peter,
>
> Please try latest git. Moc's commit has fixed the issue.
>
> Regards,
> guru
>
> On Sun, Apr 24, 2011 at 6:09 PM, guru singh <grsingh750 at gmail.com> wrote:
>   
>> Hi Peter,
>>
>> You're right. Please ignore my previous message, status 'Available (On
>> Demand)' should also be fine.
>> I've tried it and I see the same behavior as you. Reading the logs, I
>> can see that nothing after playback gets executed once the call is
>> hungup. It's not just transfer, any other application also is not
>> getting executing. Something is amiss.
>> Maybe moc or somebody else will point it out.
>>
>> Regards,
>> guru
>>
>> On Sun, Apr 24, 2011 at 2:03 PM, Peter Steinbach <lists at telefaks.de> wrote:
>>     
>>> Hello Guru,
>>>
>>> thanks for your hint, however this did not help.
>>> The point is that according to the dialplan the agent should be transferred
>>> to the same extension again an wait. In my case, when the call hanges up,
>>> there is no attempt to continue with the dialplan.
>>> So I expect it does not have to do with the agent's setings.
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>> guru singh schrieb:
>>>
>>> Hi Peter
>>>
>>> Try setting the status as 'Available' instead of 'Available (On Demand)'
>>> In case of 'Available (On Demand)' after the call ends, the agent's
>>> status is set to 'idle', so therefore no calls are given to the
>>> specific agent. I'm not too sure if this is the only change required
>>> to get the behavior you expect. I've not tried it on my box yet. I can
>>> only do it Monday and let you know.
>>>
>>> Regards,
>>> guru
>>>
>>> On Sun, Apr 24, 2011 at 12:59 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>>
>>>
>>> Thank you guru,
>>>
>>> I tried the example in the wiki. This worked.
>>> I wanted the agent also to wait again with MOH after the caller hung up.
>>> This did not work in my environment (Freeswitch git April 2011). The
>>> agent was always hungup after the caller hung up and was not transferred
>>> to the same dialplan extension again.
>>> Also hangup after bridge =false did not work.
>>>
>>> Does this work in your environment?
>>>
>>> Best regards
>>> Peter
>>>
>>>
>>> guru singh schrieb:
>>>
>>>
>>> Hi Peter,
>>>
>>> I've been using mod_callcenter for a while and must say it works really
>>> well.
>>> I just tried the uuid-standby strategy and basically it's exactly what
>>> you say the asterisk thing does.
>>> See the dialplan example.
>>> http://wiki.freeswitch.org/wiki/Mod_callcenter#uuid-standby
>>> Agent is dialing 4099 and listening to MOH. When a call arrives, it's
>>> bridged directly to the agent.
>>>
>>> Regards
>>> guru
>>>
>>> On Wed, Apr 20, 2011 at 5:38 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>>
>>>
>>>
>>> Hello,
>>>
>>> I am trying to use the mod_callcenter functionality. This works nicely
>>> so far so thank you to everybody involved for programming this nice module!
>>> But I am stuck somehow with uuid-standby. Can anybody explain how
>>> uuid-standby works?
>>>
>>> Another question: In the Asterisk based callcenter solution named
>>> "Vicidial", an agent can be held permanently in a conference, waiting
>>> for calls who are bridged to his uuid in the conference. Can this be
>>> haviour be done with mod_callcenter?
>>>
>>>
>>> Best regards
>>> Peter
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>>
>>> --
>>> With kind regards
>>> Peter Steinbach
>>>
>>> Telefaks Services GmbH
>>> mailto:lists (att) telefaks.de
>>> Internet: www.telefaks.de
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at 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
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>       
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 nico at clickfono.com  Wed Apr 27 14:17:16 2011
From: nico at clickfono.com (Nicolas Brenner)
Date: Wed, 27 Apr 2011 06:17:16 -0400
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
Message-ID: <BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>

If you know the rate and the balance for the call, you may calculate the
maximum time for the call as max_time = balance/rate. Then you may schedule
a hangup for that call max_time in the future, something like this:

sched_hangup +" + max_time + " "+ call_uuid +" 'ALLOTED_TIMEOUT'

That way the balance shouldn't go below 0, since you are hanging up the call
before it does, with hangup cause ALLOTED_TIMEOUT.



On Fri, Mar 4, 2011 at 2:44 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> Hello,
>
> I do the route selecting with a lua script. overflow and load distribution
> with mod_distributor loaded via curl.
>
> the prepaid side is done with nibble, yes. But i don't like it too much, i
> might just deduct the balance when the call disconnects and let the
> authorization block new calls, so the balance might go under 0 a little...
>
>
>
>
> On Sun, Feb 27, 2011 at 9:06 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:
>
>> press sent too quick..
>>
>> what did you use for routing? curl? esl?
>>
>> did you use nibble bill for prepaid app?
>>
>>
>> On Sun, Feb 27, 2011 at 9:05 PM, Saeed Ahmed <saeedahmad1981 at gmail.com>wrote:
>>
>>> Great!
>>>
>>> want to see it soon.
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 eagle.antonio at gmail.com  Wed Apr 27 14:25:01 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Wed, 27 Apr 2011 10:25:01 +0000
Subject: [Freeswitch-users] Python vs LUA
In-Reply-To: <BANLkTinEdZgBSCxFz4XiWXoy-jLBiJ9uNg@mail.gmail.com>
References: <BANLkTinHmV=GTEQKoGuzvmcjBxM02Nd4xg@mail.gmail.com>
	<BANLkTinEdZgBSCxFz4XiWXoy-jLBiJ9uNg@mail.gmail.com>
Message-ID: <BANLkTinCDuvy22R_W8WTV3Hu1VqRaBQgzg@mail.gmail.com>

Hello Anton

If you are designing simple stuff  with python , stick with it.
If you are using tons and tons of IVR's calls , Lua is more lightweight ,
don't forget that python is a highly heavy duty language.

If you process a large number of ivrs ( millions) ESL will be the best
solution you can find a server.py in your pythonesl folder , if you can't
find it drop me an e-mail.

Currently i'm developing a framework that will behind a load Balancer scale
the "Application" part of the ivr's.

A smal framework for your case could be :

server.py from pythonesl > It's A threaded Server Not MP , means every call
will be equal to 1 thread ! Be carefull :D
Sqlalchemy will support connection polling.

If you need further help just ask :)

A/T



2011/4/27 Nicolas Brenner <nico at clickfono.com>

> After having used Javascript and Lua for scripting in FreeSWITCH, I would
> recommend using ESL for anything serious. There are many advantages to using
> ESL instead of a script inside FreeSWITCH, but the main one for me was
> easier call control, especially when doing asynchronous stuff. I believe
> there's a Python version of the ESL library, so you shouldn't have much
> trouble getting started.
>
>
> On Wed, Apr 27, 2011 at 2:35 AM, Anton VG <anton.vazir at gmail.com> wrote:
>
>> Hello!
>> Cannot get answer for myself for languages processing overhead/preference,
>>
>> I personally would like to use Python for scripting under Freeswitch,
>> but considering some recent list activity about low CPS while using
>> python scripts, etc
>> querstioning, maybe it's better to use LUA instead? WIKI or list does
>> not answer this questions in satisfying manner unfortunatelly. For DB
>> intensive scripts - DB connections pool is a very important stuff,
>> having direct influence performance and CPS specifically.
>>
>> It looks that lua might be lighter and more tightly integrated to FS,
>> but if processing overhead difference is just a few times python is
>> still preferred for me (and many others) since much more
>> powerfull/libraryreach/etc.
>> I'ts would be just nice to clear this question.
>>
>> 1. Is there any comparisions on plain processing overhead?
>> 2. Any data on memory consumtption per active python process?
>> 2. Is there a freeswitch DB pool connection interface like for LUA in
>> mod_python?
>> 3. If not - what would be the known technique to use external DB
>> connection pool from inside the FS python script?
>> 4. Seems logical, but please confirm/reject - once executed (compiled
>> to the bytecode) - the second time the python script under FS will be
>> executed as precompiled bytecode?
>>
>> Regards,
>> Anton.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 eagle.antonio at gmail.com  Wed Apr 27 14:30:56 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Wed, 27 Apr 2011 10:30:56 +0000
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <4DB7CAB5.9080801@cupis.co.uk>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<4DB7CAB5.9080801@cupis.co.uk>
Message-ID: <BANLkTikpzWH450Kd12bLLgcbiWNFzYNeQw@mail.gmail.com>

Just a little update for you guys.

I have jumped from debian to centos did a full reinstall +  Raid 1 + DB in
Ram.

And was able to get 78 Simult. Calls at only 10 % CPU so there is a huge
difference so far from 120 % on debian :\.
Call quality has improved , local lan call quality has rised , G729 Licenses
working again thanks Brian West !!!.

I will now start to deploy our IVRs using ESL to see if i can offload some
of the load.

Regards
A/T


2011/4/27 Paul Cupis <paul at cupis.co.uk>

> On 27/04/11 01:24, Ariel Monaco wrote:
> > We had high CPU utilization peaks in the past, which lead to some
> > audio issues (clipping). We were using debian at that time, which
> > was a customer-side requirement.
> >
> > I'm not a kernel guru but I remember this had something to do with
> > kernel timer cycles and the issue was address by adding "divider=10"
> > or "divider=100" as a kernel's boot loader option.
>
> Interesting. This kernel option is designed to reduce the valy of
> CONFIG_HZ at run=time, so if you have a 1000Hz kernel, setting
> divider=10 at boot time will effectively give you a 100Hz kernel.
>
> This seems to fly in the face of other recommendations on this kernel
> setting!
>
> Debian default kernel sets CONFIG_HZ=100
> CentOS default kernel sets CONFIG_HZ=1000
>
> 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 steveayre at gmail.com  Wed Apr 27 14:30:17 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 27 Apr 2011 11:30:17 +0100
Subject: [Freeswitch-users] Complete wholesale app in freeswitch
In-Reply-To: <BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
References: <AANLkTinxHb+JG1GSx=H36RC749sV+k7nQrrdECw81d0P@mail.gmail.com>
	<AANLkTikmutDcTQQnXpmST6NERx70t+Tzc-7w3jkTGX5L@mail.gmail.com>
	<AANLkTimeRTUixHHNR0CL19iyf0je+_cJGbwz9_BSdW2K@mail.gmail.com>
	<AANLkTimur7MT5aiFU1q=d0AB5T9ba4yjUVNm8VOifKE-@mail.gmail.com>
	<AANLkTimCB+Bi_EOP1n-cqHe1+q5Tt=z0Ztwict9R9+Oc@mail.gmail.com>
	<BANLkTi=a=c4RK=b2ho8=xAdtNPWT9NtfDQ@mail.gmail.com>
Message-ID: <B7C56260-4065-4B4B-A6C3-159FD12655B4@gmail.com>

You are open to losing money though if they can make more than one call at once - each would be scheduled as if they were the only call, allowing the callers to drop below 0.

Steve on iPhone

On 27 Apr 2011, at 11:17, Nicolas Brenner <nico at clickfono.com> wrote:

> If you know the rate and the balance for the call, you may calculate the maximum time for the call as max_time = balance/rate. Then you may schedule a hangup for that call max_time in the future, something like this:
> 
> sched_hangup +" + max_time + " "+ call_uuid +" 'ALLOTED_TIMEOUT'
> 
> That way the balance shouldn't go below 0, since you are hanging up the call before it does, with hangup cause ALLOTED_TIMEOUT.
> 
> 
> 
> On Fri, Mar 4, 2011 at 2:44 PM, David Villasmil <david.villasmil.work at gmail.com> wrote:
> Hello,
> 
> I do the route selecting with a lua script. overflow and load distribution with mod_distributor loaded via curl.
> 
> the prepaid side is done with nibble, yes. But i don't like it too much, i might just deduct the balance when the call disconnects and let the authorization block new calls, so the balance might go under 0 a little... 
> 
> 
> 
> 
> On Sun, Feb 27, 2011 at 9:06 PM, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> press sent too quick..
> 
> what did you use for routing? curl? esl?
> 
> did you use nibble bill for prepaid app?
> 
> 
> On Sun, Feb 27, 2011 at 9:05 PM, Saeed Ahmed <saeedahmad1981 at gmail.com> wrote:
> Great!
> 
> want to see it soon.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 avi at avimarcus.net  Wed Apr 27 14:35:44 2011
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 27 Apr 2011 13:35:44 +0300
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTikpzWH450Kd12bLLgcbiWNFzYNeQw@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<4DB7CAB5.9080801@cupis.co.uk>
	<BANLkTikpzWH450Kd12bLLgcbiWNFzYNeQw@mail.gmail.com>
Message-ID: <BANLkTimDXy68p5f7FpWDHryT7iJcLD0bcA@mail.gmail.com>

Did you have the Raid 1 and DB in ram on debian, too..?
-Avi

On Wed, Apr 27, 2011 at 1:30 PM, Antonio Teixeira
<eagle.antonio at gmail.com>wrote:

> Just a little update for you guys.
>
> I have jumped from debian to centos did a full reinstall +  Raid 1 + DB in
> Ram.
>
> And was able to get 78 Simult. Calls at only 10 % CPU so there is a huge
> difference so far from 120 % on debian :\.
> Call quality has improved , local lan call quality has rised , G729
> Licenses working again thanks Brian West !!!.
>
> I will now start to deploy our IVRs using ESL to see if i can offload some
> of the load.
>
> Regards
> A/T
>
>
> 2011/4/27 Paul Cupis <paul at cupis.co.uk>
>
>> On 27/04/11 01:24, Ariel Monaco wrote:
>> > We had high CPU utilization peaks in the past, which lead to some
>> > audio issues (clipping). We were using debian at that time, which
>> > was a customer-side requirement.
>> >
>> > I'm not a kernel guru but I remember this had something to do with
>> > kernel timer cycles and the issue was address by adding "divider=10"
>> > or "divider=100" as a kernel's boot loader option.
>>
>> Interesting. This kernel option is designed to reduce the valy of
>> CONFIG_HZ at run=time, so if you have a 1000Hz kernel, setting
>> divider=10 at boot time will effectively give you a 100Hz kernel.
>>
>> This seems to fly in the face of other recommendations on this kernel
>> setting!
>>
>> Debian default kernel sets CONFIG_HZ=100
>> CentOS default kernel sets CONFIG_HZ=1000
>>
>> 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
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 eagle.antonio at gmail.com  Wed Apr 27 14:38:39 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Wed, 27 Apr 2011 10:38:39 +0000
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTimDXy68p5f7FpWDHryT7iJcLD0bcA@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<4DB7CAB5.9080801@cupis.co.uk>
	<BANLkTikpzWH450Kd12bLLgcbiWNFzYNeQw@mail.gmail.com>
	<BANLkTimDXy68p5f7FpWDHryT7iJcLD0bcA@mail.gmail.com>
Message-ID: <BANLkTimHv+BHfarvo4Mc_3W=4u1PYzL1oA@mail.gmail.com>

Raid 1 yes , but no DB in ram


2011/4/27 Avi Marcus <avi at avimarcus.net>

> Did you have the Raid 1 and DB in ram on debian, too..?
> -Avi
>
>
> On Wed, Apr 27, 2011 at 1:30 PM, Antonio Teixeira <eagle.antonio at gmail.com
> > wrote:
>
>> Just a little update for you guys.
>>
>> I have jumped from debian to centos did a full reinstall +  Raid 1 + DB in
>> Ram.
>>
>> And was able to get 78 Simult. Calls at only 10 % CPU so there is a huge
>> difference so far from 120 % on debian :\.
>> Call quality has improved , local lan call quality has rised , G729
>> Licenses working again thanks Brian West !!!.
>>
>> I will now start to deploy our IVRs using ESL to see if i can offload some
>> of the load.
>>
>> Regards
>> A/T
>>
>>
>> 2011/4/27 Paul Cupis <paul at cupis.co.uk>
>>
>>> On 27/04/11 01:24, Ariel Monaco wrote:
>>> > We had high CPU utilization peaks in the past, which lead to some
>>> > audio issues (clipping). We were using debian at that time, which
>>> > was a customer-side requirement.
>>> >
>>> > I'm not a kernel guru but I remember this had something to do with
>>> > kernel timer cycles and the issue was address by adding "divider=10"
>>> > or "divider=100" as a kernel's boot loader option.
>>>
>>> Interesting. This kernel option is designed to reduce the valy of
>>> CONFIG_HZ at run=time, so if you have a 1000Hz kernel, setting
>>> divider=10 at boot time will effectively give you a 100Hz kernel.
>>>
>>> This seems to fly in the face of other recommendations on this kernel
>>> setting!
>>>
>>> Debian default kernel sets CONFIG_HZ=100
>>> CentOS default kernel sets CONFIG_HZ=1000
>>>
>>> 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
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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  Wed Apr 27 14:45:30 2011
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde)
Date: Wed, 27 Apr 2011 12:45:30 +0200
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <BANLkTikpzWH450Kd12bLLgcbiWNFzYNeQw@mail.gmail.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<4DB7CAB5.9080801@cupis.co.uk>
	<BANLkTikpzWH450Kd12bLLgcbiWNFzYNeQw@mail.gmail.com>
Message-ID: <1303901130.3046.23.camel@luna.tc.commsmundi.com>

Did you use the default kernel in your debian setup?

Fran?ois.

On Wed, 2011-04-27 at 10:30 +0000, Antonio Teixeira wrote:
> Just a little update for you guys.
> 
> I have jumped from debian to centos did a full reinstall +  Raid 1 +
> DB in Ram.
> 
> And was able to get 78 Simult. Calls at only 10 % CPU so there is a
> huge difference so far from 120 % on debian :\.
> Call quality has improved , local lan call quality has rised , G729
> Licenses working again thanks Brian West !!!.
> 
> I will now start to deploy our IVRs using ESL to see if i can offload
> some of the load.
> 
> Regards
> A/T
> 
> 
> 2011/4/27 Paul Cupis <paul at cupis.co.uk>
>         On 27/04/11 01:24, Ariel Monaco wrote:
>         > We had high CPU utilization peaks in the past, which lead to
>         some
>         > audio issues (clipping). We were using debian at that time,
>         which
>         > was a customer-side requirement.
>         >
>         > I'm not a kernel guru but I remember this had something to
>         do with
>         > kernel timer cycles and the issue was address by adding
>         "divider=10"
>         > or "divider=100" as a kernel's boot loader option.
>         
>         
>         Interesting. This kernel option is designed to reduce the valy
>         of
>         CONFIG_HZ at run=time, so if you have a 1000Hz kernel, setting
>         divider=10 at boot time will effectively give you a 100Hz
>         kernel.
>         
>         This seems to fly in the face of other recommendations on this
>         kernel
>         setting!
>         
>         Debian default kernel sets CONFIG_HZ=100
>         CentOS default kernel sets CONFIG_HZ=1000
>         
>         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
>         
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From eagle.antonio at gmail.com  Wed Apr 27 14:49:32 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Wed, 27 Apr 2011 10:49:32 +0000
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <1303901130.3046.23.camel@luna.tc.commsmundi.com>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<4DB7CAB5.9080801@cupis.co.uk>
	<BANLkTikpzWH450Kd12bLLgcbiWNFzYNeQw@mail.gmail.com>
	<1303901130.3046.23.camel@luna.tc.commsmundi.com>
Message-ID: <BANLkTik5Edby9o94ygNRHJCeFne4j_2+EA@mail.gmail.com>

Yes a default installation of debian ( minimal/base ) also take into
consideration this install was before that wiki warning regarding the
divider of the kernel HZ thingy.


2011/4/27 Fran?ois Delawarde <fdelawarde at wirelessmundi.com>

> Did you use the default kernel in your debian setup?
>
> Fran?ois.
>
> On Wed, 2011-04-27 at 10:30 +0000, Antonio Teixeira wrote:
> > Just a little update for you guys.
> >
> > I have jumped from debian to centos did a full reinstall +  Raid 1 +
> > DB in Ram.
> >
> > And was able to get 78 Simult. Calls at only 10 % CPU so there is a
> > huge difference so far from 120 % on debian :\.
> > Call quality has improved , local lan call quality has rised , G729
> > Licenses working again thanks Brian West !!!.
> >
> > I will now start to deploy our IVRs using ESL to see if i can offload
> > some of the load.
> >
> > Regards
> > A/T
> >
> >
> > 2011/4/27 Paul Cupis <paul at cupis.co.uk>
> >         On 27/04/11 01:24, Ariel Monaco wrote:
> >         > We had high CPU utilization peaks in the past, which lead to
> >         some
> >         > audio issues (clipping). We were using debian at that time,
> >         which
> >         > was a customer-side requirement.
> >         >
> >         > I'm not a kernel guru but I remember this had something to
> >         do with
> >         > kernel timer cycles and the issue was address by adding
> >         "divider=10"
> >         > or "divider=100" as a kernel's boot loader option.
> >
> >
> >         Interesting. This kernel option is designed to reduce the valy
> >         of
> >         CONFIG_HZ at run=time, so if you have a 1000Hz kernel, setting
> >         divider=10 at boot time will effectively give you a 100Hz
> >         kernel.
> >
> >         This seems to fly in the face of other recommendations on this
> >         kernel
> >         setting!
> >
> >         Debian default kernel sets CONFIG_HZ=100
> >         CentOS default kernel sets CONFIG_HZ=1000
> >
> >         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
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing 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 grsingh750 at gmail.com  Wed Apr 27 15:43:59 2011
From: grsingh750 at gmail.com (guru singh)
Date: Wed, 27 Apr 2011 17:13:59 +0530
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
In-Reply-To: <4DB7ECE1.2060603@telefaks.de>
References: <4DAECCA8.1050203@gmx.net>
	<BANLkTikH9QMuhvKXqfBu78fOzqxoME3X8A@mail.gmail.com>
	<4DB328AF.8090107@telefaks.de>
	<BANLkTikz_PSaE6T99jnrJSgEFJSWBpyxvw@mail.gmail.com>
	<4DB3E075.1050202@telefaks.de>
	<BANLkTi=_nMDp_kjCrDmUkfLM-tMXt+nZ1g@mail.gmail.com>
	<BANLkTi=aKo+E1Dj1gC2Q+OZLeegr6V0-rQ@mail.gmail.com>
	<4DB7ECE1.2060603@telefaks.de>
Message-ID: <BANLkTi=ACjRN5xUW5HuqN36DJcg01RSFHw@mail.gmail.com>

Hi Peter,

Yes, there were some changes made again. Try using
transfer_after_bridge application.
http://wiki.freeswitch.org/wiki/Variable_transfer_after_bridge

Regards,
guru

On Wed, Apr 27, 2011 at 3:46 PM, Peter Steinbach <lists at telefaks.de> wrote:
> Hello Guru,
>
> I upgraded to latest GIT, but behaviour is still the same.
>
> Best regards
> Peter
>
> guru singh schrieb:
>
> Hi Peter,
>
> Please try latest git. Moc's commit has fixed the issue.
>
> Regards,
> guru
>
> On Sun, Apr 24, 2011 at 6:09 PM, guru singh <grsingh750 at gmail.com> wrote:
>
>
> Hi Peter,
>
> You're right. Please ignore my previous message, status 'Available (On
> Demand)' should also be fine.
> I've tried it and I see the same behavior as you. Reading the logs, I
> can see that nothing after playback gets executed once the call is
> hungup. It's not just transfer, any other application also is not
> getting executing. Something is amiss.
> Maybe moc or somebody else will point it out.
>
> Regards,
> guru
>
> On Sun, Apr 24, 2011 at 2:03 PM, Peter Steinbach <lists at telefaks.de> wrote:
>
>
> Hello Guru,
>
> thanks for your hint, however this did not help.
> The point is that according to the dialplan the agent should be transferred
> to the same extension again an wait. In my case, when the call hanges up,
> there is no attempt to continue with the dialplan.
> So I expect it does not have to do with the agent's setings.
>
> Best regards
> Peter
>
>
> guru singh schrieb:
>
> Hi Peter
>
> Try setting the status as 'Available' instead of 'Available (On Demand)'
> In case of 'Available (On Demand)' after the call ends, the agent's
> status is set to 'idle', so therefore no calls are given to the
> specific agent. I'm not too sure if this is the only change required
> to get the behavior you expect. I've not tried it on my box yet. I can
> only do it Monday and let you know.
>
> Regards,
> guru
>
> On Sun, Apr 24, 2011 at 12:59 AM, Peter Steinbach <lists at telefaks.de> wrote:
>
>
> Thank you guru,
>
> I tried the example in the wiki. This worked.
> I wanted the agent also to wait again with MOH after the caller hung up.
> This did not work in my environment (Freeswitch git April 2011). The
> agent was always hungup after the caller hung up and was not transferred
> to the same dialplan extension again.
> Also hangup after bridge =false did not work.
>
> Does this work in your environment?
>
> Best regards
> Peter
>
>
> guru singh schrieb:
>
>
> Hi Peter,
>
> I've been using mod_callcenter for a while and must say it works really
> well.
> I just tried the uuid-standby strategy and basically it's exactly what
> you say the asterisk thing does.
> See the dialplan example.
> http://wiki.freeswitch.org/wiki/Mod_callcenter#uuid-standby
> Agent is dialing 4099 and listening to MOH. When a call arrives, it's
> bridged directly to the agent.
>
> Regards
> guru
>
> On Wed, Apr 20, 2011 at 5:38 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>
>
>
> Hello,
>
> I am trying to use the mod_callcenter functionality. This works nicely
> so far so thank you to everybody involved for programming this nice module!
> But I am stuck somehow with uuid-standby. Can anybody explain how
> uuid-standby works?
>
> Another question: In the Asterisk based callcenter solution named
> "Vicidial", an agent can be held permanently in a conference, waiting
> for calls who are bridged to his uuid in the conference. Can this be
> haviour be done with mod_callcenter?
>
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbH
> mailto:lists (att) telefaks.de
> Internet: www.telefaks.de
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anton.vazir at gmail.com  Wed Apr 27 16:16:02 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 27 Apr 2011 17:16:02 +0500
Subject: [Freeswitch-users] How to filter Skypopen chat message in ESL
	mode?
In-Reply-To: <AANLkTimBd0msAdo0uGP7RfSek3MWd7g3vkQg_BPar5mL@mail.gmail.com>
References: <AANLkTi=T8G8LjKytzpSTw+bjGUZS7_3TAJDySxKevC0+@mail.gmail.com>
	<AANLkTi=HstdSAsVqe836AXpYwvTg8r3_UdovPCbrLUaO@mail.gmail.com>
	<AANLkTimW=fueqE-iMhPQ03+UEO5dj1-Mghdj7b-ZoM1-@mail.gmail.com>
	<AANLkTik5fMn4Yi_bU_KFKyufdJ1b13AOBD2-H47UHNWQ@mail.gmail.com>
	<AANLkTi=eE+AQ-F-xndR-6zBajomHY-a5A_8D1ULrZwM_@mail.gmail.com>
	<AANLkTinjo9k-UgOm-kF4dkYVn7XbXZz4MzS=g_RrmZmS@mail.gmail.com>
	<AANLkTik-NdpdJRYxBWZ1KYOOk8xxS59KWiJJmMeFERiP@mail.gmail.com>
	<AANLkTi=n4nbyjKcxGR4pM74XWLzC5VqR2nSJtuiN0WHa@mail.gmail.com>
	<AANLkTin8rCY8t6+buLn_N3RhBR768txZHPgrUOwCdacD@mail.gmail.com>
	<AANLkTin0EPm2LLUMCNHHD8oh+2XZk59PkO0Jxu85+Uki@mail.gmail.com>
	<AANLkTimUWBDyvN7-NhFR1JWEc3pBnQtnnfcHC8UEjVXd@mail.gmail.com>
	<AANLkTi=HGgC4BS6chKMWmU+dwnEFXLvgOq1yFB=pUC+q@mail.gmail.com>
	<AANLkTimBd0msAdo0uGP7RfSek3MWd7g3vkQg_BPar5mL@mail.gmail.com>
Message-ID: <BANLkTinVMOj0Ki-uPnzJMjBxnHH4CF=9vw@mail.gmail.com>

Giovanni, just curious, is the problem have been confirmed or not?

2010/12/2 Giovanni Maruzzelli <gmaruzz at celliax.org>:
> On Thu, Dec 2, 2010 at 3:40 PM, xuyan yang <xyangni at gmail.com> wrote:
>>>
>>> 1) Is that the complete skypopen.conf.xml? I mean, do you use one only
>>> interface?
>>
>> Yes, it is complete. During the test, I use only 1 client with latest
>> version running on ubuntu GUI. The older version can not be downloaded from
>> skype now.
>>>
>
> OK Xuyan, thanks for the infos.
>
> I'll test it asap (some hours).
>
> Maybe the problem comes from using the beta (2.1.81) version of the
> Skype client.
>
> The only Skype client version that is supported (because is the only
> one that works correctly) is the 2.0.72. I know that version is not on
> the Skype website (and I cannot give it to you, because Skype policies
> on distribution are probably not allowing me to distribute it), but
> maybe you can find someone that gives it to you.
>
> 2.1.81 is known to give a lot of problems of various nature (this one
> is very small), and will never be supported by mod_skypopen.
>
> Anyway, I'll test your problem asap with 2.0.72.
>
> -giovanni
>
>>>
>>>
>>> 2) An incoming skype call goes directly (because is in
>>> skypopen.conf.xml) to 5655 and ear the continuous message.
>>
>> yes, go directly to 5655.
>>
>>>
>>> 3) While the incoming skype call is hearing the message, the same
>>> incoming skypeclient sends chatmessages to the skypopen skypeclient
>>
>> yes, you are right
>>
>>>
>>> 4) You connect via telnet to the ESL port, do the "events plain message"
>>
>> not exactly, but I think fs_cli should function the same as telnet.
>>>
>>>
>>>
>>> 5) You get sometimes those longer events intermixed with the regular
>>> message events, but not everytime
>>> yes, you are right.
>>
>>
>>>
>>> Is this correct? If this is correct I will test asap and fix the
>>> possible problem.
>>>
>>> Btw, I tested yesterday with skypeclient not in a call and I got no
>>> problems at all. I don't think to be in a call would make any
>>> difference.
>>
>> ?It is not the call, but the?continuous sound playback in the call??which
>> cause the problem. If the call is parked idle, there is no problem.
>>>
>>>
>>>
>>> Please be certain you gave me all the info needed to exactly replicate
>>> your problem.
>>>
>>> -giovanni
>>>
>>> On Thu, Dec 2, 2010 at 12:09 PM, xuyan yang <xyangni at gmail.com> wrote:
>>> > OK. I will try to make it clear.
>>> > In general, my test case is to forward skype call to a extention such as
>>> > 5655. And then in the dialplan for 5655,?continuously?play some sound.
>>> > During playing, you can send skype chat messages from the caller and
>>> > check
>>> > what esl events has been generated from it. According to my test, the
>>> > raw
>>> > type event is not generated for most of chat messages.
>>> > In details,
>>> > this is my skypeopen.conf.xml:
>>> > <configuration name="skypopen.conf" description="Skypopen
>>> > Configuration">
>>> > ??<global_settings>
>>> > ?? ?<param name="debug" value="8"/>
>>> > ?? ?<param name="dialplan" value="XML"/>
>>> > ?? ?<param name="context" value="default"/>
>>> > ?? ?<param name="destination" value="1000"/>
>>> > ?? ?<param name="skype_user" value="user1"/>
>>> > ?? ?<param name="report_incoming_chatmessages" value="true"/>
>>> > ?? ?<param name="silent_mode" value="false"/>
>>> > ?? ?<param name="write_silence_when_idle" value="true"/>
>>> > ??</global_settings>
>>> > ??<!-- one entry here per each skypopen interface -->
>>> > ??<per_interface_settings>
>>> > ?? ?<interface id="1" name="interface1">
>>> > ?? ? ? ?<param name="X11-display" value=":0.0"/>
>>> > ?? ? ? ?<param name="skype_user" value="xxxxxxxxx"/>
>>> > ?? ? ? ?<param name="destination" value="5655"/>
>>> > ?? ?</interface>
>>> > ??</per_interface_settings>
>>> > </configuration>
>>> >
>>> >
>>> > in dialplan for default context, add the following to describe extention
>>> > 5655:
>>> > <extension name="skype" ?break="on-false">
>>> > <condition field="destination_number" expression="^5655$">
>>> > ?? ? ? ?<action application="answer"/>
>>> > ?? ? ? ?<action application="sleep" data="1000"/>
>>> > <action application="playback" data="ivr/ivr-sample_submenu.wav"/>
>>> > <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/> <action
>>> > application="playback" data="ivr/ivr-sample_submenu.wav"/>
>>> > ?? </condition>
>>> > </extension>
>>> > ivr-sample_submenu.wav is a sound file in en/us/callie/ivr/8000, but it
>>> > can
>>> > be replaced by any other wav files.
>>> > after setup, make a call to your skype user xxxxxxxxx.
>>> > in the mean time use fs_cli to connect FS and execute /event plain
>>> > MESSAGE
>>> > to listen esl events.
>>> > sent any chat messages from caller, then you are very likely to find
>>> > only
>>> > the short version events.
>>> >
>>> >
>>> > On Thu, Dec 2, 2010 at 8:50 AM, Giovanni Maruzzelli
>>> > <gmaruzz at celliax.org>
>>> > wrote:
>>> >>
>>> >> Xuyan,
>>> >>
>>> >> please.
>>> >>
>>> >> Can you give a clear defined procedure to reproduce the problem?
>>> >>
>>> >> Please, take the time to write it, complete of all relevant info
>>> >> (dialplan, script, etc etc) so I can reproduce it from a freshly
>>> >> installed FreeSWITCH.
>>> >>
>>> >> If you don't give this information I cannot fix the problem. And I
>>> >> cannot neither read your mind, nor finding the time to try every
>>> >> possible combination.
>>> >>
>>> >> Please, post something someone can cut and paste and a clear step by
>>> >> step procedure to replicate.
>>> >>
>>> >> -giovanni
>>> >>
>>> >> On Thu, Dec 2, 2010 at 4:45 AM, xuyan yang <xyangni at gmail.com> wrote:
>>> >> > I tried it again. When the line is parked and idle. the raw events
>>> >> > can
>>> >> > always be generated.
>>> >> > but if the line is kept busy such as the following case this events
>>> >> > is
>>> >> > trend
>>> >> > to have problem
>>> >> > <extension name="skype-in" ?break="on-false">
>>> >> > <condition field="destination_number" expression="^5655$">
>>> >> > ?? ? ? ?<action application="answer"/>
>>> >> > ?? ? ? ?<action application="sleep" data="1000"/>
>>> >> > <!--<action application="park"/> -->
>>> >> > <action application="playback" data="test.wav"/>
>>> >> > <action application="playback" data="test.wav"/>
>>> >> > <action application="playback" data="test.wav"/>
>>> >> > ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<action application="playback"
>>> >> > data="test.wav"/>
>>> >> > <action application="playback" data="test.wav"/>
>>> >> > <action application="playback" data="test.wav"/>
>>> >> > ?? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<action application="playback"
>>> >> > data="test.wav"/>
>>> >> > <action application="playback" data="test.wav"/>
>>> >> > <action application="playback" data="test.wav"/>
>>> >> > ?? </condition>
>>> >> > </extension>
>>> >> >
>>> >> >
>>> >> > On Thu, Dec 2, 2010 at 2:01 AM, Giovanni Maruzzelli
>>> >> > <gmaruzz at celliax.org>
>>> >> > wrote:
>>> >> >>
>>> >> >> I repeat, if you subscribe to "message" events you get only those.
>>> >> >> Also, how I can replicate your problem?
>>> >> >> I've tested events and I had no problem at all with spurious or
>>> >> >> unreliable events in mod-skypopen.
>>> >> >> Please, can you indicate a detailed way to reproduce your problem?
>>> >> >> -giovanni
>>> >> >>
>>> >> >> On 12/2/10, xuyan yang <xyangni at gmail.com> wrote:
>>> >> >> > I got it. Then the problem should be the unstable behavior of raw
>>> >> >> > events
>>> >> >> > which has about 25% chances of being missed.
>>> >> >> >
>>> >> >> > Fortunately, I have found a way to avoid this issue. Ignore all
>>> >> >> > message
>>> >> >> > events which contains Unique-ID field.
>>> >> >> >
>>> >> >> > On Wed, Dec 1, 2010 at 11:18 PM, Giovanni Maruzzelli
>>> >> >> > <gmaruzz at celliax.org>wrote:
>>> >> >> >
>>> >> >> >> On Thu, Dec 2, 2010 at 12:15 AM, Giovanni Maruzzelli
>>> >> >> >> <gmaruzz at celliax.org> wrote:
>>> >> >> >> > If you subscribe only to "MESSAGE" kind of events, you'll get
>>> >> >> >> > only
>>> >> >> >> > those.
>>> >> >> >> >
>>> >> >> >> > The other events are "raw" events, that other users have
>>> >> >> >> > requested
>>> >> >> >> > for
>>> >> >> >> > other purposes.
>>> >> >> >>
>>> >> >> >> or at ?least that is the expected behavior, please let me know if
>>> >> >> >> I
>>> >> >> >> introduced some regression in integrating that "raw event"
>>> >> >> >> thingy.
>>> >> >> >>
>>> >> >> >> -giovanni
>>> >> >> >>
>>> >> >> >> >
>>> >> >> >> > -giovanni
>>> >> >> >> >
>>> >> >> >> > On Wed, Dec 1, 2010 at 11:35 PM, xuyan yang <xyangni at gmail.com>
>>> >> >> >> > wrote:
>>> >> >> >> >> Thanks for your reply. I have read this page before. but the
>>> >> >> >> >> random
>>> >> >> >> emerging
>>> >> >> >> >> of the second verbose esl events is causing trouble. And I
>>> >> >> >> >> guess
>>> >> >> >> >> it
>>> >> >> >> >> may
>>> >> >> >> be a
>>> >> >> >> >> bug, or it should be predictable.
>>> >> >> >> >> Before reporting to jira, I just want to check whether I have
>>> >> >> >> >> made
>>> >> >> >> >> any
>>> >> >> >> >> mistake.
>>> >> >> >> >>
>>> >> >> >> >>
>>> >> >> >> >> On Wed, Dec 1, 2010 at 9:35 PM, Giovanni Maruzzelli
>>> >> >> >> >> <gmaruzz at gmail.com>
>>> >> >> >> >> wrote:
>>> >> >> >> >>>
>>> >> >> >> >>>
>>> >> >> >> >>>
>>> >> >> >>
>>> >> >> >>
>>> >> >> >>
>>> >> >> >> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#MESSAGE_.28Chat.29
>>> >> >> >> >>>
>>> >> >> >> >>> On Wed, Dec 1, 2010 at 10:23 PM, xuyan yang
>>> >> >> >> >>> <xyangni at gmail.com>
>>> >> >> >> >>> wrote:
>>> >> >> >> >>> > hi,
>>> >> >> >> >>> > I am writing ESL program on FreeSWITCH Version 1.0.head
>>> >> >> >> >>> > (git-8825b6e
>>> >> >> >> >>> > 2010-11-28 17-15-39 -0500)
>>> >> >> >> >>> > I need to handle skype chat message with a inbound ESL
>>> >> >> >> >>> > connection.
>>> >> >> >> But
>>> >> >> >> >>> > there
>>> >> >> >> >>> > are 1 or 2 esl events received randomly for each message.
>>> >> >> >> >>> > the first short one is alway generated, but the second one
>>> >> >> >> >>> > is
>>> >> >> >> >>> > random.
>>> >> >> >> So
>>> >> >> >> >>> > when trying to filter according to UUID, nothing is caught
>>> >> >> >> >>> > in
>>> >> >> >> >>> > many
>>> >> >> >> >>> > cases.
>>> >> >> >> >>> > If I make filter based on skype ID, duplicated messages are
>>> >> >> >> >>> > often
>>> >> >> >> heard.
>>> >> >> >> >>> > I do not why the behavior of the second event is random.
>>> >> >> >> >>> > How should I setup the filter to get 1 and only 1 event for
>>> >> >> >> >>> > each
>>> >> >> >> >>> > chat
>>> >> >> >> >>> > message? Thanks.
>>> >> >> >> >>> >
>>> >> >> >> >>> > the first is a short one with the following header:
>>> >> >> >> >>> > ?[Event-Name] = [MESSAGE]
>>> >> >> >> >>> > ?[Event-Calling-Function] = [incoming_chatmessage]
>>> >> >> >> >>> > ?[Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
>>> >> >> >> >>> > ?[Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
>>> >> >> >> >>> > ?[Content-Length] = [1]
>>> >> >> >> >>> > ?[subject] = [SIMPLE MESSAGE]
>>> >> >> >> >>> > ?[FreeSWITCH-IPv4] = [192.168.0.3]
>>> >> >> >> >>> > ?[hint] = [niqizhi]
>>> >> >> >> >>> > ?[from] = [niqizhi]
>>> >> >> >> >>> > ?[Event-Date-Local] = [2010-12-01 21:02:37]
>>> >> >> >> >>> > ?[proto] = [skype]
>>> >> >> >> >>> > ?[FreeSWITCH-IPv6] = [::1]
>>> >> >> >> >>> > ?[id] = [5334]
>>> >> >> >> >>> > ?[Event-Calling-File] = [mod_skypopen.c]
>>> >> >> >> >>> > ?[Event-Date-Timestamp] = [1291237357051788]
>>> >> >> >> >>> > ?[FreeSWITCH-Hostname] = [EYSRV]
>>> >> >> >> >>> > ?[login] = [interface1]
>>> >> >> >> >>> > ?[during-call] = [true]
>>> >> >> >> >>> > ?[Event-Calling-Line-Number] = [2915]
>>> >> >> >> >>> > ?[chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
>>> >> >> >> >>> > the second is much longer:
>>> >> >> >> >>> > ?[Caller-Source] = [mod_skypopen]
>>> >> >> >> >>> > ?[Event-Calling-Function] = [incoming_chatmessage]
>>> >> >> >> >>> > ?[Event-Date-GMT] = [Wed, 01 Dec 2010 21:02:37 GMT]
>>> >> >> >> >>> > ?[Answer-State] = [answered]
>>> >> >> >> >>> > ?[FreeSWITCH-IPv4] = [192.168.0.3]
>>> >> >> >> >>> > ?[Channel-State] = [CS_EXECUTE]
>>> >> >> >> >>> > ?[Channel-Read-Codec-Bit-Rate] = [256000]
>>> >> >> >> >>> > ?[FreeSWITCH-IPv6] = [::1]
>>> >> >> >> >>> > ?[Unique-ID] = [412764c0-fd8e-11df-9019-835ae03a7500]
>>> >> >> >> >>> > ?[Channel-Read-Codec-Rate] = [16000]
>>> >> >> >> >>> > ?[Caller-Destination-Number] = [5655]
>>> >> >> >> >>> > ?[Caller-Channel-Transfer-Time] = [0]
>>> >> >> >> >>> > ?[Channel-Call-State] = [ACTIVE]
>>> >> >> >> >>> > ?[Caller-Channel-Progress-Media-Time] = [0]
>>> >> >> >> >>> > ?[FreeSWITCH-Hostname] = [EYSRV]
>>> >> >> >> >>> > ?[Caller-Channel-Answered-Time] = [1291237326697085]
>>> >> >> >> >>> > ?[login] = [interface1]
>>> >> >> >> >>> > ?[during-call] = [true]
>>> >> >> >> >>> > ?[Channel-Name] = [skypopen/interface1]
>>> >> >> >> >>> > ?[Caller-Unique-ID] =
>>> >> >> >> >>> > [412764c0-fd8e-11df-9019-835ae03a7500]
>>> >> >> >> >>> > ?[Core-UUID] = [7d858a18-fcb8-11df-8f82-835ae03a7500]
>>> >> >> >> >>> > ?[Channel-Read-Codec-Name] = [L16]
>>> >> >> >> >>> > ?[Caller-Channel-Name] = [skypopen/interface1]
>>> >> >> >> >>> > ?[Caller-Caller-ID-Number] = [niqizhi]
>>> >> >> >> >>> > ?[Event-Date-Timestamp] = [1291237357051788]
>>> >> >> >> >>> > ?[Channel-State-Number] = [4]
>>> >> >> >> >>> > ?[Event-Calling-Line-Number] = [2888]
>>> >> >> >> >>> > ?[chatname] = [#niqizhi/$abcdericunion;631dd843d9b3eb1a]
>>> >> >> >> >>> > ?[Event-Name] = [MESSAGE]
>>> >> >> >> >>> > ?[Content-Length] = [1]
>>> >> >> >> >>> > ?[subject] = [SIMPLE MESSAGE]
>>> >> >> >> >>> > ?[Caller-Caller-ID-Name] = [niqizhi]
>>> >> >> >> >>> > ?[from] = [niqizhi]
>>> >> >> >> >>> > ?[Caller-Dialplan] = [XML]
>>> >> >> >> >>> > ?[Caller-Channel-Hangup-Time] = [0]
>>> >> >> >> >>> > ?[id] = [5334]
>>> >> >> >> >>> > ?[Caller-Profile-Index] = [1]
>>> >> >> >> >>> > ?[Caller-Direction] = [inbound]
>>> >> >> >> >>> > ?[Caller-Username] = [skypopen]
>>> >> >> >> >>> > ?[Channel-Write-Codec-Name] = [L16]
>>> >> >> >> >>> > ?[Call-Direction] = [inbound]
>>> >> >> >> >>> > ?[Caller-Screen-Bit] = [true]
>>> >> >> >> >>> > ?[hint] = [niqizhi]
>>> >> >> >> >>> > ?[Caller-Privacy-Hide-Number] = [false]
>>> >> >> >> >>> > ?[Event-Date-Local] = [2010-12-01 21:02:37]
>>> >> >> >> >>> > ?[proto] = [skype]
>>> >> >> >> >>> > ?[Caller-Channel-Created-Time] = [1291237326468855]
>>> >> >> >> >>> > ?[Event-Calling-File] = [mod_skypopen.c]
>>> >> >> >> >>> > ?[Caller-Channel-Progress-Time] = [0]
>>> >> >> >> >>> > ?[Caller-Privacy-Hide-Name] = [false]
>>> >> >> >> >>> > ?[Channel-Write-Codec-Rate] = [16000]
>>> >> >> >> >>> > ?[Caller-Context] = [default]
>>> >> >> >> >>> > ?[Channel-Write-Codec-Bit-Rate] = [256000]
>>> >> >> >> >>> > ?[Presence-Call-Direction] = [inbound]
>>> >> >> >> >>> > ?[Caller-Profile-Created-Time] = [1291237326468855]
>>> >> >> >> >>> > _______________________________________________
>>> >> >> >> >>> > FreeSWITCH-users mailing list
>>> >> >> >> >>> > FreeSWITCH-users at lists.freeswitch.org
>>> >> >> >> >>> >
>>> >> >> >> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >>> > UNSUBSCRIBE:
>>> >> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> >>> > http://www.freeswitch.org
>>> >> >> >> >>> >
>>> >> >> >> >>> >
>>> >> >> >> >>>
>>> >> >> >> >>>
>>> >> >> >> >>>
>>> >> >> >> >>> --
>>> >> >> >> >>> Sincerely,
>>> >> >> >> >>>
>>> >> >> >> >>> Giovanni Maruzzelli
>>> >> >> >> >>> Cell : +39-347-2665618
>>> >> >> >> >>>
>>> >> >> >> >>> _______________________________________________
>>> >> >> >> >>> FreeSWITCH-users mailing list
>>> >> >> >> >>> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >>> UNSUBSCRIBE:
>>> >> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> >>> http://www.freeswitch.org
>>> >> >> >> >>
>>> >> >> >> >>
>>> >> >> >> >> _______________________________________________
>>> >> >> >> >> FreeSWITCH-users mailing list
>>> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >> >> UNSUBSCRIBE:
>>> >> >> >> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> >> >> http://www.freeswitch.org
>>> >> >> >> >>
>>> >> >> >> >>
>>> >> >> >> >
>>> >> >> >> >
>>> >> >> >> >
>>> >> >> >> > --
>>> >> >> >> > Sincerely,
>>> >> >> >> >
>>> >> >> >> > Giovanni Maruzzelli
>>> >> >> >> > Cell : +39-347-2665618
>>> >> >> >> >
>>> >> >> >>
>>> >> >> >>
>>> >> >> >>
>>> >> >> >> --
>>> >> >> >> 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
>>> >> >> >>
>>> >> >> >
>>> >> >>
>>> >> >> --
>>> >> >> 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
>>> >> >
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >
>>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >>
>>> >>
>>> >>
>>> >> --
>>> >> Sincerely,
>>> >>
>>> >> Giovanni Maruzzelli
>>> >> Cell : +39-347-2665618
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From nboric at yx.cl  Wed Apr 27 16:19:00 2011
From: nboric at yx.cl (Neven Boric)
Date: Wed, 27 Apr 2011 09:19:00 -0300
Subject: [Freeswitch-users] mod_crd_sqlite entry limit and sqlite
	segfaults on triggers
In-Reply-To: <4DB5E0F1.4050602@yx.cl>
References: <4DB5E0F1.4050602@yx.cl>
Message-ID: <4DB809B4.5060601@yx.cl>

It seems I have fixed the issue. I did a backtrace with gdb and saw that 
the crash was when sqlite3ExprListDup called strdup passing a NULL 
pointer. I asked on the SQLIte mailing list and they told me that SQLite 
doesn't use strdup. So, I downloaded the original 3.3.13 SQLite release 
and compared it to the Freeswitch code. I found out that calls to 
sqlite3StrDup had been replaced with calls to strdup (see sqliteInt.h). 
Problem is sqlite3StrDup checks for NULL input, whereas strdup does not. 
I put the following instead:

#define sqliteStrDup(x)          (x?strdup(x):NULL)//sqlite3StrDup(x)

And now I can create triggers without crashes. Seems to me that as 
nobody uses triggers inside Freeswitch, this issue has gone unnoticed so 
far.

I will continue testing to discard any side effects.

Neven Boric escribi?:
> Hi,
>
> By looking at the code, I believe there is no limit on the number of 
> rows mod_cdr_sqlite will add to the cdr table. This could lead to 
> (eventually) eating all RAM if you DB is in tmpfs or eating all disk 
> space if it is stored on disk. Is this correct or am I missing something?
>
> To correct this I tried to add a trigger to the table along the lines of:
>
> CREATE TRIGGER IF NOT EXISTS CDR_LIMIT AFTER INSERT ON cdr
> BEGIN
>     DELETE FROM crd WHERE rowid <= (SELECT MAX(rowid) from cdr) - 100000;
> END
>
> and no matter how I try to add the trigger, I get a segmentation fault 
> when the code calls sqlite3_exec in switch_core_db_exec.
>
> Currently I'm blaiming SQLite, as there is a somewhat similar issue 
> involving triggers reported for versions older than 3.5.4 (FS is using 
> 3.3.13)
>
> http://sqlite.org/cvstrac/wiki?p=DatabaseCorruption
>
> Of course I could just execute the delete statement manually after 
> every insert, but it is uglier (more costly, non atomic).
>
> Has anyone any experience using triggers with the sqlite version 
> included in FS? Or has anyone tried other solution to limit the number 
> on entries on the cdr table?
>
> Thanks in advance
> Neven Boric
>
>


From anton.vazir at gmail.com  Wed Apr 27 16:18:33 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Wed, 27 Apr 2011 17:18:33 +0500
Subject: [Freeswitch-users] Python vs LUA
In-Reply-To: <BANLkTinCDuvy22R_W8WTV3Hu1VqRaBQgzg@mail.gmail.com>
References: <BANLkTinHmV=GTEQKoGuzvmcjBxM02Nd4xg@mail.gmail.com>
	<BANLkTinEdZgBSCxFz4XiWXoy-jLBiJ9uNg@mail.gmail.com>
	<BANLkTinCDuvy22R_W8WTV3Hu1VqRaBQgzg@mail.gmail.com>
Message-ID: <BANLkTimFKZ0AjjK9Z31pHq=gy1qGOstLEg@mail.gmail.com>

Thanks Antonio! Got the idea.


2011/4/27 Antonio Teixeira <eagle.antonio at gmail.com>:
> Hello Anton
>
> If you are designing simple stuff? with python , stick with it.
> If you are using tons and tons of IVR's calls , Lua is more lightweight ,
> don't forget that python is a highly heavy duty language.
>
> If you process a large number of ivrs ( millions) ESL will be the best
> solution you can find a server.py in your pythonesl folder , if you can't
> find it drop me an e-mail.
>
> Currently i'm developing a framework that will behind a load Balancer scale
> the "Application" part of the ivr's.
>
> A smal framework for your case could be :
>
> server.py from pythonesl > It's A threaded Server Not MP , means every call
> will be equal to 1 thread ! Be carefull :D
> Sqlalchemy will support connection polling.
>
> If you need further help just ask :)
>
> A/T
>
>
>
> 2011/4/27 Nicolas Brenner <nico at clickfono.com>
>>
>> After having used Javascript and Lua for scripting in FreeSWITCH, I would
>> recommend using ESL for anything serious. There are many advantages to using
>> ESL instead of a script inside FreeSWITCH, but the main one for me was
>> easier call control, especially when doing asynchronous stuff. I believe
>> there's a Python version of the ESL library, so you shouldn't have much
>> trouble getting started.
>>
>> On Wed, Apr 27, 2011 at 2:35 AM, Anton VG <anton.vazir at gmail.com> wrote:
>>>
>>> Hello!
>>> Cannot get answer for myself for languages processing
>>> overhead/preference,
>>>
>>> I personally would like to use Python for scripting under Freeswitch,
>>> but considering some recent list activity about low CPS while using
>>> python scripts, etc
>>> querstioning, maybe it's better to use LUA instead? WIKI or list does
>>> not answer this questions in satisfying manner unfortunatelly. For DB
>>> intensive scripts - DB connections pool is a very important stuff,
>>> having direct influence performance and CPS specifically.
>>>
>>> It looks that lua might be lighter and more tightly integrated to FS,
>>> but if processing overhead difference is just a few times python is
>>> still preferred for me (and many others) since much more
>>> powerfull/libraryreach/etc.
>>> I'ts would be just nice to clear this question.
>>>
>>> 1. Is there any comparisions on plain processing overhead?
>>> 2. Any data on memory consumtption per active python process?
>>> 2. Is there a freeswitch DB pool connection interface like for LUA in
>>> mod_python?
>>> 3. If not - what would be the known technique to use external DB
>>> connection pool from inside the FS python script?
>>> 4. Seems logical, but please confirm/reject - once executed (compiled
>>> to the bytecode) - the second time the python script under FS will be
>>> executed as precompiled bytecode?
>>>
>>> Regards,
>>> Anton.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From Tim.Meade at millicorp.com  Wed Apr 27 16:43:01 2011
From: Tim.Meade at millicorp.com (Tim Meade)
Date: Wed, 27 Apr 2011 12:43:01 +0000
Subject: [Freeswitch-users] bypass_media=true and socket
Message-ID: <6B85FABC3A341E4CA5D5D96AD9E4FE581745B9FE@Mailbox.millicorp.com>

We are somewhat new to freeswitch, but have found it to be an excellent solution to some projects.

With bypass_media=false before our socket command and the perl app returning the bridge everything works fine. Calls sound great.

But if we set bypass_media=true immediately before our socket command we receive an almost immediate CANCEL back from ?????

If we just hard code the bridge instead of the socket, and bypass_media=true then everything works fine.

It's just the combination of bypass_media=false and socket.

Here is the dialplan....

<include>
                <extension name="default">
                                <condition field="destination_number" expression="^\d\d\d\d\d\d\d+$">
                                                <action application="set" data="bypass_media=true"/>
                <action application="socket" data="0.0.0.0.0:8040 full"/>
</condition>
                </extension>
</include>

I have also tried:  <action application="set" data="bypass_media_after_bridge=true"/>

And the behavior is the same.

Is there something simple we may be missing?

Much thanks

Tim


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From zetruger at gmail.com  Wed Apr 27 17:32:31 2011
From: zetruger at gmail.com (=?KOI8-R?B?6dfBziD+ydPU0cvP1w==?=)
Date: Wed, 27 Apr 2011 17:32:31 +0400
Subject: [Freeswitch-users] FreeSWITCH with remote Music on Hold server. How
	to do it?
Message-ID: <BANLkTikCefhDoiDrx5=SaYi=OX73KxL3JA@mail.gmail.com>

FreeSWITCH with remote Music on Hold server. How to do it?


From eagle.antonio at gmail.com  Wed Apr 27 17:36:01 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Wed, 27 Apr 2011 13:36:01 +0000
Subject: [Freeswitch-users] FreeSWITCH with remote Music on Hold server.
 How to do it?
In-Reply-To: <BANLkTikCefhDoiDrx5=SaYi=OX73KxL3JA@mail.gmail.com>
References: <BANLkTikCefhDoiDrx5=SaYi=OX73KxL3JA@mail.gmail.com>
Message-ID: <BANLkTiniO40i6Q3LzEVd9+oup1MGQyCnYw@mail.gmail.com>

Checks The Docs / Wiki

http://wiki.freeswitch.org/wiki/Mod_shout


2011/4/27 ???? ???????? <zetruger at gmail.com>

> FreeSWITCH with remote Music on Hold server. How to do it?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From zetruger at gmail.com  Wed Apr 27 17:53:28 2011
From: zetruger at gmail.com (=?KOI8-R?B?6dfBziD+ydPU0cvP1w==?=)
Date: Wed, 27 Apr 2011 17:53:28 +0400
Subject: [Freeswitch-users] FreeSWITCH with remote Music on Hold server.
 How to do it?
In-Reply-To: <BANLkTiniO40i6Q3LzEVd9+oup1MGQyCnYw@mail.gmail.com>
References: <BANLkTikCefhDoiDrx5=SaYi=OX73KxL3JA@mail.gmail.com>
	<BANLkTiniO40i6Q3LzEVd9+oup1MGQyCnYw@mail.gmail.com>
Message-ID: <BANLkTik-1hnNbLBze9H2mHjaRHNRmWz9Mw@mail.gmail.com>

RTP data must be transmitted directly between Subscriber and MoH server.
Obviously, that mod_shout transmits data through FreeSWITCH for mp3 decoding.

,
2011/4/27 Antonio Teixeira <eagle.antonio at gmail.com>:
> Checks The Docs / Wiki
>
> http://wiki.freeswitch.org/wiki/Mod_shout
>
>
> 2011/4/27 ???? ???????? <zetruger at gmail.com>
>>
>> FreeSWITCH with remote Music on Hold server. How to do it?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From eagle.antonio at gmail.com  Wed Apr 27 18:02:22 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Wed, 27 Apr 2011 14:02:22 +0000
Subject: [Freeswitch-users] FreeSWITCH with remote Music on Hold server.
 How to do it?
In-Reply-To: <BANLkTik-1hnNbLBze9H2mHjaRHNRmWz9Mw@mail.gmail.com>
References: <BANLkTikCefhDoiDrx5=SaYi=OX73KxL3JA@mail.gmail.com>
	<BANLkTiniO40i6Q3LzEVd9+oup1MGQyCnYw@mail.gmail.com>
	<BANLkTik-1hnNbLBze9H2mHjaRHNRmWz9Mw@mail.gmail.com>
Message-ID: <BANLkTimB=Sa6hZKm6Xw2LVwrG1UdP+A5Yg@mail.gmail.com>

Well in that case you need to explain further what are you requirements so
the list can help you out.

Anyway so you want to bypass freeswitch in all the media ? or just the MOH ?
I'm assuming you don't need transcoding on the FS side.



2011/4/27 ???? ???????? <zetruger at gmail.com>

> RTP data must be transmitted directly between Subscriber and MoH server.
> Obviously, that mod_shout transmits data through FreeSWITCH for mp3
> decoding.
>
> ,
> 2011/4/27 Antonio Teixeira <eagle.antonio at gmail.com>:
> > Checks The Docs / Wiki
> >
> > http://wiki.freeswitch.org/wiki/Mod_shout
> >
> >
> > 2011/4/27 ???? ???????? <zetruger at gmail.com>
> >>
> >> FreeSWITCH with remote Music on Hold server. How to do it?
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Wed Apr 27 18:36:07 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 27 Apr 2011 07:36:07 -0700 (PDT)
Subject: [Freeswitch-users] attended transfer to gateway
In-Reply-To: <BANLkTin=3jrY0kzjw-CsOZM07O4XtVBtWQ@mail.gmail.com>
References: <BANLkTinB6uFAWDnq-Ot1tiSPi99pu0Oruw@mail.gmail.com>
	<BANLkTikkF95DCGgpQJ-kKCR-U_=X7fF5+g@mail.gmail.com>
	<BANLkTinAXp_u4+0OwNjY5dEDH5YHmFR-fw@mail.gmail.com>
	<65A0D45D-0666-493C-B53A-D9DC882EE77C@freeswitch.org>
	<BANLkTin8Vdg=a3em_2O1bkX3C5qoGLCBBg@mail.gmail.com>
	<BANLkTi=u9d5NWeO=Z1Rtnn_7+qAqb1cuKQ@mail.gmail.com>
	<BANLkTin=3jrY0kzjw-CsOZM07O4XtVBtWQ@mail.gmail.com>
Message-ID: <1303914967008-6309679.post@n2.nabble.com>

Please open a Jira on this and attach all relevent information

enable debug logging
sofia siptrace global on

And if it is reproducable by the default config - outline the flow and make
notes in the bug report.

Thanks

--
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From krice at freeswitch.org  Wed Apr 27 18:38:47 2011
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 27 Apr 2011 09:38:47 -0500
Subject: [Freeswitch-users] webphone app
In-Reply-To: <BANLkTimL8jAcRCzg8eDFnDwszyVpdZd8-w@mail.gmail.com>
Message-ID: <C9DD94A7.1488F%krice@freeswitch.org>

If you want something that works why not just look at FlashPhoner, no its
not Free, but it works, the Client side is opensource, used Wowza Media
Server for SIP/RTP-RTMP and scales really well. Also has the bonus that
there is NO GPLv3 liability...




On 4/27/11 4:52 AM, "Nicolas Brenner" <nico at clickfono.com> wrote:

> There's a sample client included, you can check it out here:
> 
> http://myprojectguide.org/p/siprtmp/
> 
> You can use that to test it for connecting to your rtpm/sip gateway and sip
> server. The source code is included in a folder of the rtmplite project, which
> is locate here:
> 
> http://code.google.com/p/rtmplite/
> 
> 
> On Tue, Apr 26, 2011 at 9:13 PM, Henry Huang <b_ball_henry at hotmail.com> wrote:
>> Do you have to develop a flash client from scratch or this project has a
>> sample client to use?
>> 
>> Henry
>> 
>> On Wed, Apr 27, 2011 at 1:45 AM, Nicolas Brenner <nico at clickfono.com> wrote:
>>> It's worked a lot better than red5 for me, using a lot less resources, but
>>> that's my experience.
>>> 
>>> 
>>> On Tue, Apr 26, 2011 at 1:38 PM, Madovsky <infos at madovsky.org> wrote:
>>>> sirtmp is not really stable and use a lot of CPU
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Dennis.Young at supportkids.com  Wed Apr 27 18:42:00 2011
From: Dennis.Young at supportkids.com (Dennis Young)
Date: Wed, 27 Apr 2011 09:42:00 -0500
Subject: [Freeswitch-users] Freetdm - Wanpipe - ftmod_sangoma_isdn - no
	caller_name
Message-ID: <32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A46A@jehuty.supportkids.com>

Setting "facility" to "yes" caused the following errors.   However I do see the caller_number on the second line and the caller_name in fifth line, but as you can tell it can't make out what the DNIS/DID number is.  The switchtype is set to "national" just like it was set with boost. 

With "facility" set to "no", everything works except no caller_name.

Any ideas?

2011-04-26 20:57:12.685250 [INFO] ftmod_sangoma_isdn_stack_rcv.c:75 [s1c1][1:1] Received SETUP (suId:1 suInstId:0 spInstId:1)
2011-04-26 20:57:12.685250 [INFO] ftmod_sangoma_isdn_stack_hndl.c:142 [s1c1][1:1] Incoming call: Called No:[6025] Calling No:[5127518487]
2011-04-26 20:57:12.685250 [CRIT] ftdm_io.c:5993 Overwriting non-cleared call-id
f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.685250 [NOTICE] switch_channel.c:816 New Channel FreeTDM/1:1/????????????????????????????????????????????????????????????????????????????????? [f1e75758-8028-4e07-a118-247c49cbe85c]
2011-04-26 20:57:12.700875 [INFO] ftmod_sangoma_isdn_stack_out.c:165 [s1c1][1:1] Sending PROCEED (suId:1 suInstId:1 spInstId:1 dchan:1 ces:0)
f1e75758-8028-4e07-a118-247c49cbe85c 
2011-04-26 20:57:12.700875 [INFO] mod_dialplan_xml.c:331 Processing YOUNG DENNIS   ???????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????? <???????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????>->????????????????????????????????????????????????????????????????????????????????? in context public
f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.700875 [NOTICE] mod_dptools.c:1184 outside_call=true, ringback=%(2000,4000,440.0,480.0)
2011-04-26 20:57:12.700875 [INFO] ftmod_sangoma_isdn_stack_out.c:223 [s1c1][1:1] Sending ALERT (suId:1 suInstId:1 spInstId:1 dchan:1 ces:0)
f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.700875 [NOTICE] mod_dptools.c:698 Ring Ready FreeTDM/1:1/?????????????????????????????????????????????????????????????????????????????????!
f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.700875 [NOTICE] mod_dptools.c:698 Ring-Ready FreeTDM/1:1/?????????????????????????????????????????????????????????????????????????????????!
f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.716500 [NOTICE] switch_core_state_machine.c:189 FreeTDM/1:1/????????????????????????????????????????????????????????????????????????????????? has executed the last dialplan instruction, hanging up.
f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.716500 [NOTICE] switch_core_state_machine.c:191 Hangup FreeTDM/1:1/????????????????????????????????????????????????????????????????????????????????? [CS_EXECUTE] [NORMAL_CLEARING]
2011-04-26 20:57:12.716500 [INFO] ftmod_sangoma_isdn_stack_out.c:373 [s1c1][1:1] Sending DISCONNECT (suId:1 suInstId:1 spInstId:1)
f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.716500 [NOTICE] switch_core_session.c:1304 Session 1 (FreeTDM/1:1/?????????????????????????????????????????????????????????????????????????????????) Ended
f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.716500 [NOTICE] switch_core_session.c:1306 Close Channel FreeTDM/1:1/????????????????????????????????????????????????????????????????????????????????? [CS_DESTROY]
2011-04-26 20:57:12.747750 [INFO] ftmod_sangoma_isdn_stack_rcv.c:245 [s1c1][1:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:1 spInstId:1)
2011-04-26 20:57:12.747750 [CRIT] ftdm_io.c:6025 Cannot clear call with invalid call-id


...dly

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Tuesday, April 26, 2011 6:47 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freetdm - Wanpipe - ftmod_sangoma_isdn

Set this to YES

/b

On Apr 26, 2011, at 3:45 PM, Dennis Young wrote:

> <param name="facility" value="no" />




Notice: This electronic transmission and its attachments are confidential and protected by applicable state and/or federal law.  Any use, reading, dissemination, distribution, copying or storage of this information by anyone other than the intended recipient is strictly prohibited.  If you are not the intended recipient, please immediately notify the sender by return email or telephone and delete this message and its attachments from your system.


From jeff at jefflenk.com  Wed Apr 27 18:45:46 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 27 Apr 2011 07:45:46 -0700 (PDT)
Subject: [Freeswitch-users] bypass_media=true and socket
In-Reply-To: <6B85FABC3A341E4CA5D5D96AD9E4FE581745B9FE@Mailbox.millicorp.com>
References: <6B85FABC3A341E4CA5D5D96AD9E4FE581745B9FE@Mailbox.millicorp.com>
Message-ID: <1303915546268-6309705.post@n2.nabble.com>

Socket is a custom app wrote by you?

What are you trying to accomplish with bypass media?

--
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From lists at telefaks.de  Wed Apr 27 19:03:54 2011
From: lists at telefaks.de (Peter Steinbach)
Date: Wed, 27 Apr 2011 17:03:54 +0200
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
In-Reply-To: <BANLkTi=ACjRN5xUW5HuqN36DJcg01RSFHw@mail.gmail.com>
References: <4DAECCA8.1050203@gmx.net>	<BANLkTikH9QMuhvKXqfBu78fOzqxoME3X8A@mail.gmail.com>	<4DB328AF.8090107@telefaks.de>	<BANLkTikz_PSaE6T99jnrJSgEFJSWBpyxvw@mail.gmail.com>	<4DB3E075.1050202@telefaks.de>	<BANLkTi=_nMDp_kjCrDmUkfLM-tMXt+nZ1g@mail.gmail.com>	<BANLkTi=aKo+E1Dj1gC2Q+OZLeegr6V0-rQ@mail.gmail.com>	<4DB7ECE1.2060603@telefaks.de>
	<BANLkTi=ACjRN5xUW5HuqN36DJcg01RSFHw@mail.gmail.com>
Message-ID: <4DB8305A.3030409@telefaks.de>

Hello Guru,

I got one step further. Thanks for your hint. The agent can now receive
a call via uuid-standby. When caller hangs up then the agent is taken
back into uuid-standby listening MOH.

However this only works max 3-4 times.
After the agent has taken 2-4 calls, the agent (208 below) and the
caller (200 below) are hung at the time the agent receives a new call
with the following error:

2011-04-27 14:56:24.046711 [WARNING] mod_callcenter.c:2516 Couldn't play
file 'local_stream://moh', continuing wait with no audio
2011-04-27 14:56:24.047716 [DEBUG] mod_callcenter.c:2544 Member 200
<200> abandoned waiting in queue Sales at my.domain
2011-04-27 14:56:24.048717 [DEBUG] mod_callcenter.c:2577 Member "200"
<200> exit queue Sales at my.domain due to BREAK_OUT
2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:371
(sofia/internal/200 at my.domain) State EXECUTE going to sleep
2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/200 at my.domain) Running State Change CS_HIBERNATE
2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:386
(sofia/internal/200 at my.domain) State HIBERNATE
2011-04-27 14:56:24.049764 [DEBUG] mod_sofia.c:221
sofia/internal/200 at my.domain SOFIA HIBERNATE
2011-04-27 14:56:24.049764 [DEBUG] switch_ivr_bridge.c:732
(sofia/internal/200 at my.domain) State Change CS_HIBERNATE -> CS_RESET
2011-04-27 14:56:24.049764 [DEBUG] switch_core_session.c:1114 Send
signal sofia/internal/200 at my.domain [BREAK]
2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:386
(sofia/internal/200 at my.domain) State HIBERNATE going to sleep
2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/200 at my.domain) Running State Change CS_RESET
2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:367
(sofia/internal/200 at my.domain) State RESET
2011-04-27 14:56:24.049764 [DEBUG] mod_sofia.c:165
sofia/internal/200 at my.domain SOFIA RESET
2011-04-27 14:56:24.049764 [DEBUG] switch_ivr_bridge.c:717
sofia/internal/200 at my.domain CUSTOM RESET
2011-04-27 14:56:24.049764 [DEBUG] switch_ivr_bridge.c:724
(sofia/internal/200 at my.domain) State Change CS_RESET -> CS_SOFT_EXECUTE
2011-04-27 14:56:24.049764 [DEBUG] switch_core_session.c:1114 Send
signal sofia/internal/200 at my.domain [BREAK]
2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:367
(sofia/internal/200 at my.domain) State RESET going to sleep
2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:325
(sofia/internal/200 at my.domain) Running State Change CS_SOFT_EXECUTE
2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:377
(sofia/internal/200 at my.domain) State SOFT_EXECUTE
2011-04-27 14:56:24.049764 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
2011-04-27 14:56:24.049764 [DEBUG] switch_ivr_bridge.c:742
sofia/internal/200 at my.domain CUSTOM SOFT_EXECUTE
2011-04-27 14:56:24.049764 [DEBUG] switch_channel.c:2562
(sofia/internal/208 at my.domain) Callstate Change ACTIVE -> HANGUP
2011-04-27 14:56:24.050715 [NOTICE] switch_ivr_originate.c:1045 Hangup
sofia/internal/208 at my.domain [CS_HIBERNATE] [DESTINATION_OUT_OF_ORDER]

Any idea how to solve this?

Best regards
Peter

guru singh schrieb:
> Hi Peter,
>
> Yes, there were some changes made again. Try using
> transfer_after_bridge application.
> http://wiki.freeswitch.org/wiki/Variable_transfer_after_bridge
>
> Regards,
> guru
>
> On Wed, Apr 27, 2011 at 3:46 PM, Peter Steinbach <lists at telefaks.de> wrote:
>   
>> Hello Guru,
>>
>> I upgraded to latest GIT, but behaviour is still the same.
>>
>> Best regards
>> Peter
>>
>> guru singh schrieb:
>>
>> Hi Peter,
>>
>> Please try latest git. Moc's commit has fixed the issue.
>>
>> Regards,
>> guru
>>
>> On Sun, Apr 24, 2011 at 6:09 PM, guru singh <grsingh750 at gmail.com> wrote:
>>
>>
>> Hi Peter,
>>
>> You're right. Please ignore my previous message, status 'Available (On
>> Demand)' should also be fine.
>> I've tried it and I see the same behavior as you. Reading the logs, I
>> can see that nothing after playback gets executed once the call is
>> hungup. It's not just transfer, any other application also is not
>> getting executing. Something is amiss.
>> Maybe moc or somebody else will point it out.
>>
>> Regards,
>> guru
>>
>> On Sun, Apr 24, 2011 at 2:03 PM, Peter Steinbach <lists at telefaks.de> wrote:
>>
>>
>> Hello Guru,
>>
>> thanks for your hint, however this did not help.
>> The point is that according to the dialplan the agent should be transferred
>> to the same extension again an wait. In my case, when the call hanges up,
>> there is no attempt to continue with the dialplan.
>> So I expect it does not have to do with the agent's setings.
>>
>> Best regards
>> Peter
>>
>>
>> guru singh schrieb:
>>
>> Hi Peter
>>
>> Try setting the status as 'Available' instead of 'Available (On Demand)'
>> In case of 'Available (On Demand)' after the call ends, the agent's
>> status is set to 'idle', so therefore no calls are given to the
>> specific agent. I'm not too sure if this is the only change required
>> to get the behavior you expect. I've not tried it on my box yet. I can
>> only do it Monday and let you know.
>>
>> Regards,
>> guru
>>
>> On Sun, Apr 24, 2011 at 12:59 AM, Peter Steinbach <lists at telefaks.de> wrote:
>>
>>
>> Thank you guru,
>>
>> I tried the example in the wiki. This worked.
>> I wanted the agent also to wait again with MOH after the caller hung up.
>> This did not work in my environment (Freeswitch git April 2011). The
>> agent was always hungup after the caller hung up and was not transferred
>> to the same dialplan extension again.
>> Also hangup after bridge =false did not work.
>>
>> Does this work in your environment?
>>
>> Best regards
>> Peter
>>
>>
>> guru singh schrieb:
>>
>>
>> Hi Peter,
>>
>> I've been using mod_callcenter for a while and must say it works really
>> well.
>> I just tried the uuid-standby strategy and basically it's exactly what
>> you say the asterisk thing does.
>> See the dialplan example.
>> http://wiki.freeswitch.org/wiki/Mod_callcenter#uuid-standby
>> Agent is dialing 4099 and listening to MOH. When a call arrives, it's
>> bridged directly to the agent.
>>
>> Regards
>> guru
>>
>> On Wed, Apr 20, 2011 at 5:38 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:
>>
>>
>>
>> Hello,
>>
>> I am trying to use the mod_callcenter functionality. This works nicely
>> so far so thank you to everybody involved for programming this nice module!
>> But I am stuck somehow with uuid-standby. Can anybody explain how
>> uuid-standby works?
>>
>> Another question: In the Asterisk based callcenter solution named
>> "Vicidial", an agent can be held permanently in a conference, waiting
>> for calls who are bridged to his uuid in the conference. Can this be
>> haviour be done with mod_callcenter?
>>
>>
>> Best regards
>> Peter
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> With kind regards
>> Peter Steinbach
>>
>> Telefaks Services GmbH
>> mailto:lists (att) telefaks.de
>> Internet: www.telefaks.de
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>     
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jeff at jefflenk.com  Wed Apr 27 20:24:34 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 27 Apr 2011 09:24:34 -0700 (PDT)
Subject: [Freeswitch-users] mod_crd_sqlite entry limit and sqlite
	segfaults on triggers
In-Reply-To: <4DB809B4.5060601@yx.cl>
References: <4DB5E0F1.4050602@yx.cl> <4DB809B4.5060601@yx.cl>
Message-ID: <1303921474677-6310155.post@n2.nabble.com>

Please post this type of thing to http://jira.freeswitch.org in the future.

I have taken your change and committed it this time. It is very easy for
these changes to get lost otherwise.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-crd-sqlite-entry-limit-and-sqlite-segfaults-on-triggers-tp6304052p6310155.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Wed Apr 27 20:29:30 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Apr 2011 09:29:30 -0700
Subject: [Freeswitch-users] FreeSWITCH Conference Call Today - Safi Systems
	IVR Builder
Message-ID: <BANLkTikSdnsKMwVJ1SL1jCJSWqrBJ+7Qew@mail.gmail.com>

Hello all!

We are happy to have Zac Wolfe from Safi Systems joining us today to talk
about the IVR building system they have created. There is a
gotomeeting.comlink on the agenda page here:

http://wiki.freeswitch.org/wiki/FS_weekly_2011_04_27

We look forward to this discussion!

-Michael
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From benkokakao at gmail.com  Wed Apr 27 20:33:16 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Wed, 27 Apr 2011 18:33:16 +0200
Subject: [Freeswitch-users] Tuning Up Freeswitch
In-Reply-To: <4DB7CAB5.9080801@cupis.co.uk>
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
	<4DB7CAB5.9080801@cupis.co.uk>
Message-ID: <BANLkTinWH+ccEUX3+i93CrBZz2bXdAwWPg@mail.gmail.com>

> Debian default kernel sets CONFIG_HZ=100
> CentOS default kernel sets CONFIG_HZ=1000

Actually, Debians default is CONFIG_HZ=250

So has anyone baked a 1000Hz Kernel on Debian and seen any performance changes?

Best regards
Christian


From msc at freeswitch.org  Wed Apr 27 20:42:11 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Apr 2011 09:42:11 -0700
Subject: [Freeswitch-users] mod_callcenter and uuid-standby
In-Reply-To: <4DB8305A.3030409@telefaks.de>
References: <4DAECCA8.1050203@gmx.net>
	<BANLkTikH9QMuhvKXqfBu78fOzqxoME3X8A@mail.gmail.com>
	<4DB328AF.8090107@telefaks.de>
	<BANLkTikz_PSaE6T99jnrJSgEFJSWBpyxvw@mail.gmail.com>
	<4DB3E075.1050202@telefaks.de>
	<BANLkTi=_nMDp_kjCrDmUkfLM-tMXt+nZ1g@mail.gmail.com>
	<BANLkTi=aKo+E1Dj1gC2Q+OZLeegr6V0-rQ@mail.gmail.com>
	<4DB7ECE1.2060603@telefaks.de>
	<BANLkTi=ACjRN5xUW5HuqN36DJcg01RSFHw@mail.gmail.com>
	<4DB8305A.3030409@telefaks.de>
Message-ID: <BANLkTikgKTC4P2F-bDDxfjExZ4oVcekEGg@mail.gmail.com>

Get a complete SIP trace and debug log of this happening for one agent, then
open a Jira and assign it to Moc. (Marc Olivier Chouinard.)

-MC

On Wed, Apr 27, 2011 at 8:03 AM, Peter Steinbach <lists at telefaks.de> wrote:

>  Hello Guru,
>
> I got one step further. Thanks for your hint. The agent can now receive a
> call via uuid-standby. When caller hangs up then the agent is taken back
> into uuid-standby listening MOH.
>
> However this only works max 3-4 times.
> After the agent has taken 2-4 calls, the agent (208 below) and the caller
> (200 below) are hung at the time the agent receives a new call with the
> following error:
>
> 2011-04-27 14:56:24.046711 [WARNING] mod_callcenter.c:2516 Couldn't play
> file 'local_stream://moh', continuing wait with no audio
> 2011-04-27 14:56:24.047716 [DEBUG] mod_callcenter.c:2544 Member 200 <200>
> abandoned waiting in queue Sales at my.domain
> 2011-04-27 14:56:24.048717 [DEBUG] mod_callcenter.c:2577 Member "200" <200>
> exit queue Sales at my.domain due to BREAK_OUT
> 2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:371 (
> sofia/internal/200 at my.domain) State EXECUTE going to sleep
> 2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:325 (
> sofia/internal/200 at my.domain) Running State Change CS_HIBERNATE
> 2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:386 (
> sofia/internal/200 at my.domain) State HIBERNATE
> 2011-04-27 14:56:24.049764 [DEBUG] mod_sofia.c:221
> sofia/internal/200 at my.domain SOFIA HIBERNATE
> 2011-04-27 14:56:24.049764 [DEBUG] switch_ivr_bridge.c:732 (
> sofia/internal/200 at my.domain) State Change CS_HIBERNATE -> CS_RESET
> 2011-04-27 14:56:24.049764 [DEBUG] switch_core_session.c:1114 Send signal
> sofia/internal/200 at my.domain [BREAK]
> 2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:386 (
> sofia/internal/200 at my.domain) State HIBERNATE going to sleep
> 2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:325 (
> sofia/internal/200 at my.domain) Running State Change CS_RESET
> 2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:367 (
> sofia/internal/200 at my.domain) State RESET
> 2011-04-27 14:56:24.049764 [DEBUG] mod_sofia.c:165
> sofia/internal/200 at my.domain SOFIA RESET
> 2011-04-27 14:56:24.049764 [DEBUG] switch_ivr_bridge.c:717
> sofia/internal/200 at my.domain CUSTOM RESET
> 2011-04-27 14:56:24.049764 [DEBUG] switch_ivr_bridge.c:724 (
> sofia/internal/200 at my.domain) State Change CS_RESET -> CS_SOFT_EXECUTE
> 2011-04-27 14:56:24.049764 [DEBUG] switch_core_session.c:1114 Send signal
> sofia/internal/200 at my.domain [BREAK]
> 2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:367 (
> sofia/internal/200 at my.domain) State RESET going to sleep
> 2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:325 (
> sofia/internal/200 at my.domain) Running State Change CS_SOFT_EXECUTE
> 2011-04-27 14:56:24.049764 [DEBUG] switch_core_state_machine.c:377 (
> sofia/internal/200 at my.domain) State SOFT_EXECUTE
> 2011-04-27 14:56:24.049764 [DEBUG] mod_sofia.c:558 SOFIA SOFT_EXECUTE
> 2011-04-27 14:56:24.049764 [DEBUG] switch_ivr_bridge.c:742
> sofia/internal/200 at my.domain CUSTOM SOFT_EXECUTE
> 2011-04-27 14:56:24.049764 [DEBUG] switch_channel.c:2562 (
> sofia/internal/208 at my.domain) Callstate Change ACTIVE -> HANGUP
> 2011-04-27 14:56:24.050715 [NOTICE] switch_ivr_originate.c:1045 Hangup
> sofia/internal/208 at my.domain [CS_HIBERNATE] [DESTINATION_OUT_OF_ORDER]
>
> Any idea how to solve this?
>
>
> Best regards
> Peter
>
> guru singh schrieb:
>
> Hi Peter,
>
> Yes, there were some changes made again. Try using
> transfer_after_bridge application.http://wiki.freeswitch.org/wiki/Variable_transfer_after_bridge
>
> Regards,
> guru
>
> On Wed, Apr 27, 2011 at 3:46 PM, Peter Steinbach <lists at telefaks.de> <lists at telefaks.de> wrote:
>
>
>  Hello Guru,
>
> I upgraded to latest GIT, but behaviour is still the same.
>
> Best regards
> Peter
>
> guru singh schrieb:
>
> Hi Peter,
>
> Please try latest git. Moc's commit has fixed the issue.
>
> Regards,
> guru
>
> On Sun, Apr 24, 2011 at 6:09 PM, guru singh <grsingh750 at gmail.com> <grsingh750 at gmail.com> wrote:
>
>
> Hi Peter,
>
> You're right. Please ignore my previous message, status 'Available (On
> Demand)' should also be fine.
> I've tried it and I see the same behavior as you. Reading the logs, I
> can see that nothing after playback gets executed once the call is
> hungup. It's not just transfer, any other application also is not
> getting executing. Something is amiss.
> Maybe moc or somebody else will point it out.
>
> Regards,
> guru
>
> On Sun, Apr 24, 2011 at 2:03 PM, Peter Steinbach <lists at telefaks.de> <lists at telefaks.de> wrote:
>
>
> Hello Guru,
>
> thanks for your hint, however this did not help.
> The point is that according to the dialplan the agent should be transferred
> to the same extension again an wait. In my case, when the call hanges up,
> there is no attempt to continue with the dialplan.
> So I expect it does not have to do with the agent's setings.
>
> Best regards
> Peter
>
>
> guru singh schrieb:
>
> Hi Peter
>
> Try setting the status as 'Available' instead of 'Available (On Demand)'
> In case of 'Available (On Demand)' after the call ends, the agent's
> status is set to 'idle', so therefore no calls are given to the
> specific agent. I'm not too sure if this is the only change required
> to get the behavior you expect. I've not tried it on my box yet. I can
> only do it Monday and let you know.
>
> Regards,
> guru
>
> On Sun, Apr 24, 2011 at 12:59 AM, Peter Steinbach <lists at telefaks.de> <lists at telefaks.de> wrote:
>
>
> Thank you guru,
>
> I tried the example in the wiki. This worked.
> I wanted the agent also to wait again with MOH after the caller hung up.
> This did not work in my environment (Freeswitch git April 2011). The
> agent was always hungup after the caller hung up and was not transferred
> to the same dialplan extension again.
> Also hangup after bridge =false did not work.
>
> Does this work in your environment?
>
> Best regards
> Peter
>
>
> guru singh schrieb:
>
>
> Hi Peter,
>
> I've been using mod_callcenter for a while and must say it works really
> well.
> I just tried the uuid-standby strategy and basically it's exactly what
> you say the asterisk thing does.
> See the dialplan example.http://wiki.freeswitch.org/wiki/Mod_callcenter#uuid-standby
> Agent is dialing 4099 and listening to MOH. When a call arrives, it's
> bridged directly to the agent.
>
> Regards
> guru
>
> On Wed, Apr 20, 2011 at 5:38 PM, Peter P GMX <Prometheus001 at gmx.net> <Prometheus001 at gmx.net> wrote:
>
>
>
> Hello,
>
> I am trying to use the mod_callcenter functionality. This works nicely
> so far so thank you to everybody involved for programming this nice module!
> But I am stuck somehow with uuid-standby. Can anybody explain how
> uuid-standby works?
>
> Another question: In the Asterisk based callcenter solution named
> "Vicidial", an agent can be held permanently in a conference, waiting
> for calls who are bridged to his uuid in the conference. Can this be
> haviour be done with mod_callcenter?
>
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing 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
>
>
>
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing 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
>
>
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing 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
>
>
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbHmailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>      _______________________________________________
> FreeSWITCH-users mailing 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
>
>
>
> --
> With kind regards
> Peter Steinbach
>
> Telefaks Services GmbH
> mailto:lists <lists> (att) telefaks.de
> Internet: www.telefaks.de
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ktaylor91 at yahoo.com  Wed Apr 27 21:31:56 2011
From: ktaylor91 at yahoo.com (Kenneth Taylor)
Date: Wed, 27 Apr 2011 10:31:56 -0700 (PDT)
Subject: [Freeswitch-users] frame per packet
In-Reply-To: <6496B73C-A3EA-4E0A-889A-973613B35CC1@ipeva.fr>
References: <630958.78492.qm@web121503.mail.ne1.yahoo.com>
	<546FE599-8C29-44AD-8D78-9711BF397544@gmail.com>
	<934071.68468.qm@web121509.mail.ne1.yahoo.com>
	<6496B73C-A3EA-4E0A-889A-973613B35CC1@ipeva.fr>
Message-ID: <478413.79262.qm@web121502.mail.ne1.yahoo.com>

Hi David thanks.
This was the solution, I have added in my client's sdp the ptime attribute and 
it work.

thanks again,
ken



________________________________
From: David Ponzone <david.ponzone at ipeva.fr>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
Sent: Fri, April 22, 2011 11:14:54 AM
Subject: Re: [Freeswitch-users] frame per packet

Kenneth,

It seemed to me that it's the endpoint which must first decide the ptime it 
wants to use.
Then FreeSWITCH will relay it.
What you did will probably help FreeSWITCH to advertise only GSM with 60ms ptime 
to the other endpoint, but I think you may have achieved that by using:
GSM at 60i
in your codec strings.


David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? 
l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non 
autoris?e est interdite. Tout message ?lectronique est susceptible 
d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a 
?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, 
merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




Le 22/04/2011 ? 07:56, Kenneth Taylor a ?crit :

Hello,
>I think I found a solution.
>At mod_spandsp_codecs_load function on mod_spandsp_codecs, there is a place 
>where FS is loading all of its codecs.
>for the gsm codec (the codec i'm using) it is loading 6 implementations of the 
>codec sorted by the number of frames per network packet.
>When a conversation is starting FS choose the last implementation which is for 1 
>frame per packet.
>I disabled all the implementations except the one with 3 frames per packet, and 
>now it's working.
>
>
>Just to make sure I'm not missing anything.
>
>
>TNX everyone
>Ken
>
>
>
>
>
________________________________
From: Steven Ayre <steveayre at gmail.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>Sent: Mon, April 18, 2011 11:04:40 PM
>Subject: Re: [Freeswitch-users] frame per packet
>
>
>AFAIK, a rtp packet *is* a frame, so no you can't.
>
>
>As the other poster said, use a higher ptime. That will make the frame store a 
>longer period of time which will do the same as that you want.
>
>
>Overhead will be lower, but quality will be worse if you drop packets or they 
>fail to arrive in a timely manner. That's very likely on gsm, and the packet 
>size will mean it takes longer to be delivered which'll mean more delay and 
>probably more packets arriving too late. You'll need to experiment to find a 
>good balance. And remember what works well in a city might fail to work in the 
>country where coverage is poorer.
>
>Steve on iPhone
>
>On 18 Apr 2011, at 09:49, Kenneth Taylor <ktaylor91 at yahoo.com> wrote:
>
>
>Hi,
>>I'm building a voip client for android phones over gsm and consider using FS as 
>>sever.
>>In order to save cellular costs I want to put more than one frame per Rtp 
>>packet.
>>
>>
>>Is there any easy solution to do it in FS? 
>>
>>
>>Or, where is the right place to add it? In mod_sofia or sofia_glue? should I go 
>>to lower levels?
>>I don't think I should do it in the codec, it may cause problems in the rtp 
>>sequence number and probably a lot more.
>>
>>
>>TNX,
>>Ken
_______________________________________________
>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>_______________________________________________
>FreeSWITCH-users mailing list
>FreeSWITCH-users at 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 ktaylor91 at yahoo.com  Wed Apr 27 21:39:22 2011
From: ktaylor91 at yahoo.com (Kenneth Taylor)
Date: Wed, 27 Apr 2011 10:39:22 -0700 (PDT)
Subject: [Freeswitch-users] Instant message problem
Message-ID: <257577.18012.qm@web121507.mail.ne1.yahoo.com>

Hello,

I have a strange problem when trying to send IM messages between two clients 
that connected to FS.
When not during call, I'm sending a MESSAGE with the IM string and it works 
well.
But when the two clients are during call, I see in wireshark that the message is 
coming to FS, but not forwarded to the second client.
There is nothing in the log file regarding this message.

Is this a known bug?

Thank you for helping,
Kenneth.
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From infos at madovsky.org  Wed Apr 27 21:50:17 2011
From: infos at madovsky.org (Madovsky)
Date: Wed, 27 Apr 2011 13:50:17 -0400
Subject: [Freeswitch-users] Tuning Up Freeswitch
References: <BANLkTimVN12g+X6O7WziUwc2io2nZfH8PA@mail.gmail.com>
	<C5390B3C-4F5E-4D88-824D-95F261A29385@flylabs.com>
Message-ID: <A32AFBCD54D14437A199337660F137F4@e1705>

for info I'm using the old Fedora10 64bits
and everything is working fine with fq at 1000hz


----- Original Message ----- 
From: "Ariel Monaco" <arielmonaco at flylabs.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Tuesday, April 26, 2011 8:24 PM
Subject: Re: [Freeswitch-users] Tuning Up Freeswitch


We had high CPU utilization peaks in the past, which lead to some audio 
issues (clipping). We were using debian at that time, which was a 
customer-side requirement.

I'm not a kernel guru but I remember this had something to do with kernel 
timer cycles and the issue was address by adding "divider=10" or 
"divider=100" as a kernel's boot loader option.

My 2 cents,
Ariel

On Apr 21, 2011, at 13:24 , Antonio Teixeira wrote:

> Hello List.
>
> I'm currently integrating an IVR in python together with freeswitch using 
> mod_python and ESL and my life has been well until ...
> The flow of calls went over 80 simultaneous calls.
> Now freeswitch starts sending packets with huge delays ( even when 
> establishing the call , mainly the 200 ) and firing up the IVR with tons 
> of delay up to 20 seconds.
>
> So i searched the wiki forums and mailing list:
>
> Put freeswitch on a diet , trimmed modules.conf
> Played with the ulimit stuff.
> Played with the IVRS to reduce load to a minimum and i was able to squeeze 
> more 5 calls of performance.
>
> The problem is :
>
> Top shows
> top - 16:14:33 up 35 days,  8:15,  3 users,  load average: 1.92, 1.76, 
> 1.78
> Tasks: 133 total,   1 running, 132 sleeping,   0 stopped,   0 zombie
> Cpu(s):  1.4%us,  3.3%sy,  0.0%ni, 94.6%id,  0.0%wa,  0.3%hi,  0.5%si, 
> 0.0%st
> Mem:   8193336k total,  1639156k used,  6554180k free,   177208k buffers
> Swap: 19534904k total,        0k used, 19534904k free,  1062272k cached
>
>   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+  COMMAND
> 31361 yadayada      20   0  716m 164m 9628 S   73  2.1 155:17.85 
> freeswitch
>
> Freeswitch goes up to 150 % and puff there goes the MOS down to 0.
>
>
> Some basic System Info :
> Debian 6.0 ( i heard the timming module is affected by Debian , but if the 
> CPU % gets lower than 95% everything will be more stable)
> Python 2.5
>
> 2 x Intel(R) Xeon(R) CPU           E5506  @ 2.13GHz
> 8 GB of Ram
>
> as you can see 94 % of the "Cpu Power" is sleeping :\
>
>
> It appears freeswitch is only capable of using let's say "one cpu"/thread 
> ??
> Do you guys recommend simply starting more instances or redoing the IVR 
> stuff.
>
>
> Hope you guys can help me out.
>
> Thanks
> Ant?nio Teixeira
>
>
>
>
>
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Ariel Monaco ? Systems Engineer
Flylabs - Open Source Telecommunications and  IT Consultants

Address: Potos? 4456 C1199ACP - Buenos Aires - Argentina
Web: http://flylabs.com
E-Mail: arielmonaco at flylabs.com
Tel. +54 (11) 4982-2689, +1 (315) 688-7333


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


-- 
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.



From steveayre at gmail.com  Wed Apr 27 22:00:19 2011
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 27 Apr 2011 19:00:19 +0100
Subject: [Freeswitch-users] bypass_media=true and socket
In-Reply-To: <1303915546268-6309705.post@n2.nabble.com>
References: <6B85FABC3A341E4CA5D5D96AD9E4FE581745B9FE@Mailbox.millicorp.com>
	<1303915546268-6309705.post@n2.nabble.com>
Message-ID: <BANLkTi=OLO5GiDXcnYXGUHC3n_KpGRxbCQ@mail.gmail.com>

Jeff, socket is the outbound ESL.

Tim, it would be useful to know what you're sending via ESL and see
debug-level logs with siptrace enabled so we can see what's going on.

-Steve


On 27 April 2011 15:45, Jeff Lenk <jeff at jefflenk.com> wrote:

> Socket is a custom app wrote by you?
>
> What are you trying to accomplish with bypass media?
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/bypass-media-true-and-socket-tp6309332p6309705.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 michel at arneill-py.sacramento.ca.us  Wed Apr 27 22:24:49 2011
From: michel at arneill-py.sacramento.ca.us (Michel Py)
Date: Wed, 27 Apr 2011 11:24:49 -0700
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
	with productivity license.
In-Reply-To: <BANLkTimzyrNQOLKs_CRRtFKRpw6isCg+AQ@mail.gmail.com>
References: <471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local><BANLkTi=XmNDV51=xoNsViHUYaUZeyMpz_A@mail.gmail.com><471D76419F9EF642962323D13DF1DF69011E58@newserver.arneill-py.local><BANLkTi=o1nUa70X_PHU51M_TTGTfnSSdKg@mail.gmail.com><471D76419F9EF642962323D13DF1DF69011E59@newserver.arneill-py.local><BANLkTi=9Gu1vOD-KVdCSs3MbNxng+Tqv2w@mail.gmail.com><471D76419F9EF642962323D13DF1DF69011E5A@newserver.arneill-py.local>
	<BANLkTimzyrNQOLKs_CRRtFKRpw6isCg+AQ@mail.gmail.com>
Message-ID: <471D76419F9EF642962323D13DF1DF69011E5B@newserver.arneill-py.local>

Christian,

> So the BLF-configuration is probably working, but not as
> straightforward as documented in the wiki(attendant.uri and stuff).

That would not solve my problem; I have read in multiple places that
there could be only one attendant.uri, and I need more.

I guess I will fall back to buddies; if only I could make the light
blink when ringing instead of going directly solid red.

I have the feeling I'm missing something stupid, since it's my first
encounter with Polycom (I'm not thrilled, BTW). Oh well, thanks for the
time.

Michel.



From jeff at jefflenk.com  Wed Apr 27 22:42:20 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 27 Apr 2011 11:42:20 -0700 (PDT)
Subject: [Freeswitch-users] bypass_media=true and socket
In-Reply-To: <BANLkTi=OLO5GiDXcnYXGUHC3n_KpGRxbCQ@mail.gmail.com>
References: <6B85FABC3A341E4CA5D5D96AD9E4FE581745B9FE@Mailbox.millicorp.com>
	<1303915546268-6309705.post@n2.nabble.com>
	<BANLkTi=OLO5GiDXcnYXGUHC3n_KpGRxbCQ@mail.gmail.com>
Message-ID: <1303929740733-6310681.post@n2.nabble.com>

Ok sorry for the confusion Thanks Steven.

--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/bypass-media-true-and-socket-tp6309332p6310681.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From nboric at yx.cl  Wed Apr 27 23:56:47 2011
From: nboric at yx.cl (Neven Boric)
Date: Wed, 27 Apr 2011 16:56:47 -0300
Subject: [Freeswitch-users] mod_crd_sqlite entry limit and sqlite
 segfaults on triggers
In-Reply-To: <1303921474677-6310155.post@n2.nabble.com>
References: <4DB5E0F1.4050602@yx.cl> <4DB809B4.5060601@yx.cl>
	<1303921474677-6310155.post@n2.nabble.com>
Message-ID: <4DB874FF.5060901@yx.cl>



Jeff Lenk escribi?:
> Please post this type of thing to http://jira.freeswitch.org in the future.

Will do. What about the limit on the number of cdrs? Should I report 
that too?
> I have taken your change and committed it this time. It is very easy for
> these changes to get lost otherwise.
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-crd-sqlite-entry-limit-and-sqlite-segfaults-on-triggers-tp6304052p6310155.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From ctroncoso at redvoiss.net  Wed Apr 27 22:29:23 2011
From: ctroncoso at redvoiss.net (Camila Troncoso)
Date: Wed, 27 Apr 2011 15:29:23 -0300
Subject: [Freeswitch-users] mod_radius_cdr and Acct-Delay-Time
Message-ID: <9b86b6d280317812f3b7de59136aa457@mail.gmail.com>

Hi,



I?m working with radius accounting using the mod_radius_cdr module. I?m
having trouble with the date format that Radius server receive.  An example
of what the radius server receive is:



?Freeswitch-Callstartdate = "2011-04-21T18:23:59.780945-0400"?



This date format is very difficult to read and I want to change it  to make
the accounting easier. I search all around for some param or configuration
that allows me to do so, but I only find that this date format is define in
mod_radius_cdr.c module.



I?m also having problems with the parameter Acct-Delay-Time, it is not
increasing when the client resend the radius packet. I read the buildreq.c
code but didn?t find the problem.



Please some help with this matter.



Regards,



*Camila Troncoso **|* Ingeniero de Desarrollo

RedVoiss *|*ctroncoso at redvoiss.net

Santiago - Chile *|* +56 2 2408535

www.redvoiss.net
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From shukalo83 at yahoo.com  Wed Apr 27 23:55:38 2011
From: shukalo83 at yahoo.com (Bojan Sukalo)
Date: Wed, 27 Apr 2011 12:55:38 -0700 (PDT)
Subject: [Freeswitch-users] custom kernel
Message-ID: <95468.35434.qm@web38102.mail.mud.yahoo.com>

Hello.
Regarding this headline:?"FreeSWITCH Adds Custom Libc And Kernel To
Code Repository"
Where I can download this custom kernel and libc code?



From oseslija at gmail.com  Thu Apr 28 00:10:45 2011
From: oseslija at gmail.com (Ognjen Seslija)
Date: Wed, 27 Apr 2011 22:10:45 +0200
Subject: [Freeswitch-users] custom kernel
In-Reply-To: <95468.35434.qm@web38102.mail.mud.yahoo.com>
References: <95468.35434.qm@web38102.mail.mud.yahoo.com>
Message-ID: <BANLkTim4V0sEKPF9KMGZNYdNX803W_=nzw@mail.gmail.com>

Check the date.

On Wed, Apr 27, 2011 at 9:55 PM, Bojan Sukalo <shukalo83 at yahoo.com> wrote:

> Hello.
> Regarding this headline: "FreeSWITCH Adds Custom Libc And Kernel To
> Code Repository"
> Where I can download this custom kernel and libc code?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 brian at freeswitch.org  Thu Apr 28 00:14:00 2011
From: brian at freeswitch.org (Brian West)
Date: Wed, 27 Apr 2011 15:14:00 -0500
Subject: [Freeswitch-users] custom kernel
In-Reply-To: <BANLkTim4V0sEKPF9KMGZNYdNX803W_=nzw@mail.gmail.com>
References: <95468.35434.qm@web38102.mail.mud.yahoo.com>
	<BANLkTim4V0sEKPF9KMGZNYdNX803W_=nzw@mail.gmail.com>
Message-ID: <2341890C-1252-423F-9A7B-CE2FEB621B03@freeswitch.org>

/me falls over laughing.

/b

On Apr 27, 2011, at 3:10 PM, Ognjen Seslija wrote:

> Check the date.
> 
> On Wed, Apr 27, 2011 at 9:55 PM, Bojan Sukalo <shukalo83 at yahoo.com> wrote:
> 
>> Hello.
>> Regarding this headline: "FreeSWITCH Adds Custom Libc And Kernel To
>> Code Repository"
>> Where I can download this custom kernel and libc code?
>> 
> 

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From tayeb.meftah at gmail.com  Thu Apr 28 00:18:31 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Wed, 27 Apr 2011 22:18:31 +0200
Subject: [Freeswitch-users] custom kernel
In-Reply-To: <95468.35434.qm@web38102.mail.mud.yahoo.com>
References: <95468.35434.qm@web38102.mail.mud.yahoo.com>
Message-ID: <4DB87A17.1000409@gmail.com>

if you're seriou you're welcome to do it yourself
april fool ;)
:P
On 27/04/2011 21:55, Bojan Sukalo wrote:
> Hello.
> Regarding this headline: "FreeSWITCH Adds Custom Libc And Kernel To
> Code Repository"
> Where I can download this custom kernel and libc code?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 benkokakao at gmail.com  Thu Apr 28 00:25:29 2011
From: benkokakao at gmail.com (Christian Benke)
Date: Wed, 27 Apr 2011 23:25:29 +0300
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
 with productivity license.
In-Reply-To: <471D76419F9EF642962323D13DF1DF69011E5B@newserver.arneill-py.local>
References: <471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local>
	<BANLkTi=XmNDV51=xoNsViHUYaUZeyMpz_A@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E58@newserver.arneill-py.local>
	<BANLkTi=o1nUa70X_PHU51M_TTGTfnSSdKg@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E59@newserver.arneill-py.local>
	<BANLkTi=9Gu1vOD-KVdCSs3MbNxng+Tqv2w@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E5A@newserver.arneill-py.local>
	<BANLkTimzyrNQOLKs_CRRtFKRpw6isCg+AQ@mail.gmail.com>
	<471D76419F9EF642962323D13DF1DF69011E5B@newserver.arneill-py.local>
Message-ID: <BANLkTik=Ap6Yevy0OBc+Dto2-HdzLYjoBA@mail.gmail.com>

On 27 April 2011 21:24, Michel Py <michel at arneill-py.sacramento.ca.us> wrote:
> Christian,
>
>> So the BLF-configuration is probably working, but not as
>> straightforward as documented in the wiki(attendant.uri and stuff).
>
> That would not solve my problem; I have read in multiple places that
> there could be only one attendant.uri, and I need more.

Yeah, sorry, i'm out of ideas. Think about upgrading the phones if you
have a chance, the feature works just fine on the new generation of
phones(Had it setup in 10min last week).

> I have the feeling I'm missing something stupid, since it's my first
> encounter with Polycom (I'm not thrilled, BTW).

Well, it seems you did not yet see what nightmares other vendors sell ;-)
Polycom are phantastic phones(imho the best on the market), the
documentation is extensive and detailed and the provisioning options
are marvellous. The interface is intuitive and the hardware is well
laid-out, both mechanical and performance-wise.
These phones are clearly built by engineers and i'm not surprised they
are so popular.

Best regards
Christian


From jeff at jefflenk.com  Thu Apr 28 00:27:04 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Wed, 27 Apr 2011 13:27:04 -0700 (PDT)
Subject: [Freeswitch-users] mod_crd_sqlite entry limit and sqlite
	segfaults on triggers
In-Reply-To: <4DB874FF.5060901@yx.cl>
References: <4DB5E0F1.4050602@yx.cl> <4DB809B4.5060601@yx.cl>
	<1303921474677-6310155.post@n2.nabble.com> <4DB874FF.5060901@yx.cl>
Message-ID: <1303936024626-6311050.post@n2.nabble.com>

Thats more of an implementation question for the author. Daniel Swarbrick.

but you could open an issue on that too and assign it to him and he will
check into it.


--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-crd-sqlite-entry-limit-and-sqlite-segfaults-on-triggers-tp6304052p6311050.html
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From shukalo83 at yahoo.com  Thu Apr 28 00:24:42 2011
From: shukalo83 at yahoo.com (Bojan Sukalo)
Date: Wed, 27 Apr 2011 13:24:42 -0700 (PDT)
Subject: [Freeswitch-users] custom kernel
In-Reply-To: <2341890C-1252-423F-9A7B-CE2FEB621B03@freeswitch.org>
Message-ID: <644050.54425.qm@web38105.mail.mud.yahoo.com>

I don't get it.
http://www.freeswitch.org/node/316?is some kind of announcement for custom kernel??2.6.11.41 in git repo. I've used git clone for free switch but couldn't seem to find it.As for date, it is?Fri, 04/01/2011
Maybe you fell over laughing but I still don't get it.
Sorry.
--- On Wed, 4/27/11, Brian West <brian at freeswitch.org> wrote:

From: Brian West <brian at freeswitch.org>
Subject: Re: [Freeswitch-users] custom kernel
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Wednesday, April 27, 2011, 1:14 PM

/me falls over laughing.
/b
On Apr 27, 2011, at 3:10 PM, Ognjen Seslija wrote:
Check the date.

On Wed, Apr 27, 2011 at 9:55 PM, Bojan Sukalo <shukalo83 at yahoo.com> wrote:

Hello.
Regarding this headline: "FreeSWITCH Adds Custom Libc And Kernel To
Code Repository"
Where I can download this custom kernel and libc code?




-----Inline Attachment Follows-----

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From shukalo83 at yahoo.com  Thu Apr 28 00:29:57 2011
From: shukalo83 at yahoo.com (Bojan Sukalo)
Date: Wed, 27 Apr 2011 13:29:57 -0700 (PDT)
Subject: [Freeswitch-users] custom kernel
In-Reply-To: <2341890C-1252-423F-9A7B-CE2FEB621B03@freeswitch.org>
Message-ID: <70795.28354.qm@web38107.mail.mud.yahoo.com>

LOL!

Sorry guys.

Hahah. I fell over laughing too. And I was mad because of your resoponse in first place.

LOL.

Thank You all.

--- On Wed, 4/27/11, Brian West <brian at freeswitch.org> wrote:

From: Brian West <brian at freeswitch.org>
Subject: Re: [Freeswitch-users] custom kernel
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Wednesday, April 27, 2011, 1:14 PM

/me falls over laughing.
/b
On Apr 27, 2011, at 3:10 PM, Ognjen Seslija wrote:
Check the date.

On Wed, Apr 27, 2011 at 9:55 PM, Bojan Sukalo <shukalo83 at yahoo.com> wrote:

Hello.
Regarding this headline: "FreeSWITCH Adds Custom Libc And Kernel To
Code Repository"
Where I can download this custom kernel and libc code?




-----Inline Attachment Follows-----

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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 richocet2 at hotmail.com  Thu Apr 28 01:35:57 2011
From: richocet2 at hotmail.com (Dave Bracken)
Date: Wed, 27 Apr 2011 16:35:57 -0500
Subject: [Freeswitch-users] Holding properly on "outbound" socket
Message-ID: <SNT144-w5316D253A8F7CB0E335F5F0980@phx.gbl>


This is my scenario, i have a caller call in, and i bridge them to an operator. The operator would need to put the caller on hold(with music) and then the operator freed up for other calls,etc. Then the operator can pull the caller back off hold.
I need to know how to do this using the commands on an "outbound" socket connection to fs. What the commands are, the order, etc. Cant seem to find anyone who knows this yet, so i hope one of you fs gurus can point me in the right direction.
Thanks in advance,
Dave
in irc aka FSnewbDelphi.
  		 	   		  
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From eagle.antonio at gmail.com  Thu Apr 28 01:36:46 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Wed, 27 Apr 2011 22:36:46 +0100
Subject: [Freeswitch-users] custom kernel
In-Reply-To: <644050.54425.qm@web38105.mail.mud.yahoo.com>
References: <2341890C-1252-423F-9A7B-CE2FEB621B03@freeswitch.org>
	<644050.54425.qm@web38105.mail.mud.yahoo.com>
Message-ID: <BANLkTinBGymsG_B=B32EWaVepACyUb_HWQ@mail.gmail.com>

it's 1st of april ,

http://en.wikipedia.org/wiki/April_Fools%27_Day


2011/4/27 Bojan Sukalo <shukalo83 at yahoo.com>

> I don't get it.
>
> http://www.freeswitch.org/node/316 is some kind of announcement for custom
> kernel  2.6.11.41 in git repo. I've used git clone for free switch but
> couldn't seem to find it.
> As for date, it is Fri, 04/01/2011
>
> Maybe you fell over laughing but I still don't get it.
>
> Sorry.
>
> --- On *Wed, 4/27/11, Brian West <brian at freeswitch.org>* wrote:
>
>
> From: Brian West <brian at freeswitch.org>
> Subject: Re: [Freeswitch-users] custom kernel
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Wednesday, April 27, 2011, 1:14 PM
>
> /me falls over laughing.
>
> /b
>
> On Apr 27, 2011, at 3:10 PM, Ognjen Seslija wrote:
>
> Check the date.
>
> On Wed, Apr 27, 2011 at 9:55 PM, Bojan Sukalo <shukalo83 at yahoo.com<http://mc/compose?to=shukalo83 at yahoo.com>>
> wrote:
>
> Hello.
>
> Regarding this headline: "FreeSWITCH Adds Custom Libc And Kernel To
>
> Code Repository"
>
> Where I can download this custom kernel and libc code?
>
>
>
>
>
> -----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 fvillarroel at yahoo.com  Thu Apr 28 05:05:56 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Wed, 27 Apr 2011 18:05:56 -0700 (PDT)
Subject: [Freeswitch-users] Script para chequear un numero de destino
Message-ID: <758836.42920.qm@web34305.mail.mud.yahoo.com>

Hola a todos.

Voy a recibir trafico desde un Gateway A que mi FS enrutar? hacia un Gateway B. 

Hasta ahy todo bien, el asunto es que antes de enrutar la llamada hacia el gateway B de alguna manera debo consultar el numero de destino en una base de datos si es as? entonces se enruta la llamada en caso contrario la llamda la debo declinar de mi lado para que el Gateway A pueda enrutar esa llamada por otro proveedor distinto de mi.

Entonces lo que pienso hacer es un Script usando event_socket para que haga la consulta en la base de datos, entonces si el numero de destino esta la base de datos se enruta la llamada en caso contrario la declino para que la llamada continue por otro proveedor.

Mi tema es que no se me ocurre como hacer esto, imagino que  en el dialplan para que dependiendo de la existencia del numero de destino en la base de datos haga una llamada o en caso contrario haga otra cosa.

Agredecere aguna idea o sugerencia sobre como resolver este problema.

Con Asterisk usaria DeadAgi para pasarle el control a un script que consulte con la base de datos y me retorne true si puedo llamar o false en caso contario. Y digo DeadAgi ya que el control lo toma el script, ya que si usara Agi el control sigue en Asterisk independiente de lo que este haciendo el Script.

Aprovecho entonces de consultar con event_socket el control pasa al script o el dialplan continua ejecutandose.

Agradecere cualquier idea o sugerencia.

Saludos






From msc at freeswitch.org  Thu Apr 28 05:08:14 2011
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 27 Apr 2011 18:08:14 -0700
Subject: [Freeswitch-users] Holding properly on "outbound" socket
In-Reply-To: <SNT144-w5316D253A8F7CB0E335F5F0980@phx.gbl>
References: <SNT144-w5316D253A8F7CB0E335F5F0980@phx.gbl>
Message-ID: <BANLkTinXSfr09RfEeVDfEmHyt5cthW62DQ@mail.gmail.com>

What kind of phone is the operator using? Worst case scenario is you can use
valet_park app. I think we have examples on the wiki of how to use it. If
you run into any questions ping us back here or in IRC.

-MC (irc:mercutioviz)

On Wed, Apr 27, 2011 at 2:35 PM, Dave Bracken <richocet2 at hotmail.com> wrote:

>  This is my scenario, i have a caller call in, and i bridge them to an
> operator. The operator would need to put the caller on hold(with music) and
> then the operator freed up for other calls,etc. Then the operator can pull
> the caller back off hold.
> I need to know how to do this using the commands on an "outbound" socket
> connection to fs. What the commands are, the order, etc. Cant seem to find
> anyone who knows this yet, so i hope one of you fs gurus can point me in the
> right direction.
> Thanks in advance,
> Dave
> in irc aka FSnewbDelphi.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 Tim.Meade at millicorp.com  Thu Apr 28 05:15:05 2011
From: Tim.Meade at millicorp.com (Tim Meade)
Date: Thu, 28 Apr 2011 01:15:05 +0000
Subject: [Freeswitch-users] bypass_media=true and socket
In-Reply-To: <BANLkTi=OLO5GiDXcnYXGUHC3n_KpGRxbCQ@mail.gmail.com>
References: <6B85FABC3A341E4CA5D5D96AD9E4FE581745B9FE@Mailbox.millicorp.com>
	<1303915546268-6309705.post@n2.nabble.com>
	<BANLkTi=OLO5GiDXcnYXGUHC3n_KpGRxbCQ@mail.gmail.com>
Message-ID: <6B85FABC3A341E4CA5D5D96AD9E4FE5817464A53@Mailbox.millicorp.com>

We have identified the issue.  It was in the perl code.   So false alarm.   We do not have it working fully at this time,  but are VERY close.

Thanks for the quick response though everyone.



From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steven Ayre
Sent: Wednesday, April 27, 2011 2:00 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] bypass_media=true and socket

Jeff, socket is the outbound ESL.

Tim, it would be useful to know what you're sending via ESL and see debug-level logs with siptrace enabled so we can see what's going on.

-Steve

On 27 April 2011 15:45, Jeff Lenk <jeff at jefflenk.com<mailto:jeff at jefflenk.com>> wrote:
Socket is a custom app wrote by you?

What are you trying to accomplish with bypass media?

--
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From roger.castaldo at gmail.com  Thu Apr 28 05:24:02 2011
From: roger.castaldo at gmail.com (Roger Castaldo)
Date: Wed, 27 Apr 2011 21:24:02 -0400
Subject: [Freeswitch-users] Script para chequear un numero de destino
In-Reply-To: <758836.42920.qm@web34305.mail.mud.yahoo.com>
References: <758836.42920.qm@web34305.mail.mud.yahoo.com>
Message-ID: <BANLkTimUGwDwD40M1ujPOi7ETKT++1zFmQ@mail.gmail.com>

I ran your email through google translate and from what I can tell you want
freeswitch to route a call if the number exists in database X.  That being
said, the event socket is not the way to go, look at doing it via a lua
script call from the dial plan, then you can have the lua query the
database, if it exists, use the api to route the call, or set a flag for the
dial plan, otherwise it can hang up, or play a message and hang up which
ends the call and stops processing in your dial plan.

2011/4/27 FERNANDO VILLARROEL <fvillarroel at yahoo.com>

> Hola a todos.
>
> Voy a recibir trafico desde un Gateway A que mi FS enrutar? hacia un
> Gateway B.
>
> Hasta ahy todo bien, el asunto es que antes de enrutar la llamada hacia el
> gateway B de alguna manera debo consultar el numero de destino en una base
> de datos si es as? entonces se enruta la llamada en caso contrario la llamda
> la debo declinar de mi lado para que el Gateway A pueda enrutar esa llamada
> por otro proveedor distinto de mi.
>
> Entonces lo que pienso hacer es un Script usando event_socket para que haga
> la consulta en la base de datos, entonces si el numero de destino esta la
> base de datos se enruta la llamada en caso contrario la declino para que la
> llamada continue por otro proveedor.
>
> Mi tema es que no se me ocurre como hacer esto, imagino que  en el dialplan
> para que dependiendo de la existencia del numero de destino en la base de
> datos haga una llamada o en caso contrario haga otra cosa.
>
> Agredecere aguna idea o sugerencia sobre como resolver este problema.
>
> Con Asterisk usaria DeadAgi para pasarle el control a un script que
> consulte con la base de datos y me retorne true si puedo llamar o false en
> caso contario. Y digo DeadAgi ya que el control lo toma el script, ya que si
> usara Agi el control sigue en Asterisk independiente de lo que este haciendo
> el Script.
>
> Aprovecho entonces de consultar con event_socket el control pasa al script
> o el dialplan continua ejecutandose.
>
> Agradecere cualquier idea o sugerencia.
>
> Saludos
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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  Thu Apr 28 05:34:22 2011
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 27 Apr 2011 20:34:22 -0500
Subject: [Freeswitch-users] Script para chequear un numero de destino
In-Reply-To: <BANLkTimUGwDwD40M1ujPOi7ETKT++1zFmQ@mail.gmail.com>
Message-ID: <C9DE2E4E.148D2%krice@freeswitch.org>

Routing a call if a number exists in a database...  Hmmm sounds suspiciously
a lot like mod_easyroute

k


On 4/27/11 8:24 PM, "Roger Castaldo" <roger.castaldo at gmail.com> wrote:

> I ran your email through google translate and from what I can tell you want
> freeswitch to route a call if the number exists in database X.? That being
> said, the event socket is not the way to go, look at doing it via a lua script
> call from the dial plan, then you can have the lua query the database, if it
> exists, use the api to route the call, or set a flag for the dial plan,
> otherwise it can hang up, or play a message and hang up which ends the call
> and stops processing in your dial plan.
> 
> 2011/4/27 FERNANDO VILLARROEL <fvillarroel at yahoo.com>
>> Hola a todos.
>> 
>> Voy a recibir trafico desde un Gateway A que mi FS enrutar? hacia un Gateway
>> B.
>> 
>> Hasta ahy todo bien, el asunto es que antes de enrutar la llamada hacia el
>> gateway B de alguna manera debo consultar el numero de destino en una base de
>> datos si es as? entonces se enruta la llamada en caso contrario la llamda la
>> debo declinar de mi lado para que el Gateway A pueda enrutar esa llamada por
>> otro proveedor distinto de mi.
>> 
>> Entonces lo que pienso hacer es un Script usando event_socket para que haga
>> la consulta en la base de datos, entonces si el numero de destino esta la
>> base de datos se enruta la llamada en caso contrario la declino para que la
>> llamada continue por otro proveedor.
>> 
>> Mi tema es que no se me ocurre como hacer esto, imagino que ?en el dialplan
>> para que dependiendo de la existencia del numero de destino en la base de
>> datos haga una llamada o en caso contrario haga otra cosa.
>> 
>> Agredecere aguna idea o sugerencia sobre como resolver este problema.
>> 
>> Con Asterisk usaria DeadAgi para pasarle el control a un script que consulte
>> con la base de datos y me retorne true si puedo llamar o false en caso
>> contario. Y digo DeadAgi ya que el control lo toma el script, ya que si usara
>> Agi el control sigue en Asterisk independiente de lo que este haciendo el
>> Script.
>> 
>> Aprovecho entonces de consultar con event_socket el control pasa al script o
>> el dialplan continua ejecutandose.
>> 
>> Agradecere cualquier idea o sugerencia.
>> 
>> Saludos
>> 
>> 
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 jmesquita at freeswitch.org  Thu Apr 28 06:36:07 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 27 Apr 2011 23:36:07 -0300
Subject: [Freeswitch-users] Script para chequear un numero de destino
In-Reply-To: <C9DE2E4E.148D2%krice@freeswitch.org>
References: <BANLkTimUGwDwD40M1ujPOi7ETKT++1zFmQ@mail.gmail.com>
	<C9DE2E4E.148D2%krice@freeswitch.org>
Message-ID: <BANLkTi=Lz=gBRrcX_WmYN3iZXhzBv4wSPA@mail.gmail.com>

Hola Fernando,

Yo mas o menos hablo espa?ol as? que trato de darte una mano. Ten?s muchas
opciones. mod_easyroute como lo menciona Ken es una, ESL (event_socket) es
otra, xml_curl es otra y uno de los "bindings" como lua, python, perl,
etc... es otra.

Lo que tenemos que encontrar es lo que a vos te es mas comodo. Program?s? En
que lenguaje? Si no program?s, mod_easyroute seguramente ser? lo mejor. ESL
es lo m?s complicadito de todos ya que el dialplan entero lo tenes que
programar desde tu socket y tirar los comandos hacia freeswitch en un orden
especifico. XML_CURL es muy facil si sabes desarrollar en web.

Enfim, espero noticias tuyas para que te pueda ayudar con un determinado
camino.

Saludos,

Jo?o Mesquita



2011/4/27 Ken Rice <krice at freeswitch.org>

>  Routing a call if a number exists in a database...  Hmmm sounds
> suspiciously a lot like mod_easyroute
>
> k
>
>
>
> On 4/27/11 8:24 PM, "Roger Castaldo" <roger.castaldo at gmail.com> wrote:
>
> I ran your email through google translate and from what I can tell you want
> freeswitch to route a call if the number exists in database X.  That being
> said, the event socket is not the way to go, look at doing it via a lua
> script call from the dial plan, then you can have the lua query the
> database, if it exists, use the api to route the call, or set a flag for the
> dial plan, otherwise it can hang up, or play a message and hang up which
> ends the call and stops processing in your dial plan.
>
> 2011/4/27 FERNANDO VILLARROEL <fvillarroel at yahoo.com>
>
> Hola a todos.
>
> Voy a recibir trafico desde un Gateway A que mi FS enrutar? hacia un
> Gateway B.
>
> Hasta ahy todo bien, el asunto es que antes de enrutar la llamada hacia el
> gateway B de alguna manera debo consultar el numero de destino en una base
> de datos si es as? entonces se enruta la llamada en caso contrario la llamda
> la debo declinar de mi lado para que el Gateway A pueda enrutar esa llamada
> por otro proveedor distinto de mi.
>
> Entonces lo que pienso hacer es un Script usando event_socket para que haga
> la consulta en la base de datos, entonces si el numero de destino esta la
> base de datos se enruta la llamada en caso contrario la declino para que la
> llamada continue por otro proveedor.
>
> Mi tema es que no se me ocurre como hacer esto, imagino que  en el dialplan
> para que dependiendo de la existencia del numero de destino en la base de
> datos haga una llamada o en caso contrario haga otra cosa.
>
> Agredecere aguna idea o sugerencia sobre como resolver este problema.
>
> Con Asterisk usaria DeadAgi para pasarle el control a un script que
> consulte con la base de datos y me retorne true si puedo llamar o false en
> caso contario. Y digo DeadAgi ya que el control lo toma el script, ya que si
> usara Agi el control sigue en Asterisk independiente de lo que este haciendo
> el Script.
>
> Aprovecho entonces de consultar con event_socket el control pasa al script
> o el dialplan continua ejecutandose.
>
> Agradecere cualquier idea o sugerencia.
>
> Saludos
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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
> 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 krice at freeswitch.org  Thu Apr 28 06:47:47 2011
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 27 Apr 2011 21:47:47 -0500
Subject: [Freeswitch-users] Script para chequear un numero de destino
In-Reply-To: <BANLkTi=Lz=gBRrcX_WmYN3iZXhzBv4wSPA@mail.gmail.com>
Message-ID: <C9DE3F83.148D7%krice@freeswitch.org>

Hey Mesquita!

I was just speaking with Fernando on IRC... I suggested he look at
http://wiki.freeswitch.org/wiki/Mod_odbc_query. This mixed with a little
dialplan might be just what he needs. Appears he might be trying to only
pass calls to say wireless destinations. ?Mi espa?ol es no bueno? tho so any
help on the language barrier could prove useful for him.

Thanks!


On 4/27/11 9:36 PM, "Jo?o Mesquita" <jmesquita at freeswitch.org> wrote:

> Hola Fernando,
> 
> Yo mas o menos hablo espa?ol as? que trato de darte una mano. Ten?s muchas
> opciones. mod_easyroute como lo menciona Ken es una, ESL (event_socket) es
> otra, xml_curl es otra y uno de los "bindings" como lua, python, perl, etc...
> es otra.
> 
> Lo que tenemos que encontrar es lo que a vos te es mas comodo. Program?s? En
> que lenguaje? Si no program?s, mod_easyroute seguramente ser? lo mejor. ESL es
> lo m?s complicadito de todos ya que el dialplan entero lo tenes que programar
> desde tu socket y tirar los comandos hacia freeswitch en un orden especifico.
> XML_CURL es muy facil si sabes desarrollar en web.
> 
> Enfim, espero noticias tuyas para que te pueda ayudar con un determinado
> camino.
> 
> Saludos,
> 
> Jo?o Mesquita
> 
> 
> 
> 2011/4/27 Ken Rice <krice at freeswitch.org>
>> Routing a call if a number exists in a database... ?Hmmm sounds suspiciously
>> a lot like mod_easyroute
>> 
>> k
>> 
>> 
>> 
>> On 4/27/11 8:24 PM, "Roger Castaldo" <roger.castaldo at gmail.com
>> <http://roger.castaldo at gmail.com> > wrote:
>> 
>>> I ran your email through google translate and from what I can tell you want
>>> freeswitch to route a call if the number exists in database X.? That being
>>> said, the event socket is not the way to go, look at doing it via a lua
>>> script call from the dial plan, then you can have the lua query the
>>> database, if it exists, use the api to route the call, or set a flag for the
>>> dial plan, otherwise it can hang up, or play a message and hang up which
>>> ends the call and stops processing in your dial plan.
>>> 
>>> 2011/4/27 FERNANDO VILLARROEL <fvillarroel at yahoo.com
>>> <http://fvillarroel at yahoo.com> >
>>>> Hola a todos.
>>>> 
>>>> Voy a recibir trafico desde un Gateway A que mi FS enrutar? hacia un
>>>> Gateway B.
>>>> 
>>>> Hasta ahy todo bien, el asunto es que antes de enrutar la llamada hacia el
>>>> gateway B de alguna manera debo consultar el numero de destino en una base
>>>> de datos si es as? entonces se enruta la llamada en caso contrario la
>>>> llamda la debo declinar de mi lado para que el Gateway A pueda enrutar esa
>>>> llamada por otro proveedor distinto de mi.
>>>> 
>>>> Entonces lo que pienso hacer es un Script usando event_socket para que haga
>>>> la consulta en la base de datos, entonces si el numero de destino esta la
>>>> base de datos se enruta la llamada en caso contrario la declino para que la
>>>> llamada continue por otro proveedor.
>>>> 
>>>> Mi tema es que no se me ocurre como hacer esto, imagino que ?en el dialplan
>>>> para que dependiendo de la existencia del numero de destino en la base de
>>>> datos haga una llamada o en caso contrario haga otra cosa.
>>>> 
>>>> Agredecere aguna idea o sugerencia sobre como resolver este problema.
>>>> 
>>>> Con Asterisk usaria DeadAgi para pasarle el control a un script que
>>>> consulte con la base de datos y me retorne true si puedo llamar o false en
>>>> caso contario. Y digo DeadAgi ya que el control lo toma el script, ya que
>>>> si usara Agi el control sigue en Asterisk independiente de lo que este
>>>> haciendo el Script.
>>>> 
>>>> Aprovecho entonces de consultar con event_socket el control pasa al script
>>>> o el dialplan continua ejecutandose.
>>>> 
>>>> Agradecere cualquier idea o sugerencia.
>>>> 
>>>> Saludos
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> 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://FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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  Thu Apr 28 07:09:02 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Wed, 27 Apr 2011 20:09:02 -0700 (PDT)
Subject: [Freeswitch-users] Script para chequear un numero de destino
In-Reply-To: <BANLkTi=Lz=gBRrcX_WmYN3iZXhzBv4wSPA@mail.gmail.com>
Message-ID: <968175.88614.qm@web34306.mail.mud.yahoo.com>

Hola Jo?o.

Si programo, mi lenguaje preferido es Python yo vengo de Asterisk donde usaba Python para programar AGI.

En web algo me manejo tambien por lo que creo no tendre problemas, le he metido mano a Django y Web2py.

Si no fuera mucho pedirte me gustaria aprender un poco de todo asi decido luego como lo voy haciendo, ESL me llama la atenci?n ya que me gustaria comenzar a aprehender a usarlo, ya sabes capturar eventos de FS, enviar eventos a FS, etc.

Pero bueno por ahora si me ayudas a resolver mi problema usando lo mas sencillo te lo agradeceria mucho.

El problema es que debo crear una tabla con aproximadamente 4500 breackouts de numeros moviles, entonces si me entra una llamada con destination_number = 12345678 debo consultar si los 3 primeros dijitos estan presentes en la tabla si es si entonces reenvio la llamada a un Gateway y si no entonces la declino para que mi cliente la enrute por otro proveedor.

Saludos.




--- On Wed, 4/27/11, Jo?o Mesquita <jmesquita at freeswitch.org> wrote:

From: Jo?o Mesquita <jmesquita at freeswitch.org>
Subject: Re: [Freeswitch-users] Script para chequear un numero de destino
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Date: Wednesday, April 27, 2011, 11:36 PM

Hola Fernando,

Yo mas o menos hablo espa?ol as? que trato de darte una mano. Ten?s muchas opciones. mod_easyroute como lo menciona Ken es una, ESL (event_socket) es otra, xml_curl es otra y uno de los "bindings" como lua, python, perl, etc... es otra.


Lo que tenemos que encontrar es lo que a vos te es mas comodo. Program?s? En que lenguaje? Si no program?s, mod_easyroute seguramente ser? lo mejor. ESL es lo m?s complicadito de todos ya que el dialplan entero lo tenes que programar desde tu socket y tirar los comandos hacia freeswitch en un orden especifico. XML_CURL es muy facil si sabes desarrollar en web.


Enfim, espero noticias tuyas para que te pueda ayudar con un determinado camino.

Saludos,

Jo?o Mesquita




2011/4/27 Ken Rice <krice at freeswitch.org>






Routing a call if a number exists in a database... ?Hmmm sounds suspiciously a lot like mod_easyroute



k





On 4/27/11 8:24 PM, "Roger Castaldo" <roger.castaldo at gmail.com> wrote:



I ran your email through google translate and from what I can tell you want freeswitch to route a call if the number exists in database X.? That being said, the event socket is not the way to go, look at doing it via a lua script call from the dial plan, then you can have the lua query the database, if it exists, use the api to route the call, or set a flag for the dial plan, otherwise it can hang up, or play a message and hang up which ends the call and stops processing in your dial plan.




2011/4/27 FERNANDO VILLARROEL <fvillarroel at yahoo.com>

Hola a todos.



Voy a recibir trafico desde un Gateway A que mi FS enrutar? hacia un Gateway B.



Hasta ahy todo bien, el asunto es que antes de enrutar la llamada hacia el gateway B de alguna manera debo consultar el numero de destino en una base de datos si es as? entonces se enruta la llamada en caso contrario la llamda la debo declinar de mi lado para que el Gateway A pueda enrutar esa llamada por otro proveedor distinto de mi.




Entonces lo que pienso hacer es un Script usando event_socket para que haga la consulta en la base de datos, entonces si el numero de destino esta la base de datos se enruta la llamada en caso contrario la declino para que la llamada continue por otro proveedor.




Mi tema es que no se me ocurre como hacer esto, imagino que ?en el dialplan para que dependiendo de la existencia del numero de destino en la base de datos haga una llamada o en caso contrario haga otra cosa.



Agredecere aguna idea o sugerencia sobre como resolver este problema.



Con Asterisk usaria DeadAgi para pasarle el control a un script que consulte con la base de datos y me retorne true si puedo llamar o false en caso contario. Y digo DeadAgi ya que el control lo toma el script, ya que si usara Agi el control sigue en Asterisk independiente de lo que este haciendo el Script.




Aprovecho entonces de consultar con event_socket el control pasa al script o el dialplan continua ejecutandose.



Agradecere cualquier idea o sugerencia.



Saludos











_______________________________________________

FreeSWITCH-users mailing list

FreeSWITCH-users at 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

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

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From jmesquita at freeswitch.org  Thu Apr 28 07:37:22 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 28 Apr 2011 00:37:22 -0300
Subject: [Freeswitch-users] Script para chequear un numero de destino
In-Reply-To: <968175.88614.qm@web34306.mail.mud.yahoo.com>
References: <BANLkTi=Lz=gBRrcX_WmYN3iZXhzBv4wSPA@mail.gmail.com>
	<968175.88614.qm@web34306.mail.mud.yahoo.com>
Message-ID: <BANLkTi=ZFv8D66r2SmV5VXQOvKP1uKX9pQ@mail.gmail.com>

Hey Ken! Even tho spanish is not my native language, I live in Argentina, so
I can pretty much get away with it. Any help on portuguese or spanish that
this list may require, I am available to help.

Fernando,

La manera mas rapida es algo relativo. Depende mucho de con que te sentis
mas comodo. Me gusta mucho ESL y tambi?n me gusta mucho xml_curl. odbc_query
nunca lo use, pero parece sencillo.

A mi me encanta Python tambien y suelo usarlo con ESL. Uso bastante tambi?n
Pylons pero ya trabaje con Django tambien. Sea como for, lo que vos
necesit?s si vas a usar ESL es un outbound socket. Te recomiendo que
empiezes leyendo esta pagina para que te vayas enterando de que se trata:
http://wiki.freeswitch.org/wiki/Event_Socket_Outbound

Cuando termines de leer, yo te puedo tirar unos "tips" de como implementar
lo que necesit?s en Python sin demasiados problemas.

Saludos,
Jo?o Mesquita



2011/4/28 FERNANDO VILLARROEL <fvillarroel at yahoo.com>

> Hola Jo?o.
>
> Si programo, mi lenguaje preferido es Python yo vengo de Asterisk donde
> usaba Python para programar AGI.
>
> En web algo me manejo tambien por lo que creo no tendre problemas, le he
> metido mano a Django y Web2py.
>
> Si no fuera mucho pedirte me gustaria aprender un poco de todo asi decido
> luego como lo voy haciendo, ESL me llama la atenci?n ya que me gustaria
> comenzar a aprehender a usarlo, ya sabes capturar eventos de FS, enviar
> eventos a FS, etc.
>
> Pero bueno por ahora si me ayudas a resolver mi problema usando lo mas
> sencillo te lo agradeceria mucho.
>
> El problema es que debo crear una tabla con aproximadamente 4500 breackouts
> de numeros moviles, entonces si me entra una llamada con destination_number
> = 12345678 debo consultar si los 3 primeros dijitos estan presentes en la
> tabla si es si entonces reenvio la llamada a un Gateway y si no entonces la
> declino para que mi cliente la enrute por otro proveedor.
>
> Saludos.
>
>
>
>
> --- On *Wed, 4/27/11, Jo?o Mesquita <jmesquita at freeswitch.org>* wrote:
>
>
> From: Jo?o Mesquita <jmesquita at freeswitch.org>
> Subject: Re: [Freeswitch-users] Script para chequear un numero de destino
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Wednesday, April 27, 2011, 11:36 PM
>
>
> Hola Fernando,
>
> Yo mas o menos hablo espa?ol as? que trato de darte una mano. Ten?s muchas
> opciones. mod_easyroute como lo menciona Ken es una, ESL (event_socket) es
> otra, xml_curl es otra y uno de los "bindings" como lua, python, perl,
> etc... es otra.
>
> Lo que tenemos que encontrar es lo que a vos te es mas comodo. Program?s?
> En que lenguaje? Si no program?s, mod_easyroute seguramente ser? lo mejor.
> ESL es lo m?s complicadito de todos ya que el dialplan entero lo tenes que
> programar desde tu socket y tirar los comandos hacia freeswitch en un orden
> especifico. XML_CURL es muy facil si sabes desarrollar en web.
>
> Enfim, espero noticias tuyas para que te pueda ayudar con un determinado
> camino.
>
> Saludos,
>
> Jo?o Mesquita
>
>
>
> 2011/4/27 Ken Rice <krice at freeswitch.org<http://mc/compose?to=krice at freeswitch.org>
> >
>
>  Routing a call if a number exists in a database...  Hmmm sounds
> suspiciously a lot like mod_easyroute
>
> k
>
>
>
> On 4/27/11 8:24 PM, "Roger Castaldo" <roger.castaldo at gmail.com> wrote:
>
> I ran your email through google translate and from what I can tell you want
> freeswitch to route a call if the number exists in database X.  That being
> said, the event socket is not the way to go, look at doing it via a lua
> script call from the dial plan, then you can have the lua query the
> database, if it exists, use the api to route the call, or set a flag for the
> dial plan, otherwise it can hang up, or play a message and hang up which
> ends the call and stops processing in your dial plan.
>
> 2011/4/27 FERNANDO VILLARROEL <fvillarroel at yahoo.com>
>
> Hola a todos.
>
> Voy a recibir trafico desde un Gateway A que mi FS enrutar? hacia un
> Gateway B.
>
> Hasta ahy todo bien, el asunto es que antes de enrutar la llamada hacia el
> gateway B de alguna manera debo consultar el numero de destino en una base
> de datos si es as? entonces se enruta la llamada en caso contrario la llamda
> la debo declinar de mi lado para que el Gateway A pueda enrutar esa llamada
> por otro proveedor distinto de mi.
>
> Entonces lo que pienso hacer es un Script usando event_socket para que haga
> la consulta en la base de datos, entonces si el numero de destino esta la
> base de datos se enruta la llamada en caso contrario la declino para que la
> llamada continue por otro proveedor.
>
> Mi tema es que no se me ocurre como hacer esto, imagino que  en el dialplan
> para que dependiendo de la existencia del numero de destino en la base de
> datos haga una llamada o en caso contrario haga otra cosa.
>
> Agredecere aguna idea o sugerencia sobre como resolver este problema.
>
> Con Asterisk usaria DeadAgi para pasarle el control a un script que
> consulte con la base de datos y me retorne true si puedo llamar o false en
> caso contario. Y digo DeadAgi ya que el control lo toma el script, ya que si
> usara Agi el control sigue en Asterisk independiente de lo que este haciendo
> el Script.
>
> Aprovecho entonces de consultar con event_socket el control pasa al script
> o el dialplan continua ejecutandose.
>
> Agradecere cualquier idea o sugerencia.
>
> Saludos
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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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>
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>
>
>
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>
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From anton.vazir at gmail.com  Thu Apr 28 11:38:12 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Thu, 28 Apr 2011 12:38:12 +0500
Subject: [Freeswitch-users] SIP external profile - Error looking up host name
Message-ID: <BANLkTinaJ66c3VS-rM7GjUZvhSh_wQ=aFg@mail.gmail.com>

Looks like a nasty bug.
Until I have put a gateway IP in realm sip_profiles/external - FS
gives a DNS error and will not register
Freeswitch  - fresh GIT

At the same time I can resolve a dns record via nslookup
(note real domain name was replaced by name.tld and real IP by X.X.X.)

freeswitch at host1> version
FreeSWITCH Version 1.0.head (git-adffe07 2011-04-23 23-52-03 -0400)


host1# nslookup
> set type=any
> name.tld
Server:         X.X.X.X
Address:        X.X.X.X#53

name.tld
        origin = name.tld
        mail addr = anton.name.tld
        serial = 9110229
        refresh = 10800
        retry = 3600
        expire = 259200
        minimum = 86400
Name:   name.tld
Address: X.X.X.33
name.tld  nameserver = ns1.name.tld
name.tld  nameserver = ns2.name.tld

> _sip._udp.name.tld
Server:         X.X.X.7
Address:        X.X.X.7#53

_sip._udp.name.tld        service = 0 1 5060 X.X.X.33.name.tld.
>

tcpdump for FS requests gives the following

12:24:12.135842 IP 192.168.100.12.34358 > X.X.X.X.53: 20007+ SRV?
_sip._udp.name.tld. (34)
12:24:12.136423 IP X.X.X.X.53 > 192.168.100.12.34358: 20007* 1/2/3 SRV
X.X.X.X.name.tld.:5060 0 1 (170)
12:24:12.136505 IP 192.168.100.12.34358 > X.X.X.X.53: 20263+ A? name.tld. (24)
12:24:12.136995 IP X.X.X.X.53 > 192.168.100.12.34358: 20263* 1/2/2 A
X.X.X.X (120)

Fs sofia debug gives the following:


freeswitch at host1> sofia profile external register name.tld

+OK

freeswitch at skype1> nua: nh_create_handle: entering
nua: nua_handle_bind: entering
2011-04-28 12:25:53.145595 [NOTICE] sofia_reg.c:367 Registering name.tld
nua: nua_register: entering
nua(0x169afe0): sent signal r_register
nua(0x169afe0): recv signal r_register
nua: nua_stack_set_params: entering
soa_clone(static::0x1669f30, 0x1659f50, 0x169afe0) called
soa_set_params(static::0x7f4330030580, ...) called
soa_set_params(static::0x7f4330030580, ...) called
nua(0x169afe0): adding register usage
nta_leg_tcreate(0x7f4338036980)
nta: selecting scheme sip
sres_cache_get(0x7f433000ac10, SRV, "_sip._udp.name.tld.") called
nta: for "name.tld" query "_sip._udp.name.tld" SRV
sres_query(0x7f433002ed40, 0x7f4338035d20, SRV, "_sip._udp.name.tld") called
sres_send_dns_query(0x7f433002ed40, 0x7f433800aa40) called
sres_send_dns_query(0x7f433002ed40, 0x7f433800aa40) id=10068 SRV
_sip._udp.name.tld (to [X.X.X.7]:53)
sres_resolver_receive(0x7f433002ed40, 39) called
ANSWER RR received . 8448? 448? 201335040 rdlen=256
sres_create_record: truncated message
sres_create_record: truncated message
sres_decode_msg: got 0 but expected errors=0 an=1 ar=3 ns=2
sres_resolver_receive(0x7f433002ed40, 0x7f433800aa40) id=10068 (from
[X.X.X.7]:53)
sres(q=0x7f433800aa40): reporting error NETWORK_ERR for SRV _sip._udp.name.tld
sres_cache_get(0x7f433000ac10, A, "name.tld.") called
nta: for "name.tld" query "name.tld" A
sres_query(0x7f433002ed40, 0x7f4338035d20, A, "name.tld") called
sres_send_dns_query(0x7f433002ed40, 0x7f433802a770) called
sres_send_dns_query(0x7f433002ed40, 0x7f433802a770) id=10069 A
name.tld (to [X.X.X.7]:53)
sres_resolver_receive(0x7f433002ed40, 39) called
ANSWER RR received . 256? 448? 201326848 rdlen=256
sres_create_record: truncated message
sres_create_record: truncated message
sres_decode_msg: got 0 but expected errors=0 an=1 ar=2 ns=2
sres_resolver_receive(0x7f433002ed40, 0x7f433802a770) id=10069 (from
[X.X.X.7]:53)
sres(q=0x7f433802a770): reporting error NETWORK_ERR for A name.tld
nta: timer set to 32000 ms
nta: timer shortened to 5000 ms
nua(0x169afe0): event r_register 503 DNS Error
nua(0x169afe0): removing register usage
nta_leg_destroy(0x7f4338036980)
nua: nua_application_event: entering
nua: nua_handle_destroy: entering
nua(0x169afe0): sent signal r_destroy
2011-04-28 12:25:53.146606 [ERR] sofia_reg.c:1771 name.tld
Registration Failed with status DNS Error [503]. failure #11
nua(0x169afe0): recv signal r_destroy
nta_leg_destroy((nil))
soa_destroy(static::0x7f4330030580) called
nua: nua_handle_magic: entering
nua: nua_handle_bind: entering
nua: nua_handle_destroy: entering
2011-04-28 12:25:54.145598 [WARNING] sofia_reg.c:425 name.tld Failed
Registration [0], setting retry to 30 seconds.
nta: timer K fired, terminate REGISTER (11651712)
outgoing_reclaim_all((nil), (nil), 0x7f433e970cf0)
nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free
nta: timer not set


From eagle.antonio at gmail.com  Thu Apr 28 12:58:59 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Thu, 28 Apr 2011 08:58:59 +0000
Subject: [Freeswitch-users] ESL & DTMF
Message-ID: <BANLkTimPUQC2USHkQ9b72jAV11Sms2KWkA@mail.gmail.com>

Good Morning.

I'm currently in the process of converting my IVR's into ESL , freedom :D.
>From my testing is extremely flexible until i have reached a wall.

We use TTS to generate alot of information for our users then we need to
collect some answers, in session i had GetDigists , yay :).
In ESL i have the DTMF event the problem is that from my testing and wiki ,
DTMF event is only send down the socket using asycn , now i need "sync"
because the IVRs need to understand the exact position of the user and blah
blah blah ,  need to be dynamically extended in realtime and would be hard
to implement in async , specially the timing for digit collection.

So i tried capturing the event with no luck the only one i got is
SERVER_DISCONNECT at the end.

I think It is possible if for every thread i would raise a listener for that
specific uuid that would share memory with the first one so i could read
that DTMF events that arrives trough the other socket but i think this is
way to much duct tape for a simple thing.

So any help on this ?
Thanks
A/T
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From juanito1982 at gmail.com  Thu Apr 28 13:33:40 2011
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Thu, 28 Apr 2011 11:33:40 +0200
Subject: [Freeswitch-users] Script para chequear un numero de destino
In-Reply-To: <BANLkTi=ZFv8D66r2SmV5VXQOvKP1uKX9pQ@mail.gmail.com>
References: <BANLkTi=Lz=gBRrcX_WmYN3iZXhzBv4wSPA@mail.gmail.com>
	<968175.88614.qm@web34306.mail.mud.yahoo.com>
	<BANLkTi=ZFv8D66r2SmV5VXQOvKP1uKX9pQ@mail.gmail.com>
Message-ID: <BANLkTimfcULSHkf9Zo5k+VugLS+FAKzHWg@mail.gmail.com>

Now you are talking about ESL? Which scenarios are suitable for ESL? When
would be nice to use ESL and when LUA?

Regards

2011/4/28 Jo?o Mesquita <jmesquita at freeswitch.org>

> Hey Ken! Even tho spanish is not my native language, I live in Argentina,
> so I can pretty much get away with it. Any help on portuguese or spanish
> that this list may require, I am available to help.
>
> Fernando,
>
> La manera mas rapida es algo relativo. Depende mucho de con que te sentis
> mas comodo. Me gusta mucho ESL y tambi?n me gusta mucho xml_curl. odbc_query
> nunca lo use, pero parece sencillo.
>
> A mi me encanta Python tambien y suelo usarlo con ESL. Uso bastante tambi?n
> Pylons pero ya trabaje con Django tambien. Sea como for, lo que vos
> necesit?s si vas a usar ESL es un outbound socket. Te recomiendo que
> empiezes leyendo esta pagina para que te vayas enterando de que se trata:
> http://wiki.freeswitch.org/wiki/Event_Socket_Outbound
>
> Cuando termines de leer, yo te puedo tirar unos "tips" de como implementar
> lo que necesit?s en Python sin demasiados problemas.
>
> Saludos,
> Jo?o Mesquita
>
>
>
>
> 2011/4/28 FERNANDO VILLARROEL <fvillarroel at yahoo.com>
>
>> Hola Jo?o.
>>
>> Si programo, mi lenguaje preferido es Python yo vengo de Asterisk donde
>> usaba Python para programar AGI.
>>
>> En web algo me manejo tambien por lo que creo no tendre problemas, le he
>> metido mano a Django y Web2py.
>>
>> Si no fuera mucho pedirte me gustaria aprender un poco de todo asi decido
>> luego como lo voy haciendo, ESL me llama la atenci?n ya que me gustaria
>> comenzar a aprehender a usarlo, ya sabes capturar eventos de FS, enviar
>> eventos a FS, etc.
>>
>> Pero bueno por ahora si me ayudas a resolver mi problema usando lo mas
>> sencillo te lo agradeceria mucho.
>>
>> El problema es que debo crear una tabla con aproximadamente 4500
>> breackouts de numeros moviles, entonces si me entra una llamada con
>> destination_number = 12345678 debo consultar si los 3 primeros dijitos estan
>> presentes en la tabla si es si entonces reenvio la llamada a un Gateway y si
>> no entonces la declino para que mi cliente la enrute por otro proveedor.
>>
>> Saludos.
>>
>>
>>
>>
>> --- On *Wed, 4/27/11, Jo?o Mesquita <jmesquita at freeswitch.org>* wrote:
>>
>>
>> From: Jo?o Mesquita <jmesquita at freeswitch.org>
>> Subject: Re: [Freeswitch-users] Script para chequear un numero de destino
>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>> Date: Wednesday, April 27, 2011, 11:36 PM
>>
>>
>> Hola Fernando,
>>
>> Yo mas o menos hablo espa?ol as? que trato de darte una mano. Ten?s muchas
>> opciones. mod_easyroute como lo menciona Ken es una, ESL (event_socket) es
>> otra, xml_curl es otra y uno de los "bindings" como lua, python, perl,
>> etc... es otra.
>>
>> Lo que tenemos que encontrar es lo que a vos te es mas comodo. Program?s?
>> En que lenguaje? Si no program?s, mod_easyroute seguramente ser? lo mejor.
>> ESL es lo m?s complicadito de todos ya que el dialplan entero lo tenes que
>> programar desde tu socket y tirar los comandos hacia freeswitch en un orden
>> especifico. XML_CURL es muy facil si sabes desarrollar en web.
>>
>> Enfim, espero noticias tuyas para que te pueda ayudar con un determinado
>> camino.
>>
>> Saludos,
>>
>> Jo?o Mesquita
>>
>>
>>
>> 2011/4/27 Ken Rice <krice at freeswitch.org<http://mc/compose?to=krice at freeswitch.org>
>> >
>>
>>  Routing a call if a number exists in a database...  Hmmm sounds
>> suspiciously a lot like mod_easyroute
>>
>> k
>>
>>
>>
>> On 4/27/11 8:24 PM, "Roger Castaldo" <roger.castaldo at gmail.com> wrote:
>>
>> I ran your email through google translate and from what I can tell you
>> want freeswitch to route a call if the number exists in database X.  That
>> being said, the event socket is not the way to go, look at doing it via a
>> lua script call from the dial plan, then you can have the lua query the
>> database, if it exists, use the api to route the call, or set a flag for the
>> dial plan, otherwise it can hang up, or play a message and hang up which
>> ends the call and stops processing in your dial plan.
>>
>> 2011/4/27 FERNANDO VILLARROEL <fvillarroel at yahoo.com>
>>
>> Hola a todos.
>>
>> Voy a recibir trafico desde un Gateway A que mi FS enrutar? hacia un
>> Gateway B.
>>
>> Hasta ahy todo bien, el asunto es que antes de enrutar la llamada hacia el
>> gateway B de alguna manera debo consultar el numero de destino en una base
>> de datos si es as? entonces se enruta la llamada en caso contrario la llamda
>> la debo declinar de mi lado para que el Gateway A pueda enrutar esa llamada
>> por otro proveedor distinto de mi.
>>
>> Entonces lo que pienso hacer es un Script usando event_socket para que
>> haga la consulta en la base de datos, entonces si el numero de destino esta
>> la base de datos se enruta la llamada en caso contrario la declino para que
>> la llamada continue por otro proveedor.
>>
>> Mi tema es que no se me ocurre como hacer esto, imagino que  en el
>> dialplan para que dependiendo de la existencia del numero de destino en la
>> base de datos haga una llamada o en caso contrario haga otra cosa.
>>
>> Agredecere aguna idea o sugerencia sobre como resolver este problema.
>>
>> Con Asterisk usaria DeadAgi para pasarle el control a un script que
>> consulte con la base de datos y me retorne true si puedo llamar o false en
>> caso contario. Y digo DeadAgi ya que el control lo toma el script, ya que si
>> usara Agi el control sigue en Asterisk independiente de lo que este haciendo
>> el Script.
>>
>> Aprovecho entonces de consultar con event_socket el control pasa al script
>> o el dialplan continua ejecutandose.
>>
>> Agradecere cualquier idea o sugerencia.
>>
>> Saludos
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> ------------------------------
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> 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
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 i.ivanova at mastervoice.it  Thu Apr 28 15:15:30 2011
From: i.ivanova at mastervoice.it (Irina Ivanova)
Date: Thu, 28 Apr 2011 13:15:30 +0200
Subject: [Freeswitch-users] transfer_ringback triggered by "183 Session
 progress"
In-Reply-To: <4DB159B9.80105@mastervoice.it>
References: <4DADA951.30708@mastervoice.it>
	<C1AC4D08-A4CC-49A1-8331-318DD18DF7FB@freeswitch.org>
	<4DB159B9.80105@mastervoice.it>
Message-ID: <4DB94C52.6090803@mastervoice.it>

On 04/22/2011 12:34 PM, Irina Ivanova wrote:
> Thank you for the answer. I tried with both ignore_early_media=true 
> and ignore_early_media=false, but unfortunately in both cases the 
> behaviour is the same, the ringback_transfer is triggered.
> More than this, I've noticed that also with normal outgoing calls when 
> the channel is not answered before executing the bridge there is the 
> similar behaviour. As soon as 183 is received, the ringtone generated 
> by freeswitch can be heard (I suppose it is generated by freeswitch 
> because looking on the tcpdump of the call no any RTP stream sent to 
> freeswitch could be seen).
>
> In my first test ignore_early_media is set to false.
> As can be seen from the log here http://pastebin.freeswitch.org/16155 
> the line 168 says "entering state [early][180]" and at this moment the 
> caller starts to hear ringing.
>
> 165. 2011-04-22 11:30:45.931314 [DEBUG] sofia.c:4659 Channel 
> sofia/device/3382XXXXXX entering state [proceeding][183]
> 166. 2011-04-22 11:30:45.931314 [NOTICE] sofia.c:4737 Ring-Ready 
> sofia/device/3382XXXXXX!
> 167. 2011-04-22 11:30:45.933870 [NOTICE] mod_sofia.c:2185 Ring-Ready 
> sofia/internal/15 at 192.168.1.124!
> 168. 2011-04-22 11:30:45.933870 [DEBUG] sofia.c:4659 Channel 
> sofia/internal/15 at 192.168.1.124 entering state [early][180]
>
> The second test is made with ignore_early_media set to true.
> Here is the log of the test: http://pastebin.freeswitch.org/16156. On 
> the line 167 I can still see "entering state [early][180]" and I can 
> still hear the ringing.
> 167. 2011-04-22 12:05:21.450866 [DEBUG] sofia.c:4659 Channel 
> sofia/internal/15 at 192.168.1.124 entering state [early][180]
>
> I suppose it is meant to be like this, I am just wondering if there is 
> a way to avoid this short ringing sound that could disturb the 
> callers. May be some workaround?...
> I also played with uuid_broadcast and uuid_displace trying to 
> substitute the audio stream which is played to the caller as soon as 
> channel enters the state early, but at the end didn't get the desired 
> result.
>
>
>
> On 04/21/2011 04:17 PM, Brian West wrote:
>> Happen to be setting ignore_early_media=true?
>>
>>
>> On Apr 19, 2011, at 10:25 AM, Irina Ivanova wrote:
>>
>>> Hi,
>>>
>>> I've noticed that if to set transfer_ringback (to any ringback tone) 
>>> for
>>> already answered call and then to execute the bridge to some external
>>> number through the gateway, the ringing is triggered not only by "180
>>> Ringing" SIP response, but also when "183 Session progress" is 
>>> received.
>>> Does anybody know if there is a way to make transfer_ringback not to be
>>> triggered by 183? I need it because in the case when the destination
>>> number is busy and provider sends me 183 and then 486 (Busy here) the
>>> caller hears one ringback tone and then the busy tone which makes an
>>> impression that the called party rejected the call.
>>>
>>> Thanks,
>>> Irina
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users 
>>>
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>

Ok, I found the solution in the present mailing list, as the similar 
case was already discussed: 
http://lists.freeswitch.org/pipermail/freeswitch-users/2011-February/068693.html.
Shame on me I haven't noticed it before!

The trick is to use "sip_ignore_183nosdp" variable and to set it to 
"true" inside the dial string, so it's set on the b leg of the call.

And if I am not mistaken, the crash issue that was mentioned in the 
thread, is already fixed (git revision 
http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=ce5c846). 
Good job!



From mayamatakeshi at gmail.com  Thu Apr 28 17:18:03 2011
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 28 Apr 2011 22:18:03 +0900
Subject: [Freeswitch-users] Does FS supports speex with ptime=40?
Message-ID: <BANLkTinRv5swkHVTkKPqnaT_fCtiUqP3yg@mail.gmail.com>

I am testing making calls with x-lite 4. Although it doesn't send a ptime in
its INVITE/SDP, i can see it is sending RTP using ptime=40 as I can count 25
UDP packets per second.
When I try to bridge this to PSTN which responds with PCMU, at the PSTN side
I hear trembling/metallic/robotic audio.

I was suggested to set FS to use ptime=40, so I've tried setting codecs with
speex at 8000h@40i

But it doesn't work. I get: "488 Not Acceptable Here" (Reason:
Q.850;cause=88;text="INCOMPATIBLE_DESTINATION)

So it seems only ptime=20 is supported as listed here:
http://wiki.freeswitch.org/wiki/Codecs#Transcodable_codecs

Could mod_speex be changed to work with ptime=40 too?
Or is there anything else that i can try?

regards,
takeshi
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From brian at freeswitch.org  Thu Apr 28 18:32:58 2011
From: brian at freeswitch.org (Brian West)
Date: Thu, 28 Apr 2011 09:32:58 -0500
Subject: [Freeswitch-users] Does FS supports speex with ptime=40?
In-Reply-To: <BANLkTinRv5swkHVTkKPqnaT_fCtiUqP3yg@mail.gmail.com>
References: <BANLkTinRv5swkHVTkKPqnaT_fCtiUqP3yg@mail.gmail.com>
Message-ID: <484AE924-7A82-437F-B8E1-AA38183F4ECA@freeswitch.org>

Nope.

/b

On Apr 28, 2011, at 8:18 AM, mayamatakeshi wrote:

> I am testing making calls with x-lite 4. Although it doesn't send a ptime in
> its INVITE/SDP, i can see it is sending RTP using ptime=40 as I can count 25
> UDP packets per second.
> When I try to bridge this to PSTN which responds with PCMU, at the PSTN side
> I hear trembling/metallic/robotic audio.
> 
> I was suggested to set FS to use ptime=40, so I've tried setting codecs with
> speex at 8000h@40i
> 
> But it doesn't work. I get: "488 Not Acceptable Here" (Reason:
> Q.850;cause=88;text="INCOMPATIBLE_DESTINATION)
> 
> So it seems only ptime=20 is supported as listed here:
> http://wiki.freeswitch.org/wiki/Codecs#Transcodable_codecs
> 
> Could mod_speex be changed to work with ptime=40 too?
> Or is there anything else that i can try?
> 
> regards,
> 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



From bracken_dave at yahoo.com  Thu Apr 28 18:38:06 2011
From: bracken_dave at yahoo.com (Dave Bracken)
Date: Thu, 28 Apr 2011 07:38:06 -0700 (PDT)
Subject: [Freeswitch-users] Call Hold scenario, please help.
Message-ID: <597945.23381.qm@web114503.mail.gq1.yahoo.com>

This is my scenario, i have a caller call in, and i bridge them to an operator. 
The operator would need to put the caller on hold(with music) and then the 
operator freed up for other calls,etc. Then the operator can pull the caller 
back off hold.
I need to know how to do this using the commands on an "outbound" socket 
connection to fs. What the commands are, the order, etc. Cant seem to find 
anyone who knows this yet, so i hope one of you fs gurus can point me in the 
right direction.
Thanks in advance,
Dave
in irc aka FSnewbDelphi.
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From msc at freeswitch.org  Thu Apr 28 19:22:53 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Apr 2011 08:22:53 -0700
Subject: [Freeswitch-users] ESL & DTMF
In-Reply-To: <BANLkTimPUQC2USHkQ9b72jAV11Sms2KWkA@mail.gmail.com>
References: <BANLkTimPUQC2USHkQ9b72jAV11Sms2KWkA@mail.gmail.com>
Message-ID: <BANLkTim1PZMeyjqXyYGH90T_8XVHhWxo9g@mail.gmail.com>

Can you tell us more about how you're doing this? It sounds like you are
doing IVR-ish stuff which, IMHO, does not lend itself for async ESL. I would
rather see you do something akin to the fs_ivrd + perl ESL::IVR examples.

-MC

On Thu, Apr 28, 2011 at 1:58 AM, Antonio Teixeira
<eagle.antonio at gmail.com>wrote:

> Good Morning.
>
> I'm currently in the process of converting my IVR's into ESL , freedom :D.
> From my testing is extremely flexible until i have reached a wall.
>
> We use TTS to generate alot of information for our users then we need to
> collect some answers, in session i had GetDigists , yay :).
> In ESL i have the DTMF event the problem is that from my testing and wiki ,
> DTMF event is only send down the socket using asycn , now i need "sync"
> because the IVRs need to understand the exact position of the user and blah
> blah blah ,  need to be dynamically extended in realtime and would be hard
> to implement in async , specially the timing for digit collection.
>
> So i tried capturing the event with no luck the only one i got is
> SERVER_DISCONNECT at the end.
>
> I think It is possible if for every thread i would raise a listener for
> that specific uuid that would share memory with the first one so i could
> read that DTMF events that arrives trough the other socket but i think this
> is way to much duct tape for a simple thing.
>
> So any help on this ?
> Thanks
> A/T
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From eagle.antonio at gmail.com  Thu Apr 28 20:28:56 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Thu, 28 Apr 2011 16:28:56 +0000
Subject: [Freeswitch-users] ESL & DTMF
In-Reply-To: <BANLkTim1PZMeyjqXyYGH90T_8XVHhWxo9g@mail.gmail.com>
References: <BANLkTimPUQC2USHkQ9b72jAV11Sms2KWkA@mail.gmail.com>
	<BANLkTim1PZMeyjqXyYGH90T_8XVHhWxo9g@mail.gmail.com>
Message-ID: <BANLkTikLcK5AJ_wha8ER22VOnDQtN16o0w@mail.gmail.com>

Hello Michael.

Yes we are doing ivr stuf using sync , mainly because we have some routines
that require step by step and timing processing , like PIN verification ,
Security Clearance , etc.
This is mainly why i use mode sync ( I'm not telling you I'm right just
trying to give you an ideia of what we are doing)

>From my understanding , async is a full dump to FS of all the commands for
that DP like :
Answer()
Speak (' Hello World')
GetDigits()
now if the person presses 2 , hears a menu with more options and presses
another key.
What will call for example a program that send an SMS to give you an ideia.


Example :
Answer
Speak('Press 1 or 2')
GetDigits()

if dtmf == 1:
speak('bye')
elif dtmf == 2:
SendMySMS('blahblah')
speak ('now pres 3 4 or 5')
get digits()
if dtmf == 3:
etc , etc etc.


With step by step this is simple ( except the GetDigits Part)



2011/4/28 Michael Collins <msc at freeswitch.org>

> Can you tell us more about how you're doing this? It sounds like you are
> doing IVR-ish stuff which, IMHO, does not lend itself for async ESL. I would
> rather see you do something akin to the fs_ivrd + perl ESL::IVR examples.
>
> -MC
>
> On Thu, Apr 28, 2011 at 1:58 AM, Antonio Teixeira <eagle.antonio at gmail.com
> > wrote:
>
>> Good Morning.
>>
>> I'm currently in the process of converting my IVR's into ESL , freedom :D.
>> From my testing is extremely flexible until i have reached a wall.
>>
>> We use TTS to generate alot of information for our users then we need to
>> collect some answers, in session i had GetDigists , yay :).
>> In ESL i have the DTMF event the problem is that from my testing and wiki
>> , DTMF event is only send down the socket using asycn , now i need "sync"
>> because the IVRs need to understand the exact position of the user and blah
>> blah blah ,  need to be dynamically extended in realtime and would be hard
>> to implement in async , specially the timing for digit collection.
>>
>> So i tried capturing the event with no luck the only one i got is
>> SERVER_DISCONNECT at the end.
>>
>> I think It is possible if for every thread i would raise a listener for
>> that specific uuid that would share memory with the first one so i could
>> read that DTMF events that arrives trough the other socket but i think this
>> is way to much duct tape for a simple thing.
>>
>> So any help on this ?
>> Thanks
>> A/T
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From eagle.antonio at gmail.com  Thu Apr 28 20:34:17 2011
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Thu, 28 Apr 2011 16:34:17 +0000
Subject: [Freeswitch-users] ESL & DTMF
In-Reply-To: <BANLkTikLcK5AJ_wha8ER22VOnDQtN16o0w@mail.gmail.com>
References: <BANLkTimPUQC2USHkQ9b72jAV11Sms2KWkA@mail.gmail.com>
	<BANLkTim1PZMeyjqXyYGH90T_8XVHhWxo9g@mail.gmail.com>
	<BANLkTikLcK5AJ_wha8ER22VOnDQtN16o0w@mail.gmail.com>
Message-ID: <BANLkTik8742o_Qn9CaF0xtx7HOPKxNJRyw@mail.gmail.com>

I also remembered that i can always do this with play_get_digits and
silence_stream.

:D

Going to try that tomorrow *
*
2011/4/28 Antonio Teixeira <eagle.antonio at gmail.com>

> Hello Michael.
>
> Yes we are doing ivr stuf using sync , mainly because we have some routines
> that require step by step and timing processing , like PIN verification ,
> Security Clearance , etc.
> This is mainly why i use mode sync ( I'm not telling you I'm right just
> trying to give you an ideia of what we are doing)
>
> From my understanding , async is a full dump to FS of all the commands for
> that DP like :
> Answer()
> Speak (' Hello World')
> GetDigits()
> now if the person presses 2 , hears a menu with more options and presses
> another key.
> What will call for example a program that send an SMS to give you an ideia.
>
>
> Example :
> Answer
> Speak('Press 1 or 2')
> GetDigits()
>
> if dtmf == 1:
> speak('bye')
> elif dtmf == 2:
> SendMySMS('blahblah')
> speak ('now pres 3 4 or 5')
> get digits()
> if dtmf == 3:
> etc , etc etc.
>
>
> With step by step this is simple ( except the GetDigits Part)
>
>
>
>
> 2011/4/28 Michael Collins <msc at freeswitch.org>
>
>> Can you tell us more about how you're doing this? It sounds like you are
>> doing IVR-ish stuff which, IMHO, does not lend itself for async ESL. I would
>> rather see you do something akin to the fs_ivrd + perl ESL::IVR examples.
>>
>> -MC
>>
>> On Thu, Apr 28, 2011 at 1:58 AM, Antonio Teixeira <
>> eagle.antonio at gmail.com> wrote:
>>
>>> Good Morning.
>>>
>>> I'm currently in the process of converting my IVR's into ESL , freedom
>>> :D.
>>> From my testing is extremely flexible until i have reached a wall.
>>>
>>> We use TTS to generate alot of information for our users then we need to
>>> collect some answers, in session i had GetDigists , yay :).
>>> In ESL i have the DTMF event the problem is that from my testing and wiki
>>> , DTMF event is only send down the socket using asycn , now i need "sync"
>>> because the IVRs need to understand the exact position of the user and blah
>>> blah blah ,  need to be dynamically extended in realtime and would be hard
>>> to implement in async , specially the timing for digit collection.
>>>
>>> So i tried capturing the event with no luck the only one i got is
>>> SERVER_DISCONNECT at the end.
>>>
>>> I think It is possible if for every thread i would raise a listener for
>>> that specific uuid that would share memory with the first one so i could
>>> read that DTMF events that arrives trough the other socket but i think this
>>> is way to much duct tape for a simple thing.
>>>
>>> So any help on this ?
>>> Thanks
>>> A/T
>>>
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From justlikeef at gmail.com  Thu Apr 28 21:29:21 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Thu, 28 Apr 2011 13:29:21 -0400
Subject: [Freeswitch-users] [OT] SIP Trunk Service
Message-ID: <201104281329.21369.justlikeef@gmail.com>

Sorry for the OT post....

I am having a hard time finding someone that can deliver reliable, business 
grade SIP trunks and port existing numbers at a service location (706-776) in 
Northern Georgia.

If there is anyone that either can provide service, thinks they can provide 
service, or knows of someone that think they can provide service, please 
contact me off list...


Thanks,
Rob


From justlikeef at gmail.com  Thu Apr 28 21:31:09 2011
From: justlikeef at gmail.com (Rob Hutton)
Date: Thu, 28 Apr 2011 13:31:09 -0400
Subject: [Freeswitch-users] Call Hold scenario, please help.
In-Reply-To: <597945.23381.qm@web114503.mail.gq1.yahoo.com>
References: <597945.23381.qm@web114503.mail.gq1.yahoo.com>
Message-ID: <201104281331.09281.justlikeef@gmail.com>

Look at park and valet park...

On Thursday 28 April 2011 10:38:06 Dave Bracken wrote:
> This is my scenario, i have a caller call in, and i bridge them to an
> operator. The operator would need to put the caller on hold(with music)
> and then the operator freed up for other calls,etc. Then the operator can
> pull the caller back off hold.
> I need to know how to do this using the commands on an "outbound" socket
> connection to fs. What the commands are, the order, etc. Cant seem to find
> anyone who knows this yet, so i hope one of you fs gurus can point me in
> the right direction.
> Thanks in advance,
> Dave
> in irc aka FSnewbDelphi.


From vetali100 at gmail.com  Thu Apr 28 22:27:07 2011
From: vetali100 at gmail.com (Vitalie Colosov)
Date: Thu, 28 Apr 2011 21:27:07 +0300
Subject: [Freeswitch-users] attended transfer to gateway
In-Reply-To: <1303914967008-6309679.post@n2.nabble.com>
References: <BANLkTinB6uFAWDnq-Ot1tiSPi99pu0Oruw@mail.gmail.com>
	<BANLkTikkF95DCGgpQJ-kKCR-U_=X7fF5+g@mail.gmail.com>
	<BANLkTinAXp_u4+0OwNjY5dEDH5YHmFR-fw@mail.gmail.com>
	<65A0D45D-0666-493C-B53A-D9DC882EE77C@freeswitch.org>
	<BANLkTin8Vdg=a3em_2O1bkX3C5qoGLCBBg@mail.gmail.com>
	<BANLkTi=u9d5NWeO=Z1Rtnn_7+qAqb1cuKQ@mail.gmail.com>
	<BANLkTin=3jrY0kzjw-CsOZM07O4XtVBtWQ@mail.gmail.com>
	<1303914967008-6309679.post@n2.nabble.com>
Message-ID: <BANLkTi=aQkGM_XVFk=Jsy79cWJyUe3nusg@mail.gmail.com>

I was able to reproduce it using the latest git with absolutely no changes.
So the problem is confirmed.

Jira issue was opened:
http://jira.freeswitch.org/browse/FS-3275

Thank you,
Vitalie


2011/4/27 Jeff Lenk <jeff at jefflenk.com>

> Please open a Jira on this and attach all relevent information
>
> enable debug logging
> sofia siptrace global on
>
> And if it is reproducable by the default config - outline the flow and make
> notes in the bug report.
>
> Thanks
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/attended-transfer-to-gateway-tp6288289p6309679.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 krice at freeswitch.org  Thu Apr 28 22:31:45 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 28 Apr 2011 13:31:45 -0500
Subject: [Freeswitch-users] [OT] SIP Trunk Service
In-Reply-To: <201104281329.21369.justlikeef@gmail.com>
Message-ID: <C9DF1CC1.14908%krice@freeswitch.org>

Is this a Rural ILEC there?

K


On 4/28/11 12:29 PM, "Rob Hutton" <justlikeef at gmail.com> wrote:

> Sorry for the OT post....
> 
> I am having a hard time finding someone that can deliver reliable, business
> grade SIP trunks and port existing numbers at a service location (706-776) in
> Northern Georgia.
> 
> If there is anyone that either can provide service, thinks they can provide
> service, or knows of someone that think they can provide service, please
> contact me off list...
> 
> 
> Thanks,
> Rob
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From rhuddleston at gmail.com  Thu Apr 28 22:37:52 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Thu, 28 Apr 2011 14:37:52 -0400
Subject: [Freeswitch-users] [OT] SIP Trunk Service
In-Reply-To: <C9DF1CC1.14908%krice@freeswitch.org>
References: <201104281329.21369.justlikeef@gmail.com>
	<C9DF1CC1.14908%krice@freeswitch.org>
Message-ID: <097801cc05d3$62da3e10$288eba30$@com>

No.. They are both an ILEC and a SIP ISP (International and Domestic).

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Thursday, April 28, 2011 2:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [OT] SIP Trunk Service

Is this a Rural ILEC there?

K


On 4/28/11 12:29 PM, "Rob Hutton" <justlikeef at gmail.com> wrote:

> Sorry for the OT post....
> 
> I am having a hard time finding someone that can deliver reliable,
business
> grade SIP trunks and port existing numbers at a service location (706-776)
in
> Northern Georgia.
> 
> If there is anyone that either can provide service, thinks they can
provide
> service, or knows of someone that think they can provide service, please
> contact me off list...
> 
> 
> Thanks,
> Rob
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



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



From rhuddleston at gmail.com  Thu Apr 28 22:38:33 2011
From: rhuddleston at gmail.com (Robert Huddleston)
Date: Thu, 28 Apr 2011 14:38:33 -0400
Subject: [Freeswitch-users] [OT] SIP Trunk Service
In-Reply-To: <C9DF1CC1.14908%krice@freeswitch.org>
References: <201104281329.21369.justlikeef@gmail.com>
	<C9DF1CC1.14908%krice@freeswitch.org>
Message-ID: <097a01cc05d3$7b816e20$72844a60$@com>

Disregard sorry... user error...

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Thursday, April 28, 2011 2:32 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] [OT] SIP Trunk Service

Is this a Rural ILEC there?

K


On 4/28/11 12:29 PM, "Rob Hutton" <justlikeef at gmail.com> wrote:

> Sorry for the OT post....
> 
> I am having a hard time finding someone that can deliver reliable,
business
> grade SIP trunks and port existing numbers at a service location (706-776)
in
> Northern Georgia.
> 
> If there is anyone that either can provide service, thinks they can
provide
> service, or knows of someone that think they can provide service, please
> contact me off list...
> 
> 
> Thanks,
> Rob
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



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



From jmesquita at freeswitch.org  Fri Apr 29 00:59:13 2011
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 28 Apr 2011 17:59:13 -0300
Subject: [Freeswitch-users] Script para chequear un numero de destino
In-Reply-To: <BANLkTimfcULSHkf9Zo5k+VugLS+FAKzHWg@mail.gmail.com>
References: <BANLkTi=Lz=gBRrcX_WmYN3iZXhzBv4wSPA@mail.gmail.com>
	<968175.88614.qm@web34306.mail.mud.yahoo.com>
	<BANLkTi=ZFv8D66r2SmV5VXQOvKP1uKX9pQ@mail.gmail.com>
	<BANLkTimfcULSHkf9Zo5k+VugLS+FAKzHWg@mail.gmail.com>
Message-ID: <BANLkTin0zS4CWyOC1a3e4TcHUmy=AA=u-g@mail.gmail.com>

Hello Juan,

By the Iba?ez, I would assume you're also a spanish speaking person?

IMHO, ESL and LUA for small scale applications is a matter of preference.
They can accomplish the same. I do not program in LUA very well, so I
normally prefer ESL, but I take it can be a bit more tricky since you need
to control the full call flow.

On the other hand, if you are talking about massive IVRs and such, I would
definitely suggest ESL since you can run it on another box and leave all the
processing power to freeswitch where freeswitch is hosted. LUA cant to that,
it needs to run on the same box and you mix the logic where CPU is very
sensitive.

Does that make it clearer for you?

Fernando, como estamos con la lectura? En que mas te puedo ayudar?

Regards,
Jo?o Mesquita



2011/4/28 Juan Antonio Iba?ez Santorum <juanito1982 at gmail.com>

> Now you are talking about ESL? Which scenarios are suitable for ESL? When
> would be nice to use ESL and when LUA?
>
> Regards
>
>
> 2011/4/28 Jo?o Mesquita <jmesquita at freeswitch.org>
>
>> Hey Ken! Even tho spanish is not my native language, I live in Argentina,
>> so I can pretty much get away with it. Any help on portuguese or spanish
>> that this list may require, I am available to help.
>>
>> Fernando,
>>
>> La manera mas rapida es algo relativo. Depende mucho de con que te sentis
>> mas comodo. Me gusta mucho ESL y tambi?n me gusta mucho xml_curl. odbc_query
>> nunca lo use, pero parece sencillo.
>>
>> A mi me encanta Python tambien y suelo usarlo con ESL. Uso bastante
>> tambi?n Pylons pero ya trabaje con Django tambien. Sea como for, lo que vos
>> necesit?s si vas a usar ESL es un outbound socket. Te recomiendo que
>> empiezes leyendo esta pagina para que te vayas enterando de que se trata:
>> http://wiki.freeswitch.org/wiki/Event_Socket_Outbound
>>
>> Cuando termines de leer, yo te puedo tirar unos "tips" de como implementar
>> lo que necesit?s en Python sin demasiados problemas.
>>
>> Saludos,
>> Jo?o Mesquita
>>
>>
>>
>>
>> 2011/4/28 FERNANDO VILLARROEL <fvillarroel at yahoo.com>
>>
>>> Hola Jo?o.
>>>
>>> Si programo, mi lenguaje preferido es Python yo vengo de Asterisk donde
>>> usaba Python para programar AGI.
>>>
>>> En web algo me manejo tambien por lo que creo no tendre problemas, le he
>>> metido mano a Django y Web2py.
>>>
>>> Si no fuera mucho pedirte me gustaria aprender un poco de todo asi decido
>>> luego como lo voy haciendo, ESL me llama la atenci?n ya que me gustaria
>>> comenzar a aprehender a usarlo, ya sabes capturar eventos de FS, enviar
>>> eventos a FS, etc.
>>>
>>> Pero bueno por ahora si me ayudas a resolver mi problema usando lo mas
>>> sencillo te lo agradeceria mucho.
>>>
>>> El problema es que debo crear una tabla con aproximadamente 4500
>>> breackouts de numeros moviles, entonces si me entra una llamada con
>>> destination_number = 12345678 debo consultar si los 3 primeros dijitos estan
>>> presentes en la tabla si es si entonces reenvio la llamada a un Gateway y si
>>> no entonces la declino para que mi cliente la enrute por otro proveedor.
>>>
>>> Saludos.
>>>
>>>
>>>
>>>
>>> --- On *Wed, 4/27/11, Jo?o Mesquita <jmesquita at freeswitch.org>* wrote:
>>>
>>>
>>> From: Jo?o Mesquita <jmesquita at freeswitch.org>
>>> Subject: Re: [Freeswitch-users] Script para chequear un numero de destino
>>> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
>>> Date: Wednesday, April 27, 2011, 11:36 PM
>>>
>>>
>>> Hola Fernando,
>>>
>>> Yo mas o menos hablo espa?ol as? que trato de darte una mano. Ten?s
>>> muchas opciones. mod_easyroute como lo menciona Ken es una, ESL
>>> (event_socket) es otra, xml_curl es otra y uno de los "bindings" como lua,
>>> python, perl, etc... es otra.
>>>
>>> Lo que tenemos que encontrar es lo que a vos te es mas comodo. Program?s?
>>> En que lenguaje? Si no program?s, mod_easyroute seguramente ser? lo mejor.
>>> ESL es lo m?s complicadito de todos ya que el dialplan entero lo tenes que
>>> programar desde tu socket y tirar los comandos hacia freeswitch en un orden
>>> especifico. XML_CURL es muy facil si sabes desarrollar en web.
>>>
>>> Enfim, espero noticias tuyas para que te pueda ayudar con un determinado
>>> camino.
>>>
>>> Saludos,
>>>
>>> Jo?o Mesquita
>>>
>>>
>>>
>>> 2011/4/27 Ken Rice <krice at freeswitch.org<http://mc/compose?to=krice at freeswitch.org>
>>> >
>>>
>>>  Routing a call if a number exists in a database...  Hmmm sounds
>>> suspiciously a lot like mod_easyroute
>>>
>>> k
>>>
>>>
>>>
>>> On 4/27/11 8:24 PM, "Roger Castaldo" <roger.castaldo at gmail.com> wrote:
>>>
>>> I ran your email through google translate and from what I can tell you
>>> want freeswitch to route a call if the number exists in database X.  That
>>> being said, the event socket is not the way to go, look at doing it via a
>>> lua script call from the dial plan, then you can have the lua query the
>>> database, if it exists, use the api to route the call, or set a flag for the
>>> dial plan, otherwise it can hang up, or play a message and hang up which
>>> ends the call and stops processing in your dial plan.
>>>
>>> 2011/4/27 FERNANDO VILLARROEL <fvillarroel at yahoo.com>
>>>
>>> Hola a todos.
>>>
>>> Voy a recibir trafico desde un Gateway A que mi FS enrutar? hacia un
>>> Gateway B.
>>>
>>> Hasta ahy todo bien, el asunto es que antes de enrutar la llamada hacia
>>> el gateway B de alguna manera debo consultar el numero de destino en una
>>> base de datos si es as? entonces se enruta la llamada en caso contrario la
>>> llamda la debo declinar de mi lado para que el Gateway A pueda enrutar esa
>>> llamada por otro proveedor distinto de mi.
>>>
>>> Entonces lo que pienso hacer es un Script usando event_socket para que
>>> haga la consulta en la base de datos, entonces si el numero de destino esta
>>> la base de datos se enruta la llamada en caso contrario la declino para que
>>> la llamada continue por otro proveedor.
>>>
>>> Mi tema es que no se me ocurre como hacer esto, imagino que  en el
>>> dialplan para que dependiendo de la existencia del numero de destino en la
>>> base de datos haga una llamada o en caso contrario haga otra cosa.
>>>
>>> Agredecere aguna idea o sugerencia sobre como resolver este problema.
>>>
>>> Con Asterisk usaria DeadAgi para pasarle el control a un script que
>>> consulte con la base de datos y me retorne true si puedo llamar o false en
>>> caso contario. Y digo DeadAgi ya que el control lo toma el script, ya que si
>>> usara Agi el control sigue en Asterisk independiente de lo que este haciendo
>>> el Script.
>>>
>>> Aprovecho entonces de consultar con event_socket el control pasa al
>>> script o el dialplan continua ejecutandose.
>>>
>>> Agradecere cualquier idea o sugerencia.
>>>
>>> Saludos
>>>
>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> ------------------------------
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> 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
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 29 01:18:25 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Apr 2011 14:18:25 -0700
Subject: [Freeswitch-users] Script para chequear un numero de destino
In-Reply-To: <BANLkTin0zS4CWyOC1a3e4TcHUmy=AA=u-g@mail.gmail.com>
References: <BANLkTi=Lz=gBRrcX_WmYN3iZXhzBv4wSPA@mail.gmail.com>
	<968175.88614.qm@web34306.mail.mud.yahoo.com>
	<BANLkTi=ZFv8D66r2SmV5VXQOvKP1uKX9pQ@mail.gmail.com>
	<BANLkTimfcULSHkf9Zo5k+VugLS+FAKzHWg@mail.gmail.com>
	<BANLkTin0zS4CWyOC1a3e4TcHUmy=AA=u-g@mail.gmail.com>
Message-ID: <BANLkTindwkD3NKDFCjFq1XaKx+aCjiee1Q@mail.gmail.com>

2011/4/28 Jo?o Mesquita <jmesquita at freeswitch.org>

> Hello Juan,
>
> By the Iba?ez, I would assume you're also a spanish speaking person?
>
> IMHO, ESL and LUA for small scale applications is a matter of preference.
> They can accomplish the same. I do not program in LUA very well, so I
> normally prefer ESL, but I take it can be a bit more tricky since you need
> to control the full call flow.
>
> On the other hand, if you are talking about massive IVRs and such, I would
> definitely suggest ESL since you can run it on another box and leave all the
> processing power to freeswitch where freeswitch is hosted. LUA cant to that,
> it needs to run on the same box and you mix the logic where CPU is very
> sensitive.
>
> Does that make it clearer for you?
>
> Fernando, como estamos con la lectura? En que mas te puedo ayudar?
>
> Regards,
> Jo?o Mesquita
>
>
I agree with J. Lua is a great choice for doing simple things from the
dialplan. Even basic IVRs can be done effectively with dialplan scripts in
Lua. However, if you need to scale, or if your program needs to be able to
react to system events, then ESL is probably the way to go. The good news
about going with ESL is that there is quite literally nothing that a Lua
dialplan script can do that an ESL program cannot. The only real difference
is that Lua dp scripts are simpler and more limited than an ESL-based
program.

-MC
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From bogusmaster at o2.pl  Thu Apr 28 19:08:23 2011
From: bogusmaster at o2.pl (bogusmaster at o2.pl)
Date: Thu, 28 Apr 2011 17:08:23 +0200
Subject: [Freeswitch-users] Fwd: problem with transferring calls when using
	ivr
In-Reply-To: <op.vunkngteagcvls@master>
References: <op.vunkngteagcvls@master>
Message-ID: <op.vunln9d3agcvls@master>

Hi all !

I'm a freeswitch newbie taking care of installation that I've recently
taken over from a colleague who left our company.
Currently I'm having a problem with transferring calls that went thru ivr
menu.

The call scenario looks like this: external number A calls our external
number E. Then, in ivr menu it selects some option. After that A is
connected with internal number I.
When internal number I tries to transfer call from A to another internal
number I2, call hangs up.

If I turn the ivr off, as it is commented in the excerpt from
/conf/dialplan/public/01_freephone_inbound.xml below, transferring calls
works fine.

     <extension name="general_inbound">
       <condition field="destination_number"
expression="^(freephone_XXXX_in)$">
         <action application="info"/>
         <action application="transfer" data="100 XML default"/>
         <!--action application="javascript" data="simpleivr.js"/-->
       </condition>
     </extension>

I attach my ivr script.

Any hints would be very appreciated.

Regards,
Bart


-- 
Using Opera's revolutionary email client: http://www.opera.com/mail/
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From bracken_dave at yahoo.com  Fri Apr 29 00:24:35 2011
From: bracken_dave at yahoo.com (Dave Bracken)
Date: Thu, 28 Apr 2011 13:24:35 -0700 (PDT)
Subject: [Freeswitch-users] Call Hold scenario, please help.
In-Reply-To: <201104281331.09281.justlikeef@gmail.com>
References: <597945.23381.qm@web114503.mail.gq1.yahoo.com>
	<201104281331.09281.justlikeef@gmail.com>
Message-ID: <283095.73945.qm@web114505.mail.gq1.yahoo.com>

Rob,

Thanks for the thought, but i am already using the park command. But after i 
park the call, i cannot get the music to play.(trying to use the playback 
command) Also, i am not getting dial tone on the operator phone, so i am not 
sure the line has been fully disconnected to free the operator up for other 
calls.
i was hoping someone had an ordered list of the commands to complete my original 
scenario.
thanks in advance
Dave


________________________________
From: Rob Hutton <justlikeef at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Cc: Dave Bracken <bracken_dave at yahoo.com>
Sent: Thu, April 28, 2011 12:31:09 PM
Subject: Re: [Freeswitch-users] Call Hold scenario, please help.

Look at park and valet park...

On Thursday 28 April 2011 10:38:06 Dave Bracken wrote:
> This is my scenario, i have a caller call in, and i bridge them to an
> operator. The operator would need to put the caller on hold(with music)
> and then the operator freed up for other calls,etc. Then the operator can
> pull the caller back off hold.
> I need to know how to do this using the commands on an "outbound" socket
> connection to fs. What the commands are, the order, etc. Cant seem to find
> anyone who knows this yet, so i hope one of you fs gurus can point me in
> the right direction.
> Thanks in advance,
> Dave
> in irc aka FSnewbDelphi.
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From david.ponzone at ipeva.fr  Fri Apr 29 01:24:18 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Thu, 28 Apr 2011 23:24:18 +0200
Subject: [Freeswitch-users] Trunking between Lync and FreeSWITCH
Message-ID: <3120C73F-FFD6-4979-87FD-A61AA1A2E14C@ipeva.fr>

Well, I thought I could start a thread about that.
I am trying to accomplish this, and I guess I am not the only one.
Lync 2010 seems to have quite some success (compared to OCS), so there are probably business opportunities around this.

My first attempt is a half success, as I got calls from Lync to FreeSWITCH working (need some extensive testing, though), and calls from FreeSWITCH to Lync are connecting ok, but they get disconnected after circa 30 seconds.

I am trying to pinpoint the reason for this disconnection, but untll now, I had no luck.
RTCP requirement was disabled on the Lync side (RTCPActiveCalls), but anyway I have enabled RTCP on the FreeSWITCH, so it should not be that.
RTP is flowing normally, it's not a call on hold or muted.
Refer was disabled on the Lync side.
SessionTimer is enabled on the Lync side (but I dont think I ever saw a REINVITE caused by the session timer).
I just suddenly receive a BYE from Lync, after 30-35 seconds.
On the Lync side, as you can guess, it's not easy to access any meaningful logs. I think our partner managing the Lync will have to escalate that to the SVP Product Engineering to get the right command to enable the interesting debug mode :)

I've been told that some folks at MS were claiming a such trunking was not possible.
Well, I would tend to say otherwise, as I am not far to get this working, except of course if they just hardcoded some forbidden Owner Usernames in the SDP, like FreeSWITCH, in order to save this business from the mean OpenSource world and leave it to commercial SBC vendors.
I can hardly imagine they would dare to do that in 2011, so the question remains opened: what may be closing that call after 30 seconds ?

I would be glad to discuss the subject, here or privately, with anyone who is involved on a such project, or planning to be soon.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.




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From eric at loopfx.com  Fri Apr 29 01:52:34 2011
From: eric at loopfx.com (Eric Beard)
Date: Thu, 28 Apr 2011 17:52:34 -0400
Subject: [Freeswitch-users] Call quality issues
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DF6A0EDB@doctorbob.loop.local>

Hello,

I ran into some major call quality issues this week, and I'm trying to figure out how to troubleshoot things.  I've been running FreeSwitch for a few weeks, and suddenly a few days ago my call quality dropped drastically.  I had been running more than 100 concurrent calls, with the CPU at less than 20%, but now at 20 concurrent calls, the CPU is still at a little less than 20%, and the call quality is bad - any higher and calls go almost completely silent.  There is a direct correlation between the number of simultaneous calls and call quality.

I have tested against multiple gateways, same results against each, so it's not an issue with the gateway.

I have captured packets on the machine and analyzed them with Wireshark.  It seems like the inbound packets are all fine, no jitter or loss.  But the packets being sent by FreeSwitch are degraded.

One sample call showed:

Drop by Jitter Buff:158(14.1%) Out of Seq 0 (0.0%) Wrong Timestamp 96(8.6%)

I'm not sure what that meant, but Wireshark's analysis and playback was very accurate, sounding exactly like what I hear when I test a call.  (Freeswitch recordings are by comparison very clean sounding)

So what's my next step?  Where do I go to find the problem? I know FreeSwitch is capable of handling this traffic.

I last updated from git yesterday.  I'm running on a dual-quad 64 bit openSuse machine with 8Gb RAM.

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 fvillarroel at yahoo.com  Fri Apr 29 02:43:43 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Thu, 28 Apr 2011 15:43:43 -0700 (PDT)
Subject: [Freeswitch-users] Load mod_odbc_query error
Message-ID: <391041.29065.qm@web34308.mail.mud.yahoo.com>

Hi all.

I getting mod_odbc_query from contrib.

My FS Version:

FreeSWITCH Version 1.0.head (git-496fd5d 2011-04-10 23-16-56 +0200)

But when i load the module i get the follow error:

freeswitch at internal> load mod_odbc_query
+OK Reloading XML
-ERR [module load file routine returned an error]

freeswitch at internal> 2011-04-28 18:41:22.665848 [INFO] mod_enum.c:755 ENUM Reloaded
2011-04-28 18:41:22.665848 [CRIT] switch_loadable_module.c:928 Error Loading module /usr/local/freeswitch/mod/mod_odbc_query.so
**/usr/local/freeswitch/mod/mod_odbc_query.so: invalid ELF header**
2011-04-28 18:41:22.665848 [INFO] switch_time.c:999 Timezone reloaded 530 definitions


My steps are:

mkdir /tmp/fs
cd /tmp/fs
git clone git://git.freeswitch.org/freeswitch-contrib.git
cd freeswitch-contrib/ledr/c/
mv mod_odbc_query/ /usr/src/freeswitch/src/mod/xml_int
cd /usr/src/freeswitch/src/mod/xml_in/mod_odbc_query
make
cp mod_odbc_query.so /usr/local/freeswitch/mod/
cp odbc_query.conf.xml /usr/local/freeswitch/conf/autoload_config

What are doing wrong?

Regards.


From krice at freeswitch.org  Fri Apr 29 02:53:56 2011
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 28 Apr 2011 17:53:56 -0500
Subject: [Freeswitch-users] Load mod_odbc_query error
In-Reply-To: <391041.29065.qm@web34308.mail.mud.yahoo.com>
Message-ID: <C9DF5A34.1492E%krice@freeswitch.org>

>From your mod_odbc_query directory do a make install.

The .so file you cp'd out is actually a special text file for running FS in
developer debug mode from the source tree

K


On 4/28/11 5:43 PM, "FERNANDO VILLARROEL" <fvillarroel at yahoo.com> wrote:

> Hi all.
> 
> I getting mod_odbc_query from contrib.
> 
> My FS Version:
> 
> FreeSWITCH Version 1.0.head (git-496fd5d 2011-04-10 23-16-56 +0200)
> 
> But when i load the module i get the follow error:
> 
> freeswitch at internal> load mod_odbc_query
> +OK Reloading XML
> -ERR [module load file routine returned an error]
> 
> freeswitch at internal> 2011-04-28 18:41:22.665848 [INFO] mod_enum.c:755 ENUM
> Reloaded
> 2011-04-28 18:41:22.665848 [CRIT] switch_loadable_module.c:928 Error Loading
> module /usr/local/freeswitch/mod/mod_odbc_query.so
> **/usr/local/freeswitch/mod/mod_odbc_query.so: invalid ELF header**
> 2011-04-28 18:41:22.665848 [INFO] switch_time.c:999 Timezone reloaded 530
> definitions
> 
> 
> My steps are:
> 
> mkdir /tmp/fs
> cd /tmp/fs
> git clone git://git.freeswitch.org/freeswitch-contrib.git
> cd freeswitch-contrib/ledr/c/
> mv mod_odbc_query/ /usr/src/freeswitch/src/mod/xml_int
> cd /usr/src/freeswitch/src/mod/xml_in/mod_odbc_query
> make
> cp mod_odbc_query.so /usr/local/freeswitch/mod/
> cp odbc_query.conf.xml /usr/local/freeswitch/conf/autoload_config
> 
> What are doing wrong?
> 
> 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




From msc at freeswitch.org  Fri Apr 29 03:02:22 2011
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 28 Apr 2011 16:02:22 -0700
Subject: [Freeswitch-users] Call Hold scenario, please help.
In-Reply-To: <283095.73945.qm@web114505.mail.gq1.yahoo.com>
References: <597945.23381.qm@web114503.mail.gq1.yahoo.com>
	<201104281331.09281.justlikeef@gmail.com>
	<283095.73945.qm@web114505.mail.gq1.yahoo.com>
Message-ID: <BANLkTi=2uA3k_0bV0oVj12dUbGtLSQ4jXA@mail.gmail.com>

On Thu, Apr 28, 2011 at 1:24 PM, Dave Bracken <bracken_dave at yahoo.com>wrote:

> Rob,
> Thanks for the thought, but i am already using the park command. But after
> i park the call, i cannot get the music to play.(trying to use the playback
> command) Also, i am not getting dial tone on the operator phone, so i am not
> sure the line has been fully disconnected to free the operator up for other
> calls.
> i was hoping someone had an ordered list of the commands to complete my
> original scenario.
> thanks in advance
> Dave
>
Did you already pastebin your relevant dialplan config and debug output w/
sip trace? I didn't see any PB entries with your name or nick...

-MC
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From fvillarroel at yahoo.com  Fri Apr 29 04:42:50 2011
From: fvillarroel at yahoo.com (FERNANDO VILLARROEL)
Date: Thu, 28 Apr 2011 17:42:50 -0700 (PDT)
Subject: [Freeswitch-users] Load mod_odbc_query error SOLVED
In-Reply-To: <C9DF5A34.1492E%krice@freeswitch.org>
Message-ID: <852791.9119.qm@web34302.mail.mud.yahoo.com>

Thank You Ken

--- On Thu, 4/28/11, Ken Rice <krice at freeswitch.org> wrote:

> From: Ken Rice <krice at freeswitch.org>
> Subject: Re: [Freeswitch-users] Load mod_odbc_query error
> To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
> Date: Thursday, April 28, 2011, 7:53 PM
> >From your mod_odbc_query
> directory do a make install.
> 
> The .so file you cp'd out is actually a special text file
> for running FS in
> developer debug mode from the source tree
> 
> K
> 
> 
> On 4/28/11 5:43 PM, "FERNANDO VILLARROEL" <fvillarroel at yahoo.com>
> wrote:
> 
> > Hi all.
> > 
> > I getting mod_odbc_query from contrib.
> > 
> > My FS Version:
> > 
> > FreeSWITCH Version 1.0.head (git-496fd5d 2011-04-10
> 23-16-56 +0200)
> > 
> > But when i load the module i get the follow error:
> > 
> > freeswitch at internal> load mod_odbc_query
> > +OK Reloading XML
> > -ERR [module load file routine returned an error]
> > 
> > freeswitch at internal> 2011-04-28 18:41:22.665848
> [INFO] mod_enum.c:755 ENUM
> > Reloaded
> > 2011-04-28 18:41:22.665848 [CRIT]
> switch_loadable_module.c:928 Error Loading
> > module /usr/local/freeswitch/mod/mod_odbc_query.so
> > **/usr/local/freeswitch/mod/mod_odbc_query.so: invalid
> ELF header**
> > 2011-04-28 18:41:22.665848 [INFO] switch_time.c:999
> Timezone reloaded 530
> > definitions
> > 
> > 
> > My steps are:
> > 
> > mkdir /tmp/fs
> > cd /tmp/fs
> > git clone
> git://git.freeswitch.org/freeswitch-contrib.git
> > cd freeswitch-contrib/ledr/c/
> > mv mod_odbc_query/
> /usr/src/freeswitch/src/mod/xml_int
> > cd /usr/src/freeswitch/src/mod/xml_in/mod_odbc_query
> > make
> > cp mod_odbc_query.so /usr/local/freeswitch/mod/
> > cp odbc_query.conf.xml
> /usr/local/freeswitch/conf/autoload_config
> > 
> > What are doing wrong?
> > 
> > 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
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From fieldpeak at gmail.com  Fri Apr 29 09:04:11 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Fri, 29 Apr 2011 13:04:11 +0800
Subject: [Freeswitch-users] FS - SIP profiles crashed
Message-ID: <BANLkTi=402RnR54vQmVx7_fh8z1iaX3aFw@mail.gmail.com>

Hi friendds,

the FS running for serval days, today when i checked the "sofia status",
there is no any sip profile existed...
then i checked the log as below (sip profile attached), there is errors when
creating the sip profiles, and FS detected IP changed to 0.0.0.0 before
that, it is very strange...
appreciated if anyone provide any hints, thanks.

Regards,
Charles


2011-04-29 08:16:06.484375 [INFO] mod_sofia.c:4437 EVENT_TRAP: IP change
detected
2011-04-29 08:16:06.484375 [INFO] mod_sofia.c:4438 IP change detected
[172.28.172.128]->[0.0.0.0] []->[]
2011-04-29 08:16:06.671875 [DEBUG] sofia.c:1614 Write lock internalk
2011-04-29 08:16:06.671875 [NOTICE] sofia.c:1621 Waiting for worker thread
2011-04-29 08:16:06.671875 [DEBUG] sofia.c:1678 Write unlock internal
2011-04-29 08:16:06.906250 [DEBUG] sofia.c:1614 Write lock external
2011-04-29 08:16:06.906250 [NOTICE] sofia.c:1621 Waiting for worker thread
2011-04-29 08:16:06.937500 [DEBUG] sofia.c:1614 Write lock internal_6060
2011-04-29 08:16:06.937500 [NOTICE] sofia.c:1621 Waiting for worker thread
2011-04-29 08:16:06.937500 [DEBUG] sofia.c:1678 Write unlock internal_6060
2011-04-29 08:16:07.140625 [DEBUG] sofia.c:1678 Write unlock external
2011-04-29 08:16:07.203125 [NOTICE] sofia_glue.c:4828 Reload XML [Success]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 debug [0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 debug [0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 debug [0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 sip-trace [no]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 sip-trace [no]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 sip-trace [no]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 watchdog-enabled [no]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 watchdog-enabled [no]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 watchdog-step-timeout
[30000]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rfc2833-pt [101]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 watchdog-step-timeout
[30000]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 watchdog-event-timeout
[30000]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 sip-port [5080]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 watchdog-event-timeout
[30000]
2011-04-29 08:16:07.203125 [INFO] mod_pocketsphinx.c:482 PocketSphinx
Reloaded
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 log-auth-failures [true]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 dialplan [XML]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 log-auth-failures [true]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012
forward-unsolicited-mwi-notify [false]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 context [public]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012
forward-unsolicited-mwi-notify [false]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 context [default]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 dtmf-duration [2000]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 context [default]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rfc2833-pt [101]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 inbound-codec-prefs
[G722,PCMU,PCMA,GSM]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rfc2833-pt [101]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 sip-port [6060]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 outbound-codec-prefs
[G722,PCMU,PCMA,GSM]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 sip-port [6010]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 dialplan [XML]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 hold-music
[local_stream://moh]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 dialplan [XML]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 dialplan [XML]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 dtmf-duration [2000]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rtp-timer-name [soft]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 dtmf-duration [2000]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 inbound-codec-prefs
[G722,PCMU,PCMA,GSM]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 local-network-acl
[localnet.auto]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 inbound-codec-prefs
[G722,PCMU,PCMA,GSM]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 outbound-codec-prefs
[G722,PCMU,PCMA,GSM]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 manage-presence [false]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 outbound-codec-prefs
[G722,PCMU,PCMA,GSM]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rtp-timer-name [soft]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 inbound-codec-negotiation
[generous]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rtp-ip [0.0.0.0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rtp-timer-name [soft]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 nonce-ttl [60]
2011-04-29 08:16:07.203125 [WARNING] sofia.c:3260 Invalid IP 0.0.0.0
replaced with 172.28.172.128
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rtp-ip [0.0.0.0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 sip-ip [0.0.0.0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 auth-calls [false]
2011-04-29 08:16:07.203125 [WARNING] sofia.c:3260 Invalid IP 0.0.0.0
replaced with 172.28.172.128
2011-04-29 08:16:07.203125 [WARNING] sofia.c:3273 Invalid IP 0.0.0.0
replaced with 172.28.172.128
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rtp-ip [0.0.0.0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 sip-ip [0.0.0.0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 hold-music
[local_stream://moh]
2011-04-29 08:16:07.203125 [WARNING] sofia.c:3260 Invalid IP 0.0.0.0
replaced with 172.28.172.128
2011-04-29 08:16:07.203125 [WARNING] sofia.c:3273 Invalid IP 0.0.0.0
replaced with 172.28.172.128
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 apply-nat-acl [nat.auto]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 sip-ip [0.0.0.0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 hold-music
[local_stream://moh]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 apply-inbound-acl [
192.168.200.0/24]
2011-04-29 08:16:07.203125 [WARNING] sofia.c:3273 Invalid IP 0.0.0.0
replaced with 172.28.172.128
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 apply-nat-acl [nat.auto]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 local-network-acl
[localnet.auto]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 ext-rtp-ip [60.216.3.72]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 apply-inbound-acl [
172.28.0.0/16]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 record-path
[c:\vswitch\recordings]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 local-network-acl
[localnet.auto]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 ext-sip-ip [60.216.3.72]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 record-template
[${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 record-path
[c:\vswitch\recordings]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rtp-timeout-sec [300]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 manage-presence [false]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 record-template
[${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rtp-hold-timeout-sec [1800]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 inbound-codec-negotiation
[generous]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 manage-presence [false]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls [false]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls [false]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 inbound-codec-negotiation
[generous]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls-bind-params
[transport=tls]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls-bind-params
[transport=tls]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls [false]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls-sip-port [5081]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls-sip-port [5061]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls-bind-params
[transport=tls]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls-cert-dir
[c:\vswitch/conf/ssl]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls-cert-dir
[c:\vswitch/conf/ssl]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls-sip-port [5061]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls-version [tlsv1]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls-version [tlsv1]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls-cert-dir
[c:\vswitch/conf/ssl]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 inbound-bypass-media [false]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 tls-version [tlsv1]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 inbound-proxy-media [true]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 inbound-bypass-media [false]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 nonce-ttl [60]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 inbound-proxy-media [true]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012
NDLB-received-in-nat-reg-contact [true]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 nonce-ttl [60]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 auth-calls [true]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012
NDLB-received-in-nat-reg-contact [true]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012
inbound-reg-force-matching-username [true]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 auth-calls [true]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 auth-all-packets [false]
2011-04-29 08:16:07.203125 [ERR] sofia.c:2051 ERROR: username param is
REQUIRED!
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012
inbound-reg-force-matching-username [true]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 ext-rtp-ip [60.216.3.72]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 auth-all-packets [false]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 ext-sip-ip [60.216.3.72]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 ext-rtp-ip [auto-nat]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rtp-timeout-sec [300]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 ext-sip-ip [auto-nat]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rtp-hold-timeout-sec [1800]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rtp-timeout-sec [300]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 force-register-domain
[0.0.0.0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 rtp-hold-timeout-sec [1800]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 force-subscription-domain
[0.0.0.0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 force-register-domain
[0.0.0.0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 force-register-db-domain
[0.0.0.0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 force-subscription-domain
[0.0.0.0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 NDLB-force-rport [true]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 force-register-db-domain
[0.0.0.0]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 challenge-realm [auto_from]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 NDLB-force-rport [true]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:3012 challenge-realm [auto_from]
2011-04-29 08:16:07.203125 [NOTICE] sofia.c:3883 Started Profile external
[sofia_reg_external]
2011-04-29 08:16:07.203125 [NOTICE] sofia.c:2219 Adding Alias [0.0.0.0] for
profile [internal_6060]
2011-04-29 08:16:07.203125 [WARNING] sofia.c:2221 Alias [0.0.0.0] for
profile [internal] (already exists)
2011-04-29 08:16:07.203125 [INFO] switch_time.c:950 Timezone reloaded 530
definitions
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:1430 Creating agent for external
2011-04-29 08:16:07.203125 [NOTICE] sofia.c:3883 Started Profile
internal_6060 [sofia_reg_internal_6060]
2011-04-29 08:16:07.203125 [NOTICE] sofia.c:3883 Started Profile internal
[sofia_reg_internal]
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:1430 Creating agent for
internal_6060
2011-04-29 08:16:07.203125 [DEBUG] sofia.c:1430 Creating agent for internal
2011-04-29 08:16:07.265625 [ERR] sofia.c:1492 Error Creating SIP UA for
profile: external
2011-04-29 08:16:07.265625 [ERR] sofia.c:1492 Error Creating SIP UA for
profile: internal_6060
2011-04-29 08:16:07.265625 [ERR] sofia.c:1492 Error Creating SIP UA for
profile: internal
2011-04-29 08:17:06.484375 [INFO] mod_sofia.c:4437 EVENT_TRAP: IP change
detected
2011-04-29 08:17:06.484375 [INFO] mod_sofia.c:4438 IP change detected
[0.0.0.0]->[172.28.172.128] []->[]
2011-04-29 08:17:07.187500 [NOTICE] sofia_glue.c:4828 Reload XML [Success]
2011-04-29 08:17:07.187500 [INFO] mod_pocketsphinx.c:482 PocketSphinx
Reloaded
2011-04-29 08:17:07.187500 [INFO] switch_time.c:950 Timezone reloaded 530
definitions
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From bogusmaster at o2.pl  Fri Apr 29 12:31:18 2011
From: bogusmaster at o2.pl (bogusmaster at o2.pl)
Date: Fri, 29 Apr 2011 10:31:18 +0200
Subject: [Freeswitch-users] problem with transferring calls when using ivr
Message-ID: <op.vuoxye19agcvls@master>


Hi all !

I'm a freeswitch newbie taking care of installation that I've recently
taken over from a colleague who left our company.
Currently I'm having a problem with transferring calls that went thru ivr
menu.

The call scenario looks like this: external number A calls our external
number E. Then, in ivr menu it selects some option. After that A is
connected with internal number I.
When internal number I tries to transfer call from A to another internal
number I2, call hangs up.

If I turn the ivr off, as it is commented in the excerpt from
/conf/dialplan/public/01_freephone_inbound.xml below, transferring calls
works fine.

     <extension name="general_inbound">
       <condition field="destination_number"
expression="^(freephone_XXXX_in)$">
         <action application="info"/>
         <action application="transfer" data="100 XML default"/>
         <!--action application="javascript" data="simpleivr.js"/-->
       </condition>
     </extension>

Below is my ivr script:

//====================================================================================
// Function :: Debug output
//====================================================================================
function Debug(text)
{
    var uuid_last5 = session.uuid.substr(-5)
    console_log("info", "["+uuid_last5+"] "+text+"\n");
}

//====================================================================================
// Function :: Callback handlers for collectInput
//             Returning false from these handlers causes the digit  
collection
//             to "unblock".  Depending on what behavior you want to  
achieve
//             you might want to unblock on any input or on a specific  
digit.
//====================================================================================
function Get_DTMF_Stop_On_Any(session, type, data, arg)
{
    Debug("DTMF: "+data.digit+"");
    arg.digits += data.digit;
    return false;
}


//=====================================================================================================================
//=====================================================================================================================
//=====================================================================================================================
//=====================================================================================================================


//====================================================================================
// Main Section
//====================================================================================
var dtmf_digits = new Object();

session.answer();
Debug("Playing welcome message...");
session.streamFile("/opt/freeswitch/sounds/mlife/ivr/welcome.wav");

dtmf_digits.digits = "";
var repeat = 0;
while (session.ready() && dtmf_digits.digits == "" && repeat < 3) {
      //------------------------------------------------------
      // Flush digits, ask for menu selection
      //------------------------------------------------------
      dtmf_digits.digits = "";
      session.flushDigits();

      Debug("Playing announcement...");
      session.streamFile("/opt/freeswitch/sounds/mlife/ivr/ann.wav",  
Get_DTMF_Stop_On_Any, dtmf_digits);
      if(dtmf_digits.digits == "") {
         session.collectInput(Get_DTMF_Stop_On_Any, dtmf_digits, 3000);
      }
      Debug("Menu Choice Selected = [" + dtmf_digits.digits + "]");

      if (dtmf_digits.digits == "") {
         repeat++;
         if (repeat > 2) {
             Debug("Max tries reached");
         	break;
         }
      } else {
      	if (dtmf_digits.digits == "1") {
      		session.execute("execute_extension", "100 XML default");
      	} else if (dtmf_digits.digits == "2") {
      		session.execute("execute_extension", "200 XML default");
      	} else if (dtmf_digits.digits == "3") {
			session.execute("execute_extension", "401 XML default");
      	} else if (dtmf_digits.digits == "4") {
			//session.execute("transfer", "ivr-support");
			Debug("Playing warning message about recording...");
			session.streamFile("/opt/freeswitch/sounds/mlife/ivr/support.wav");
			session.execute("execute_extension", "tech-support XML default");
      	}
      }
}

// User exceeded the ivr prompt limit, say bye-bye
// session.streamFile("conference/mlife/8000/conf-goodbye.wav");
session.hangup();

Any hints would be greatly appreciated.

Regards,
Bart

-- 
Using Opera's revolutionary email client: http://www.opera.com/mail/


From arielmonaco at flylabs.com  Fri Apr 29 03:15:04 2011
From: arielmonaco at flylabs.com (Ariel Monaco)
Date: Thu, 28 Apr 2011 20:15:04 -0300
Subject: [Freeswitch-users] Script para chequear un numero de destino
In-Reply-To: <BANLkTindwkD3NKDFCjFq1XaKx+aCjiee1Q@mail.gmail.com>
References: <BANLkTi=Lz=gBRrcX_WmYN3iZXhzBv4wSPA@mail.gmail.com>
	<968175.88614.qm@web34306.mail.mud.yahoo.com>
	<BANLkTi=ZFv8D66r2SmV5VXQOvKP1uKX9pQ@mail.gmail.com>
	<BANLkTimfcULSHkf9Zo5k+VugLS+FAKzHWg@mail.gmail.com>
	<BANLkTin0zS4CWyOC1a3e4TcHUmy=AA=u-g@mail.gmail.com>
	<BANLkTindwkD3NKDFCjFq1XaKx+aCjiee1Q@mail.gmail.com>
Message-ID: <16C3EAE5-01AC-4BBC-B9EB-EE0210A938A7@flylabs.com>

Let's make clear that this is not a limitation in Lua. You can also use Lua to connect to the Event Socket.

Go to libs/esl and type 'make luamod'

What you seem to be talking about is Mod_Lua and not the programming language iself.

My 2 cents,

On Apr 28, 2011, at 18:18 , Michael Collins wrote:

> 
> 
> 2011/4/28 Jo?o Mesquita <jmesquita at freeswitch.org>
> Hello Juan,
> 
> By the Iba?ez, I would assume you're also a spanish speaking person?
> 
> IMHO, ESL and LUA for small scale applications is a matter of preference. They can accomplish the same. I do not program in LUA very well, so I normally prefer ESL, but I take it can be a bit more tricky since you need to control the full call flow.
> 
> On the other hand, if you are talking about massive IVRs and such, I would definitely suggest ESL since you can run it on another box and leave all the processing power to freeswitch where freeswitch is hosted. LUA cant to that, it needs to run on the same box and you mix the logic where CPU is very sensitive.
> 
> Does that make it clearer for you?
> 
> Fernando, como estamos con la lectura? En que mas te puedo ayudar?
> 
> Regards,
> Jo?o Mesquita
> 
> 
> I agree with J. Lua is a great choice for doing simple things from the dialplan. Even basic IVRs can be done effectively with dialplan scripts in Lua. However, if you need to scale, or if your program needs to be able to react to system events, then ESL is probably the way to go. The good news about going with ESL is that there is quite literally nothing that a Lua dialplan script can do that an ESL program cannot. The only real difference is that Lua dp scripts are simpler and more limited than an ESL-based program.
> 
> -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

Ariel Monaco ? Systems Engineer
Flylabs - Open Source Telecommunications and  IT Consultants

Address: Potos? 4456 C1199ACP - Buenos Aires - Argentina
Web: http://flylabs.com
E-Mail: arielmonaco at flylabs.com
Tel. +54 (11) 4982-2689, +1 (315) 688-7333

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From moises.silva at gmail.com  Fri Apr 29 13:52:38 2011
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 29 Apr 2011 11:52:38 +0200
Subject: [Freeswitch-users] Freetdm - Wanpipe - ftmod_sangoma_isdn - no
	caller_name
In-Reply-To: <32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A46A@jehuty.supportkids.com>
References: <32D3DDAA3243F64CAD1EEF165D2BC3F01B11A3A46A@jehuty.supportkids.com>
Message-ID: <BANLkTinP9Ko0TsK2crwsb8P69ANY93eZKg@mail.gmail.com>

On Wed, Apr 27, 2011 at 4:42 PM, Dennis Young
<Dennis.Young at supportkids.com>wrote:

> Setting "facility" to "yes" caused the following errors.   However I do see
> the caller_number on the second line and the caller_name in fifth line, but
> as you can tell it can't make out what the DNIS/DID number is.  The
> switchtype is set to "national" just like it was set with boost.
>
> With "facility" set to "no", everything works except no caller_name.
>
> Any ideas?
>
>
Hello Denis,

Can you do the following please?:

1. Stop FreeSWITCH.
2. Remove the freeswitch.log file (typically under
/usr/local/freeswitch/log/freeswitch.log
3. Enable debug logging.
4. Start FreeSWITCH (loading FreeTDM)
5. Place a call (I'm assuming a single call triggers this error, if not,
then keep placing calls until the error happens).
6. Pastebin the freeswitch.log file

You can also contact techdesk at sangoma.com if you want a support engineer to
take a closer look at your configuration.

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



> 2011-04-26 20:57:12.685250 [INFO] ftmod_sangoma_isdn_stack_rcv.c:75
> [s1c1][1:1] Received SETUP (suId:1 suInstId:0 spInstId:1)
> 2011-04-26 20:57:12.685250 [INFO] ftmod_sangoma_isdn_stack_hndl.c:142
> [s1c1][1:1] Incoming call: Called No:[6025] Calling No:[5127518487]
> 2011-04-26 20:57:12.685250 [CRIT] ftdm_io.c:5993 Overwriting non-cleared
> call-id
> f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.685250 [NOTICE]
> switch_channel.c:816 New Channel
> FreeTDM/1:1/?????????????????????????????????????????????????????????????????????????????????
> [f1e75758-8028-4e07-a118-247c49cbe85c]
> 2011-04-26 20:57:12.700875 [INFO] ftmod_sangoma_isdn_stack_out.c:165
> [s1c1][1:1] Sending PROCEED (suId:1 suInstId:1 spInstId:1 dchan:1 ces:0)
> f1e75758-8028-4e07-a118-247c49cbe85c
> 2011-04-26 20:57:12.700875 [INFO] mod_dialplan_xml.c:331 Processing YOUNG
> DENNIS
> ????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????
> <???????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????>->?????????????????????????????????????????????????????????????????????????????????
> in context public
> f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.700875 [NOTICE]
> mod_dptools.c:1184 outside_call=true, ringback=%(2000,4000,440.0,480.0)
> 2011-04-26 20:57:12.700875 [INFO] ftmod_sangoma_isdn_stack_out.c:223
> [s1c1][1:1] Sending ALERT (suId:1 suInstId:1 spInstId:1 dchan:1 ces:0)
> f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.700875 [NOTICE]
> mod_dptools.c:698 Ring Ready
> FreeTDM/1:1/?????????????????????????????????????????????????????????????????????????????????!
> f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.700875 [NOTICE]
> mod_dptools.c:698 Ring-Ready
> FreeTDM/1:1/?????????????????????????????????????????????????????????????????????????????????!
> f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.716500 [NOTICE]
> switch_core_state_machine.c:189
> FreeTDM/1:1/?????????????????????????????????????????????????????????????????????????????????
> has executed the last dialplan instruction, hanging up.
> f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.716500 [NOTICE]
> switch_core_state_machine.c:191 Hangup
> FreeTDM/1:1/?????????????????????????????????????????????????????????????????????????????????
> [CS_EXECUTE] [NORMAL_CLEARING]
> 2011-04-26 20:57:12.716500 [INFO] ftmod_sangoma_isdn_stack_out.c:373
> [s1c1][1:1] Sending DISCONNECT (suId:1 suInstId:1 spInstId:1)
> f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.716500 [NOTICE]
> switch_core_session.c:1304 Session 1
> (FreeTDM/1:1/?????????????????????????????????????????????????????????????????????????????????)
> Ended
> f1e75758-8028-4e07-a118-247c49cbe85c 2011-04-26 20:57:12.716500 [NOTICE]
> switch_core_session.c:1306 Close Channel
> FreeTDM/1:1/?????????????????????????????????????????????????????????????????????????????????
> [CS_DESTROY]
> 2011-04-26 20:57:12.747750 [INFO] ftmod_sangoma_isdn_stack_rcv.c:245
> [s1c1][1:1] Received RELEASE/RELEASE COMPLETE (suId:1 suInstId:1 spInstId:1)
> 2011-04-26 20:57:12.747750 [CRIT] ftdm_io.c:6025 Cannot clear call with
> invalid call-id
>
>
> ...dly
>
> -----Original Message-----
> From: Brian West [mailto:brian at freeswitch.org]
> Sent: Tuesday, April 26, 2011 6:47 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Freetdm - Wanpipe - ftmod_sangoma_isdn
>
> Set this to YES
>
> /b
>
> On Apr 26, 2011, at 3:45 PM, Dennis Young wrote:
>
> > <param name="facility" value="no" />
>
>
>
>
> Notice: This electronic transmission and its attachments are confidential
> and protected by applicable state and/or federal law.  Any use, reading,
> dissemination, distribution, copying or storage of this information by
> anyone other than the intended recipient is strictly prohibited.  If you are
> not the intended recipient, please immediately notify the sender by return
> email or telephone and delete this message and its attachments from your
> system.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From frankie.k.yiu at gmail.com  Fri Apr 29 14:19:41 2011
From: frankie.k.yiu at gmail.com (Frankie Yiu)
Date: Fri, 29 Apr 2011 03:19:41 -0700
Subject: [Freeswitch-users] error LNK2019: unresolved external symbol when
 building on x64 machine but not in Win32 machine
Message-ID: <BANLkTinO3zneiUSTB-rOg+tD+cVKiZvS=w@mail.gmail.com>

Hi there,

I have a module written in c++ and it compiled and ran successfully on a 32
bit Windows Server 2003 machine with the Visual Studio 2008.  However, when
I compiled it on a x64 machine (Windows7 or Window Server) with Visual
Studio 2008, I have the following external symbol errors.  Does anyone have
an idea why this is happening and how to fix it?  Thank you.

1>   Creating library Win32\Debug/mod_baw_2008.lib and object
Win32\Debug/mod_baw_2008.exp
1>mod_baw.obj : error LNK2019: unresolved external symbol
__imp__switch_event_bind at 20 referenced in function _mod_baw_load
1>mod_baw.obj : error LNK2019: unresolved external symbol
__imp__switch_log_printf referenced in function _mod_baw_load
1>mod_baw.obj : error LNK2019: unresolved external symbol
__imp__switch_loadable_module_create_module_interface at 8 referenced in
function _mod_baw_load
1>mod_baw.obj : error LNK2019: unresolved external symbol
__imp__switch_event_get_body at 4 referenced in function
_Answered_Event_Callback
1>mod_baw.obj : error LNK2019: unresolved external symbol
__imp__switch_core_session_rwunlock at 4 referenced in function
_Answered_Event_Callback
1>mod_baw.obj : error LNK2019: unresolved external symbol
__imp__switch_ivr_record_session at 16 referenced in function
_Answered_Event_Callback
1>mod_baw.obj : error LNK2019: unresolved external symbol
__imp__switch_core_media_bug_add at 36 referenced in function
_Answered_Event_Callback
1>mod_baw.obj : error LNK2019: unresolved external symbol
__imp__switch_core_session_locate at 4 referenced in function
_Answered_Event_Callback
1>mod_baw.obj : error LNK2019: unresolved external symbol
__imp__switch_event_get_header at 8 referenced in function
_Answered_Event_Callback
1>mod_baw.obj : error LNK2019: unresolved external symbol
__imp__switch_core_media_bug_get_write_replace_frame at 4 referenced in
function _DataFrame_CallBack
1>mod_baw.obj : error LNK2019: unresolved external symbol
__imp__switch_core_media_bug_get_read_replace_frame at 4 referenced in function
_DataFrame_CallBack
1>mod_baw.obj : error LNK2019: unresolved external symbol
__imp__switch_core_session_get_uuid at 4 referenced in function
_DataFrame_CallBack
1>mod_baw.obj : error LNK2019: unresolved external symbol
__imp__switch_core_media_bug_get_session at 4 referenced in function
_DataFrame_CallBack
1>C:\SIPServices\freeswitch\Debug/mod/mod_baw.dll : fatal error LNK1120: 13
unresolved externals


Thanks,

Frankie
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From anton.vazir at gmail.com  Fri Apr 29 14:24:53 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Fri, 29 Apr 2011 15:24:53 +0500
Subject: [Freeswitch-users] SIP external profile - Error looking up host
	name
In-Reply-To: <BANLkTinaJ66c3VS-rM7GjUZvhSh_wQ=aFg@mail.gmail.com>
References: <BANLkTinaJ66c3VS-rM7GjUZvhSh_wQ=aFg@mail.gmail.com>
Message-ID: <BANLkTimZSPiHS0Lm5P1Jv=obbSJpqK1Opg@mail.gmail.com>

anyone?


From anton.vazir at gmail.com  Fri Apr 29 15:19:24 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Fri, 29 Apr 2011 16:19:24 +0500
Subject: [Freeswitch-users] SIP external profile - Error looking up host
	name
In-Reply-To: <BANLkTimZSPiHS0Lm5P1Jv=obbSJpqK1Opg@mail.gmail.com>
References: <BANLkTinaJ66c3VS-rM7GjUZvhSh_wQ=aFg@mail.gmail.com>
	<BANLkTimZSPiHS0Lm5P1Jv=obbSJpqK1Opg@mail.gmail.com>
Message-ID: <BANLkTinKyT_ya1enQTnOeLB5QDFVWrH=bw@mail.gmail.com>

Looks fixed...

commit bc19d28310cf322eab23b93c5b6152c7390c4c25
Author: Marc Olivier Chouinard <mochouinard at moctel.com>
Date:   Mon Apr 25 15:53:54 2011 -0400

    Fix regression to dns resolve in sofia.  It will make sofia and
compiler happy


From zetruger at gmail.com  Fri Apr 29 15:28:25 2011
From: zetruger at gmail.com (=?KOI8-R?B?6dfBziD+ydPU0cvP1w==?=)
Date: Fri, 29 Apr 2011 15:28:25 +0400
Subject: [Freeswitch-users] mod_radius_cdr and Acct-Delay-Time
In-Reply-To: <9b86b6d280317812f3b7de59136aa457@mail.gmail.com>
References: <9b86b6d280317812f3b7de59136aa457@mail.gmail.com>
Message-ID: <BANLkTi=Dxk4pdG7nMDHugEjoOhpDsEqVYQ@mail.gmail.com>

Just fix mod_radius_cdr )) change format of Freeswitch-Callstartdat.
Use PW_ACCT_DELAY_TIME for adding Acct-Delay-Time field.

2011/4/27 Camila Troncoso <ctroncoso at redvoiss.net>:
>
>
> Hi,
>
>
>
> I?m working with radius accounting using the mod_radius_cdr module. I?m
> having trouble with the date format that Radius server receive. ?An example
> of what the radius server receive is:
>
>
>
> ?Freeswitch-Callstartdate = "2011-04-21T18:23:59.780945-0400"?
>
>
>
> This date format is very difficult to read and I want to change it? to make
> the accounting easier. I search all around for some param or configuration
> that allows me to do so, but I only find that this date format is define in
> mod_radius_cdr.c module.
>
>
>
> I?m also having problems with the parameter Acct-Delay-Time, it is not
> increasing when the client resend the radius packet. I read the buildreq.c
> code but didn?t find the problem.
>
>
>
> Please some help with this matter.
>
>
>
> Regards,
>
>
>
> Camila Troncoso |?Ingeniero de Desarrollo
>
> RedVoiss |ctroncoso at redvoiss.net
>
> Santiago - Chile | +56 2 2408535
>
> www.redvoiss.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
>
>


From zetruger at gmail.com  Fri Apr 29 15:43:07 2011
From: zetruger at gmail.com (=?KOI8-R?B?6dfBziD+ydPU0cvP1w==?=)
Date: Fri, 29 Apr 2011 15:43:07 +0400
Subject: [Freeswitch-users] FreeSWITCH with remote Music on Hold server.
 How to do it?
In-Reply-To: <BANLkTimB=Sa6hZKm6Xw2LVwrG1UdP+A5Yg@mail.gmail.com>
References: <BANLkTikCefhDoiDrx5=SaYi=OX73KxL3JA@mail.gmail.com>
	<BANLkTiniO40i6Q3LzEVd9+oup1MGQyCnYw@mail.gmail.com>
	<BANLkTik-1hnNbLBze9H2mHjaRHNRmWz9Mw@mail.gmail.com>
	<BANLkTimB=Sa6hZKm6Xw2LVwrG1UdP+A5Yg@mail.gmail.com>
Message-ID: <BANLkTinCQ1qkSrTisCz0oW1qO6ZK=eiV_Q@mail.gmail.com>

Bypass all the media, including MOH.

27 ?????? 2011??. 18:02 ???????????? Antonio Teixeira
<eagle.antonio at gmail.com> ???????:
> Well in that case you need to explain further what are you requirements so
> the list can help you out.
>
> Anyway so you want to bypass freeswitch in all the media ? or just the MOH ?
> I'm assuming you don't need transcoding on the FS side.
>
>
>
> 2011/4/27 ???? ???????? <zetruger at gmail.com>
>>
>> RTP data must be transmitted directly between Subscriber and MoH server.
>> Obviously, that mod_shout transmits data through FreeSWITCH for mp3
>> decoding.
>>
>> ,
>> 2011/4/27 Antonio Teixeira <eagle.antonio at gmail.com>:
>> > Checks The Docs / Wiki
>> >
>> > http://wiki.freeswitch.org/wiki/Mod_shout
>> >
>> >
>> > 2011/4/27 ???? ???????? <zetruger at gmail.com>
>> >>
>> >> FreeSWITCH with remote Music on Hold server. How to do it?
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From juanito1982 at gmail.com  Fri Apr 29 15:50:09 2011
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Fri, 29 Apr 2011 13:50:09 +0200
Subject: [Freeswitch-users] Script para chequear un numero de destino
In-Reply-To: <BANLkTin0zS4CWyOC1a3e4TcHUmy=AA=u-g@mail.gmail.com>
References: <BANLkTi=Lz=gBRrcX_WmYN3iZXhzBv4wSPA@mail.gmail.com>
	<968175.88614.qm@web34306.mail.mud.yahoo.com>
	<BANLkTi=ZFv8D66r2SmV5VXQOvKP1uKX9pQ@mail.gmail.com>
	<BANLkTimfcULSHkf9Zo5k+VugLS+FAKzHWg@mail.gmail.com>
	<BANLkTin0zS4CWyOC1a3e4TcHUmy=AA=u-g@mail.gmail.com>
Message-ID: <BANLkTikKhspQvWPHd8BEnYPAsmUoh7Hwkw@mail.gmail.com>

2011/4/28 Jo?o Mesquita <jmesquita at freeswitch.org>

> Hello Juan,
>
> By the Iba?ez, I would assume you're also a spanish speaking person?
>

You are OK


>
> IMHO, ESL and LUA for small scale applications is a matter of preference.
> They can accomplish the same. I do not program in LUA very well, so I
> normally prefer ESL, but I take it can be a bit more tricky since you need
> to control the full call flow.
>
> On the other hand, if you are talking about massive IVRs and such, I would
> definitely suggest ESL since you can run it on another box and leave all the
> processing power to freeswitch where freeswitch is hosted. LUA cant to that,
> it needs to run on the same box and you mix the logic where CPU is very
> sensitive.
>
> Does that make it clearer for you?
>

Yes. I have not tested ESL already. I am going to do some tests.

Regards
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From boris at tagnet.ru  Fri Apr 29 18:42:28 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 29 Apr 2011 20:42:28 +0600
Subject: [Freeswitch-users] rfc2833-pt
Message-ID: <4DBACE54.30809@tagnet.ru>

Hello!

     In the Audiocodecs MP-118 User's Manual "The RFC 2833 DTMF relay 
dynamic payload type. The valid range is 96 to 99, and 106 to 127. The 
default is 96. The 100, 102 to 105 range is allocated for proprietary 
usage." But FS uses 101 by default. Is this ok or should I change it to 
96 for example?

-- 
Regards,
Boris




From kris at kriskinc.com  Fri Apr 29 18:45:31 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 29 Apr 2011 10:45:31 -0400
Subject: [Freeswitch-users] rfc2833-pt
In-Reply-To: <4DBACE54.30809@tagnet.ru>
References: <4DBACE54.30809@tagnet.ru>
Message-ID: <BANLkTikAzag1bwhF1OMq3Rv6pnedZ7pJow@mail.gmail.com>

You don't have to change it.  SDP will handle it for you.

I've worked with many AudioCodes gateways without every change the
RFC2833 payload type.

On Fri, Apr 29, 2011 at 10:42 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> ? ? In the Audiocodecs MP-118 User's Manual "The RFC 2833 DTMF relay
> dynamic payload type. The valid range is 96 to 99, and 106 to 127. The
> default is 96. The 100, 102 to 105 range is allocated for proprietary
> usage." But FS uses 101 by default. Is this ok or should I change it to
> 96 for example?
>
> --
> 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
>



-- 
Kristian Kielhofner


From kris at kriskinc.com  Fri Apr 29 18:46:40 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 29 Apr 2011 10:46:40 -0400
Subject: [Freeswitch-users] Trunking between Lync and FreeSWITCH
In-Reply-To: <3120C73F-FFD6-4979-87FD-A61AA1A2E14C@ipeva.fr>
References: <3120C73F-FFD6-4979-87FD-A61AA1A2E14C@ipeva.fr>
Message-ID: <BANLkTi=s_FrshgnBPoxcHzRwZYRk2w7dCw@mail.gmail.com>

Interesting.  Can you put a sip trace up between Lync and FreeSWITCH?

On Thu, Apr 28, 2011 at 5:24 PM, David Ponzone <david.ponzone at ipeva.fr> wrote:
> Well, I thought I could start a thread about that.
> I am trying to accomplish this, and I guess I am not the only one.
> Lync 2010 seems to have quite some success (compared to OCS), so there are
> probably business opportunities around this.
> My first attempt is a half success, as I got calls from Lync to FreeSWITCH
> working (need some extensive testing, though), and calls from FreeSWITCH to
> Lync are connecting ok, but they get disconnected after circa 30 seconds.
> I am trying to pinpoint the reason for this disconnection, but untll now, I
> had no luck.
> RTCP requirement was disabled on the Lync side (RTCPActiveCalls), but anyway
> I have enabled RTCP on the FreeSWITCH, so it should not be that.
> RTP is flowing normally, it's not a call on hold or muted.
> Refer was disabled on the Lync side.
> SessionTimer is enabled on the Lync side (but I dont think I ever saw a
> REINVITE caused by the session timer).
> I just suddenly receive a BYE from Lync, after 30-35 seconds.
> On the Lync side, as you can guess, it's not easy to access any meaningful
> logs. I think our partner managing the Lync will have to escalate that to
> the SVP Product Engineering to get the right command to enable the
> interesting debug mode :)
> I've been told that some folks at MS were claiming a such trunking was not
> possible.
> Well, I would tend to say otherwise, as I am not far to get this working,
> except of course if they just hardcoded some forbidden Owner Usernames in
> the SDP, like FreeSWITCH, in order to save this business from the mean
> OpenSource world and leave it to commercial SBC vendors.
> I can hardly imagine they would dare to do that in 2011, so the question
> remains opened: what may be closing that call after 30 seconds ?
> I would be glad to discuss the subject, here or privately, with anyone who
> is involved on a such project, or planning to be soon.
> David Ponzone ?Direction Technique
> email: david.ponzone at ipeva.fr
> tel: ? ? ?01 74 03 18 97
> gsm: ? 06 66 98 76 34
> Service Client?IPeva
> tel: ? ? ?0811 46 26 26
> www.ipeva.fr? -? ?www.ipeva-studio.com
> Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ?
> l'intention exclusive de ses destinataires. Toute utilisation ou diffusion
> non autoris?e est interdite. Tout message ?lectronique est susceptible
> d'alt?ration.?IPeva?d?cline toute responsabilit? au titre de ce message s'il
> a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce
> message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 ctroncoso at redvoiss.net  Fri Apr 29 17:21:01 2011
From: ctroncoso at redvoiss.net (Camila Troncoso)
Date: Fri, 29 Apr 2011 10:21:01 -0300
Subject: [Freeswitch-users] mod_radius_cdr and Acct-Delay-Time
In-Reply-To: <BANLkTi=Dxk4pdG7nMDHugEjoOhpDsEqVYQ@mail.gmail.com>
References: <9b86b6d280317812f3b7de59136aa457@mail.gmail.com>
	<BANLkTi=Dxk4pdG7nMDHugEjoOhpDsEqVYQ@mail.gmail.com>
Message-ID: <b3b623bc7e6d035914ce1a15716da299@mail.gmail.com>

Thanks,

But where do I find these changes, are they in the git source?

Regardas,

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of ????
????????
Sent: viernes, 29 de abril de 2011 8:28
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_radius_cdr and Acct-Delay-Time

Just fix mod_radius_cdr )) change format of Freeswitch-Callstartdat.
Use PW_ACCT_DELAY_TIME for adding Acct-Delay-Time field.

2011/4/27 Camila Troncoso <ctroncoso at redvoiss.net>:
>
>
> Hi,
>
>
>
> I?m working with radius accounting using the mod_radius_cdr module. I?m
> having trouble with the date format that Radius server receive. ?An
example
> of what the radius server receive is:
>
>
>
> ?Freeswitch-Callstartdate = "2011-04-21T18:23:59.780945-0400"?
>
>
>
> This date format is very difficult to read and I want to change it? to
make
> the accounting easier. I search all around for some param or
configuration
> that allows me to do so, but I only find that this date format is define
in
> mod_radius_cdr.c module.
>
>
>
> I?m also having problems with the parameter Acct-Delay-Time, it is not
> increasing when the client resend the radius packet. I read the
buildreq.c
> code but didn?t find the problem.
>
>
>
> Please some help with this matter.
>
>
>
> Regards,
>
>
>
> Camila Troncoso |?Ingeniero de Desarrollo
>
> RedVoiss |ctroncoso at redvoiss.net
>
> Santiago - Chile | +56 2 2408535
>
> www.redvoiss.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
>
>

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


From boris at tagnet.ru  Fri Apr 29 19:00:17 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Fri, 29 Apr 2011 21:00:17 +0600
Subject: [Freeswitch-users] rfc2833-pt
In-Reply-To: <BANLkTikAzag1bwhF1OMq3Rv6pnedZ7pJow@mail.gmail.com>
References: <4DBACE54.30809@tagnet.ru>
	<BANLkTikAzag1bwhF1OMq3Rv6pnedZ7pJow@mail.gmail.com>
Message-ID: <4DBAD281.8070505@tagnet.ru>

Hello!

Ok. And why FS uses 101 while valid are 96 to 99? Or this means valid 
only for MP-118?

> You don't have to change it.  SDP will handle it for you.
>
> I've worked with many AudioCodes gateways without every change the
> RFC2833 payload type.
>
> On Fri, Apr 29, 2011 at 10:42 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>>      In the Audiocodecs MP-118 User's Manual "The RFC 2833 DTMF relay
>> dynamic payload type. The valid range is 96 to 99, and 106 to 127. The
>> default is 96. The 100, 102 to 105 range is allocated for proprietary
>> usage." But FS uses 101 by default. Is this ok or should I change it to
>> 96 for example?
>>
>> --
>> 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
>>
>
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



From jeff at jefflenk.com  Fri Apr 29 19:15:21 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 29 Apr 2011 08:15:21 -0700 (PDT)
Subject: [Freeswitch-users] error LNK2019: unresolved external symbol
 when building on x64 machine but not in Win32 machine
In-Reply-To: <BANLkTinO3zneiUSTB-rOg+tD+cVKiZvS=w@mail.gmail.com>
References: <BANLkTinO3zneiUSTB-rOg+tD+cVKiZvS=w@mail.gmail.com>
Message-ID: <1304090121179-6317268.post@n2.nabble.com>

You're not linking against FreeSWITCHCore.lib--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/error-LNK2019-unresolved-external-symbol-when-building-on-x64-machine-but-not-in-Win32-machine-tp6316391p6317268.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jeff at jefflenk.com  Fri Apr 29 19:22:07 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 29 Apr 2011 08:22:07 -0700 (PDT)
Subject: [Freeswitch-users] FS - SIP profiles crashed
In-Reply-To: <BANLkTi=402RnR54vQmVx7_fh8z1iaX3aFw@mail.gmail.com>
References: <BANLkTi=402RnR54vQmVx7_fh8z1iaX3aFw@mail.gmail.com>
Message-ID: <1304090527690-6317287.post@n2.nabble.com>

If your box is configured with a static IP do this.

conf/autoload_configs/sofia.conf.xml --
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-SIP-profiles-crashed-tp6315699p6317287.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From kris at kriskinc.com  Fri Apr 29 20:03:30 2011
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 29 Apr 2011 12:03:30 -0400
Subject: [Freeswitch-users] rfc2833-pt
In-Reply-To: <4DBAD281.8070505@tagnet.ru>
References: <4DBACE54.30809@tagnet.ru>
	<BANLkTikAzag1bwhF1OMq3Rv6pnedZ7pJow@mail.gmail.com>
	<4DBAD281.8070505@tagnet.ru>
Message-ID: <BANLkTi=J2FwbXWV_BKy634JpRviCp6JKFA@mail.gmail.com>

According to the IANA payload types:

http://www.iana.org/assignments/rtp-parameters

96-127 are dynamic.  Anything is available to use them.  The
"limitations" you are describing come from Audiocodes.  In my
experience with a wide variety of endpoints 101 is fairly
typical/standard for RFC 2833.


On Fri, Apr 29, 2011 at 11:00 AM, Boris Kovalenko <boris at tagnet.ru> wrote:
> Hello!
>
> Ok. And why FS uses 101 while valid are 96 to 99? Or this means valid
> only for MP-118?
>
>> You don't have to change it. ?SDP will handle it for you.
>>
>> I've worked with many AudioCodes gateways without every change the
>> RFC2833 payload type.
>>
>> On Fri, Apr 29, 2011 at 10:42 AM, Boris Kovalenko<boris at tagnet.ru> ?wrote:
>>> Hello!
>>>
>>> ? ? ?In the Audiocodecs MP-118 User's Manual "The RFC 2833 DTMF relay
>>> dynamic payload type. The valid range is 96 to 99, and 106 to 127. The
>>> default is 96. The 100, 102 to 105 range is allocated for proprietary
>>> usage." But FS uses 101 by default. Is this ok or should I change it to
>>> 96 for example?
>>>
>>> --
>>> 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
>>>
>>
>>
>
>
> --
> ? ?????????,
> ? ????? ?????????
> ? ??? "??????"
> ? ???. +7 (3435) 230001
> ? ???? +7 (3435) 230005
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 msc at freeswitch.org  Fri Apr 29 20:49:40 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Apr 2011 09:49:40 -0700
Subject: [Freeswitch-users] Fwd: problem with transferring calls when
 using ivr
In-Reply-To: <op.vunln9d3agcvls@master>
References: <op.vunkngteagcvls@master>
	<op.vunln9d3agcvls@master>
Message-ID: <BANLkTi=rmiGVZi8M=Ymt1PcDoK9=ZWofuA@mail.gmail.com>

Bart,

Welcome to FreeSWITCH, the future of telephony! ;)

Okay, I want you to try something in your js. Everywhere that you see
"execute_extension" I want you to replace those with "transfer". Without
seeing call traces I can't know for sure, but I'm guessing that you are
"stuck in js" because of the execute_extension. Check the wiki for an
explanation of the difference between execute_extension and transfer. I'm
99% sure that your IVR actions need to be transfers.

NOTE: a "transfer" in the dialplan is different than someone pressing the
transfer key on their phone. I know it can be confusing, but you'll see the
nuances if you stick with it long enough. Welcome to the wonderful world of
telephony!

-MC

On Thu, Apr 28, 2011 at 8:08 AM, <bogusmaster at o2.pl> wrote:

> Hi all !
>
> I'm a freeswitch newbie taking care of installation that I've recently
> taken over from a colleague who left our company.
> Currently I'm having a problem with transferring calls that went thru ivr
> menu.
>
> The call scenario looks like this: external number A calls our external
> number E. Then, in ivr menu it selects some option. After that A is
> connected with internal number I.
> When internal number I tries to transfer call from A to another internal
> number I2, call hangs up.
>
> If I turn the ivr off, as it is commented in the excerpt from
> /conf/dialplan/public/01_freephone_inbound.xml below, transferring calls
> works fine.
>
>    <extension name="general_inbound">
>      <condition field="destination_number"
> expression="^(freephone_XXXX_in)$">
>        <action application="info"/>
>        <action application="transfer" data="100 XML default"/>
>        <!--action application="javascript" data="simpleivr.js"/-->
>      </condition>
>    </extension>
>
> I attach my ivr script.
>
> Any hints would be very appreciated.
>
> Regards,
> Bart
>
>
> --
> Using Opera's revolutionary email client: http://www.opera.com/mail/
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 29 20:56:39 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Apr 2011 09:56:39 -0700
Subject: [Freeswitch-users] Call quality issues
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DF6A0EDB@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DF6A0EDB@doctorbob.loop.local>
Message-ID: <BANLkTikZtoeTydHn+UJLfzKYH7bwwRLAVA@mail.gmail.com>

On Thu, Apr 28, 2011 at 2:52 PM, Eric Beard <eric at loopfx.com> wrote:

> Hello,
>
>
>
> I ran into some major call quality issues this week, and I?m trying to
> figure out how to troubleshoot things.  I?ve been running FreeSwitch for a
> few weeks, and suddenly a few days ago my call quality dropped drastically.
> I had been running more than 100 concurrent calls, with the CPU at less than
> 20%, but now at 20 concurrent calls, the CPU is still at a little less than
> 20%, and the call quality is bad ? any higher and calls go almost completely
> silent.  There is a direct correlation between the number of simultaneous
> calls and call quality.
>
>
>
> I have tested against multiple gateways, same results against each, so it?s
> not an issue with the gateway.
>
>
>
> I have captured packets on the machine and analyzed them with Wireshark.
> It seems like the inbound packets are all fine, no jitter or loss.  But the
> packets being sent by FreeSwitch are degraded.
>
>
>
> One sample call showed:
>
>
>
> Drop by Jitter Buff:158(14.1%) Out of Seq 0 (0.0%) Wrong Timestamp 96(8.6%)
>

What is the topology of the network path for the above call? Also, where on
the LAN/WAN did you capture these packets? Wrong timestamps and 14% dropped
packets suggests that something on the network is interfering with the
delivery of these packets in a timely manner.
-MC
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From eric at loopfx.com  Fri Apr 29 21:10:19 2011
From: eric at loopfx.com (Eric Beard)
Date: Fri, 29 Apr 2011 13:10:19 -0400
Subject: [Freeswitch-users] Call quality issues
In-Reply-To: <BANLkTikZtoeTydHn+UJLfzKYH7bwwRLAVA@mail.gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DF6A0EDB@doctorbob.loop.local>
	<BANLkTikZtoeTydHn+UJLfzKYH7bwwRLAVA@mail.gmail.com>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DF6A11A8@doctorbob.loop.local>

It seems to be a CPU issue.

I started from scratch today with a new machine, same exact hardware, and I am running 20 calls at less than 1% CPU, with perfect audio quality (same as the bad machine before it went bad).

On the bad machine, 20 calls causes the CPU to be at 20%, and quality is shaky.  30 calls and it's terrible.  Only the freeswitch process is eating CPU.

I have diffed the conf folders and everything is the same.  This is driving me nuts.  I have no idea  what I did to that machine to ruin the performance.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, April 29, 2011 12:57 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call quality issues


On Thu, Apr 28, 2011 at 2:52 PM, Eric Beard <eric at loopfx.com<mailto:eric at loopfx.com>> wrote:
Hello,

I ran into some major call quality issues this week, and I'm trying to figure out how to troubleshoot things.  I've been running FreeSwitch for a few weeks, and suddenly a few days ago my call quality dropped drastically.  I had been running more than 100 concurrent calls, with the CPU at less than 20%, but now at 20 concurrent calls, the CPU is still at a little less than 20%, and the call quality is bad - any higher and calls go almost completely silent.  There is a direct correlation between the number of simultaneous calls and call quality.

I have tested against multiple gateways, same results against each, so it's not an issue with the gateway.

I have captured packets on the machine and analyzed them with Wireshark.  It seems like the inbound packets are all fine, no jitter or loss.  But the packets being sent by FreeSwitch are degraded.

One sample call showed:

Drop by Jitter Buff:158(14.1%) Out of Seq 0 (0.0%) Wrong Timestamp 96(8.6%)

What is the topology of the network path for the above call? Also, where on the LAN/WAN did you capture these packets? Wrong timestamps and 14% dropped packets suggests that something on the network is interfering with the delivery of these packets in a timely manner.
-MC

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From infos at madovsky.org  Fri Apr 29 21:13:36 2011
From: infos at madovsky.org (Madovsky)
Date: Fri, 29 Apr 2011 13:13:36 -0400
Subject: [Freeswitch-users] SIP external profile - Error looking up
	hostname
References: <BANLkTinaJ66c3VS-rM7GjUZvhSh_wQ=aFg@mail.gmail.com>
	<BANLkTimZSPiHS0Lm5P1Jv=obbSJpqK1Opg@mail.gmail.com>
Message-ID: <E0BB4CD72489499F9696819DD437427E@e1705>

yeah I'm here !

----- Original Message ----- 
From: "Anton VG" <anton.vazir at gmail.com>
To: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org>
Sent: Friday, April 29, 2011 6:24 AM
Subject: Re: [Freeswitch-users] SIP external profile - Error looking up 
hostname


> anyone?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.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 29 21:32:32 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Apr 2011 10:32:32 -0700
Subject: [Freeswitch-users] Call quality issues
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DF6A11A8@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DF6A0EDB@doctorbob.loop.local>
	<BANLkTikZtoeTydHn+UJLfzKYH7bwwRLAVA@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DF6A11A8@doctorbob.loop.local>
Message-ID: <BANLkTind4gbC5p1RM1tk_Te=E-HngaMLJA@mail.gmail.com>

What about the operating system? Same rev, same kernel, same timing?
-MC

On Fri, Apr 29, 2011 at 10:10 AM, Eric Beard <eric at loopfx.com> wrote:

> It seems to be a CPU issue.
>
>
>
> I started from scratch today with a new machine, same exact hardware, and I
> am running 20 calls at less than 1% CPU, with perfect audio quality (same as
> the bad machine before it went bad).
>
>
>
> On the bad machine, 20 calls causes the CPU to be at 20%, and quality is
> shaky.  30 calls and it?s terrible.  Only the freeswitch process is eating
> CPU.
>
>
>
> I have diffed the conf folders and everything is the same.  This is driving
> me nuts.  I have no idea  what I did to that machine to ruin the
> performance.
>
>
>
> -----------------------
>
> *Eric Z. Beard, CTO*
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> eric at loopfx.com
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Friday, April 29, 2011 12:57 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Call quality issues
>
>
>
>
>
> On Thu, Apr 28, 2011 at 2:52 PM, Eric Beard <eric at loopfx.com> wrote:
>
> Hello,
>
>
>
> I ran into some major call quality issues this week, and I?m trying to
> figure out how to troubleshoot things.  I?ve been running FreeSwitch for a
> few weeks, and suddenly a few days ago my call quality dropped drastically.
> I had been running more than 100 concurrent calls, with the CPU at less than
> 20%, but now at 20 concurrent calls, the CPU is still at a little less than
> 20%, and the call quality is bad ? any higher and calls go almost completely
> silent.  There is a direct correlation between the number of simultaneous
> calls and call quality.
>
>
>
> I have tested against multiple gateways, same results against each, so it?s
> not an issue with the gateway.
>
>
>
> I have captured packets on the machine and analyzed them with Wireshark.
> It seems like the inbound packets are all fine, no jitter or loss.  But the
> packets being sent by FreeSwitch are degraded.
>
>
>
> One sample call showed:
>
>
>
> Drop by Jitter Buff:158(14.1%) Out of Seq 0 (0.0%) Wrong Timestamp 96(8.6%)
>
>
>
> What is the topology of the network path for the above call? Also, where on
> the LAN/WAN did you capture these packets? Wrong timestamps and 14% dropped
> packets suggests that something on the network is interfering with the
> delivery of these packets in a timely manner.
>
> -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  Fri Apr 29 21:34:41 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Apr 2011 10:34:41 -0700
Subject: [Freeswitch-users] FS - SIP profiles crashed
In-Reply-To: <1304090527690-6317287.post@n2.nabble.com>
References: <BANLkTi=402RnR54vQmVx7_fh8z1iaX3aFw@mail.gmail.com>
	<1304090527690-6317287.post@n2.nabble.com>
Message-ID: <BANLkTinqHZb1A0Fqz1z+BK+UeM2vUfWG9A@mail.gmail.com>

On Fri, Apr 29, 2011 at 8:22 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

> If your box is configured with a static IP do this.
>
> conf/autoload_configs/sofia.conf.xml --
>

Jeff, was there more to this message?
-MC
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From msc at freeswitch.org  Fri Apr 29 21:42:01 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Apr 2011 10:42:01 -0700
Subject: [Freeswitch-users] Reviewers needed: FreeSWITCH Cookbook
Message-ID: <BANLkTikL+6i6ovRX1EsfzMoWzDiw3TvbcQ@mail.gmail.com>

Hello,

As you know we are working on a FreeSWITCH Cookbook for Packt Publishing. We
are in need of 3 or 4 capable reviewers. Please contact me off list if you
want to help and meet these qualifications:

* Read and write English fluently
* Have at least one instance of FreeSWITCH (or a sandbox) where you can test
each recipe
* Have time to read and try all the recipes in the book
* Can offer useful feedback and proofreading
* Preferably have a later version of MS Word for editing

If you meet the above qualifications and would like to help out then please
contact me off list for more information.

Thanks,
MC
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From eric at loopfx.com  Fri Apr 29 22:03:52 2011
From: eric at loopfx.com (Eric Beard)
Date: Fri, 29 Apr 2011 14:03:52 -0400
Subject: [Freeswitch-users] Call quality issues
In-Reply-To: <BANLkTind4gbC5p1RM1tk_Te=E-HngaMLJA@mail.gmail.com>
References: <D4A666272A214C4DBBABFE569CA34CA401DF6A0EDB@doctorbob.loop.local>
	<BANLkTikZtoeTydHn+UJLfzKYH7bwwRLAVA@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DF6A11A8@doctorbob.loop.local>
	<BANLkTind4gbC5p1RM1tk_Te=E-HngaMLJA@mail.gmail.com>
Message-ID: <D4A666272A214C4DBBABFE569CA34CA401DF6A123A@doctorbob.loop.local>

On the bad machine, I just did a make clean, re-ran bootstrap.sh, configure, make, then make install, and now it works fine.  I'm mystified.  I really wish I knew what the problem was so I can avoid it in the future.

The only possible cause I can think of is that at some point I needed to rebuild to use an extra module (like curl), and I did a make, make install, without doing a make clean, and somehow that was related?  Sounds iffy.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Friday, April 29, 2011 1:33 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call quality issues

What about the operating system? Same rev, same kernel, same timing?
-MC
On Fri, Apr 29, 2011 at 10:10 AM, Eric Beard <eric at loopfx.com<mailto:eric at loopfx.com>> wrote:
It seems to be a CPU issue.

I started from scratch today with a new machine, same exact hardware, and I am running 20 calls at less than 1% CPU, with perfect audio quality (same as the bad machine before it went bad).

On the bad machine, 20 calls causes the CPU to be at 20%, and quality is shaky.  30 calls and it's terrible.  Only the freeswitch process is eating CPU.

I have diffed the conf folders and everything is the same.  This is driving me nuts.  I have no idea  what I did to that machine to ruin the performance.

-----------------------
Eric Z. Beard, CTO
Loop LLC
w (877) 850-2010 x9249
m (727) 776-2768
eric at loopfx.com<mailto:eric at loopfx.com>

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Friday, April 29, 2011 12:57 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call quality issues


On Thu, Apr 28, 2011 at 2:52 PM, Eric Beard <eric at loopfx.com<mailto:eric at loopfx.com>> wrote:
Hello,

I ran into some major call quality issues this week, and I'm trying to figure out how to troubleshoot things.  I've been running FreeSwitch for a few weeks, and suddenly a few days ago my call quality dropped drastically.  I had been running more than 100 concurrent calls, with the CPU at less than 20%, but now at 20 concurrent calls, the CPU is still at a little less than 20%, and the call quality is bad - any higher and calls go almost completely silent.  There is a direct correlation between the number of simultaneous calls and call quality.

I have tested against multiple gateways, same results against each, so it's not an issue with the gateway.

I have captured packets on the machine and analyzed them with Wireshark.  It seems like the inbound packets are all fine, no jitter or loss.  But the packets being sent by FreeSwitch are degraded.

One sample call showed:

Drop by Jitter Buff:158(14.1%) Out of Seq 0 (0.0%) Wrong Timestamp 96(8.6%)

What is the topology of the network path for the above call? Also, where on the LAN/WAN did you capture these packets? Wrong timestamps and 14% dropped packets suggests that something on the network is interfering with the delivery of these packets in a timely manner.
-MC


_______________________________________________
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 msc at freeswitch.org  Fri Apr 29 22:12:55 2011
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 29 Apr 2011 11:12:55 -0700
Subject: [Freeswitch-users] Call quality issues
In-Reply-To: <D4A666272A214C4DBBABFE569CA34CA401DF6A123A@doctorbob.loop.local>
References: <D4A666272A214C4DBBABFE569CA34CA401DF6A0EDB@doctorbob.loop.local>
	<BANLkTikZtoeTydHn+UJLfzKYH7bwwRLAVA@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DF6A11A8@doctorbob.loop.local>
	<BANLkTind4gbC5p1RM1tk_Te=E-HngaMLJA@mail.gmail.com>
	<D4A666272A214C4DBBABFE569CA34CA401DF6A123A@doctorbob.loop.local>
Message-ID: <BANLkTi=sGLYWpnV-rmn78MngdsKq0x6hQg@mail.gmail.com>

Build skew is a possibility. We always recommend a make current for
updating, just in case.
-MC

On Fri, Apr 29, 2011 at 11:03 AM, Eric Beard <eric at loopfx.com> wrote:

> On the bad machine, I just did a make clean, re-ran bootstrap.sh,
> configure, make, then make install, and now it works fine.  I?m mystified.
> I really wish I knew what the problem was so I can avoid it in the future.
>
>
>
> The only possible cause I can think of is that at some point I needed to
> rebuild to use an extra module (like curl), and I did a make, make install,
> without doing a make clean, and somehow that was related?  Sounds iffy.
>
>
>
> -----------------------
>
> *Eric Z. Beard, CTO*
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> eric at loopfx.com
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Friday, April 29, 2011 1:33 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Call quality issues
>
>
>
> What about the operating system? Same rev, same kernel, same timing?
>
> -MC
>
> On Fri, Apr 29, 2011 at 10:10 AM, Eric Beard <eric at loopfx.com> wrote:
>
> It seems to be a CPU issue.
>
>
>
> I started from scratch today with a new machine, same exact hardware, and I
> am running 20 calls at less than 1% CPU, with perfect audio quality (same as
> the bad machine before it went bad).
>
>
>
> On the bad machine, 20 calls causes the CPU to be at 20%, and quality is
> shaky.  30 calls and it?s terrible.  Only the freeswitch process is eating
> CPU.
>
>
>
> I have diffed the conf folders and everything is the same.  This is driving
> me nuts.  I have no idea  what I did to that machine to ruin the
> performance.
>
>
>
> -----------------------
>
> *Eric Z. Beard, CTO*
>
> Loop LLC
>
> w (877) 850-2010 x9249
>
> m (727) 776-2768
>
> eric at loopfx.com
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Friday, April 29, 2011 12:57 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Call quality issues
>
>
>
>
>
> On Thu, Apr 28, 2011 at 2:52 PM, Eric Beard <eric at loopfx.com> wrote:
>
> Hello,
>
>
>
> I ran into some major call quality issues this week, and I?m trying to
> figure out how to troubleshoot things.  I?ve been running FreeSwitch for a
> few weeks, and suddenly a few days ago my call quality dropped drastically.
> I had been running more than 100 concurrent calls, with the CPU at less than
> 20%, but now at 20 concurrent calls, the CPU is still at a little less than
> 20%, and the call quality is bad ? any higher and calls go almost completely
> silent.  There is a direct correlation between the number of simultaneous
> calls and call quality.
>
>
>
> I have tested against multiple gateways, same results against each, so it?s
> not an issue with the gateway.
>
>
>
> I have captured packets on the machine and analyzed them with Wireshark.
> It seems like the inbound packets are all fine, no jitter or loss.  But the
> packets being sent by FreeSwitch are degraded.
>
>
>
> One sample call showed:
>
>
>
> Drop by Jitter Buff:158(14.1%) Out of Seq 0 (0.0%) Wrong Timestamp 96(8.6%)
>
>
>
> What is the topology of the network path for the above call? Also, where on
> the LAN/WAN did you capture these packets? Wrong timestamps and 14% dropped
> packets suggests that something on the network is interfering with the
> delivery of these packets in a timely manner.
>
> -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 garmt.noname at gmail.com  Sat Apr 30 01:36:34 2011
From: garmt.noname at gmail.com (Grmt)
Date: Fri, 29 Apr 2011 23:36:34 +0200
Subject: [Freeswitch-users] FS - SIP profiles crashed
In-Reply-To: <BANLkTinqHZb1A0Fqz1z+BK+UeM2vUfWG9A@mail.gmail.com>
References: <BANLkTi=402RnR54vQmVx7_fh8z1iaX3aFw@mail.gmail.com>
	<1304090527690-6317287.post@n2.nabble.com>
	<BANLkTinqHZb1A0Fqz1z+BK+UeM2vUfWG9A@mail.gmail.com>
Message-ID: <4dbb2f69.81bf0e0a.4bad.2812@mx.google.com>

http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP
_address

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Friday, 29 April, 2011 19:35
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FS - SIP profiles crashed

 

 

On Fri, Apr 29, 2011 at 8:22 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

If your box is configured with a static IP do this.

conf/autoload_configs/sofia.conf.xml --

 

Jeff, was there more to this message?

-MC

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From michel at arneill-py.sacramento.ca.us  Sat Apr 30 03:41:24 2011
From: michel at arneill-py.sacramento.ca.us (Michel Py)
Date: Fri, 29 Apr 2011 16:41:24 -0700
Subject: [Freeswitch-users] Newbie question about Polycom presence / BLF
	with productivity license.
In-Reply-To: <BANLkTik=Ap6Yevy0OBc+Dto2-HdzLYjoBA@mail.gmail.com>
References: <471D76419F9EF642962323D13DF1DF69011E50@newserver.arneill-py.local><BANLkTi=XmNDV51=xoNsViHUYaUZeyMpz_A@mail.gmail.com><471D76419F9EF642962323D13DF1DF69011E58@newserver.arneill-py.local><BANLkTi=o1nUa70X_PHU51M_TTGTfnSSdKg@mail.gmail.com><471D76419F9EF642962323D13DF1DF69011E59@newserver.arneill-py.local><BANLkTi=9Gu1vOD-KVdCSs3MbNxng+Tqv2w@mail.gmail.com><471D76419F9EF642962323D13DF1DF69011E5A@newserver.arneill-py.local><BANLkTimzyrNQOLKs_CRRtFKRpw6isCg+AQ@mail.gmail.com><471D76419F9EF642962323D13DF1DF69011E5B@newserver.arneill-py.local>
	<BANLkTik=Ap6Yevy0OBc+Dto2-HdzLYjoBA@mail.gmail.com>
Message-ID: <471D76419F9EF642962323D13DF1DF6901B8CB@newserver.arneill-py.local>

Christian,

> Polycom are phantastic phones(imho the best on the market)

FWIW, I've been happy with Aastra so far.

Thanks for the help
Michel.



From jeff at jefflenk.com  Sat Apr 30 06:22:38 2011
From: jeff at jefflenk.com (Jeff Lenk)
Date: Fri, 29 Apr 2011 19:22:38 -0700 (PDT)
Subject: [Freeswitch-users] FS - SIP profiles crashed
In-Reply-To: <BANLkTinqHZb1A0Fqz1z+BK+UeM2vUfWG9A@mail.gmail.com>
References: <BANLkTi=402RnR54vQmVx7_fh8z1iaX3aFw@mail.gmail.com>
	<1304090527690-6317287.post@n2.nabble.com>
	<BANLkTinqHZb1A0Fqz1z+BK+UeM2vUfWG9A@mail.gmail.com>
Message-ID: <1304130158269-6318796.post@n2.nabble.com>

Hi Michael,

No idea what happened there but yes there was. But garmts response has the
same content.

Jeff--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-SIP-profiles-crashed-tp6315699p6318796.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From fieldpeak at gmail.com  Sat Apr 30 06:39:00 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sat, 30 Apr 2011 10:39:00 +0800
Subject: [Freeswitch-users] FS - SIP profiles crashed
In-Reply-To: <4dbb2f69.81bf0e0a.4bad.2812@mx.google.com>
References: <BANLkTi=402RnR54vQmVx7_fh8z1iaX3aFw@mail.gmail.com>
	<1304090527690-6317287.post@n2.nabble.com>
	<BANLkTinqHZb1A0Fqz1z+BK+UeM2vUfWG9A@mail.gmail.com>
	<4dbb2f69.81bf0e0a.4bad.2812@mx.google.com>
Message-ID: <BANLkTimYon6k0qvZtUnQWY0MVMS6AYEhmw@mail.gmail.com>

Thanks all for information, it definitely help a lot. i will try...

meanwhile, i found this link
http://lists.freeswitch.org/pipermail/freeswitch-users/2008-May/003086.html

it looks not so special resolve this issue...

i would like to know what is use for "bind_server_ip=auto" and where and
when FS get the value of $${local_ip_v4}? and if it will changed when
physical IP changed?

Thanks!

Regards,
Charles

2011/4/30 Grmt <garmt.noname at gmail.com>

>
> http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Friday, 29 April, 2011 19:35
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] FS - SIP profiles crashed
>
>
>
>
>
> On Fri, Apr 29, 2011 at 8:22 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>
> If your box is configured with a static IP do this.
>
> conf/autoload_configs/sofia.conf.xml --
>
>
>
> Jeff, was there more to this 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
>
>
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From fieldpeak at gmail.com  Sat Apr 30 06:46:58 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sat, 30 Apr 2011 10:46:58 +0800
Subject: [Freeswitch-users] FS - SIP profiles crashed
In-Reply-To: <BANLkTimYon6k0qvZtUnQWY0MVMS6AYEhmw@mail.gmail.com>
References: <BANLkTi=402RnR54vQmVx7_fh8z1iaX3aFw@mail.gmail.com>
	<1304090527690-6317287.post@n2.nabble.com>
	<BANLkTinqHZb1A0Fqz1z+BK+UeM2vUfWG9A@mail.gmail.com>
	<4dbb2f69.81bf0e0a.4bad.2812@mx.google.com>
	<BANLkTimYon6k0qvZtUnQWY0MVMS6AYEhmw@mail.gmail.com>
Message-ID: <BANLkTikXtcGECS_tZPMcS+4Kq_mvASgnXA@mail.gmail.com>

really want to know what caused that the IP address changed to '0.0.0.0',
very odd...
i tried even unplug and plugin the network cable, it will not change the
IP...let alone 0.0.0.0...

2011/4/30 fieldpeak <fieldpeak at gmail.com>

> Thanks all for information, it definitely help a lot. i will try...
>
> meanwhile, i found this link
> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-May/003086.html
>
> it looks not so special resolve this issue...
>
> i would like to know what is use for "bind_server_ip=auto" and where and
> when FS get the value of $${local_ip_v4}? and if it will changed when
> physical IP changed?
>
> Thanks!
>
> Regards,
> Charles
>
> 2011/4/30 Grmt <garmt.noname at gmail.com>
>
>>
>> http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address
>>
>>
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
>> Collins
>> *Sent:* Friday, 29 April, 2011 19:35
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] FS - SIP profiles crashed
>>
>>
>>
>>
>>
>> On Fri, Apr 29, 2011 at 8:22 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>>
>> If your box is configured with a static IP do this.
>>
>> conf/autoload_configs/sofia.conf.xml --
>>
>>
>>
>> Jeff, was there more to this 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
>>
>>
>
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From fieldpeak at gmail.com  Sat Apr 30 07:34:34 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sat, 30 Apr 2011 11:34:34 +0800
Subject: [Freeswitch-users] FS - SIP profiles crashed
In-Reply-To: <BANLkTikXtcGECS_tZPMcS+4Kq_mvASgnXA@mail.gmail.com>
References: <BANLkTi=402RnR54vQmVx7_fh8z1iaX3aFw@mail.gmail.com>
	<1304090527690-6317287.post@n2.nabble.com>
	<BANLkTinqHZb1A0Fqz1z+BK+UeM2vUfWG9A@mail.gmail.com>
	<4dbb2f69.81bf0e0a.4bad.2812@mx.google.com>
	<BANLkTimYon6k0qvZtUnQWY0MVMS6AYEhmw@mail.gmail.com>
	<BANLkTikXtcGECS_tZPMcS+4Kq_mvASgnXA@mail.gmail.com>
Message-ID: <BANLkTinhWi3Q7x4B3hM60w+1F96htTVpvA@mail.gmail.com>

oh, looks i found the root cause,

after i unplug the network cable and wait around 20 seconds, FS show ' IP
address changed to '0.0.0.0', then try to load profiles, however, due to
wrong IP address, load failure.
and then i re-plugin the cable, FS detected the IP change back, and
reloadxml, however, did not restart the profiles, so caused profiles
crashed... attached is the log for throughout the procedure...

the resolution is set

in the sofia.conf.xml file:

<param name="auto-restart" value="false"/>

welcome to any comment,

thanks.

Regards,
Charles



2011/4/30 fieldpeak <fieldpeak at gmail.com>

> really want to know what caused that the IP address changed to '0.0.0.0',
> very odd...
> i tried even unplug and plugin the network cable, it will not change the
> IP...let alone 0.0.0.0...
>
>
> 2011/4/30 fieldpeak <fieldpeak at gmail.com>
>
>> Thanks all for information, it definitely help a lot. i will try...
>>
>> meanwhile, i found this link
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-May/003086.html
>>
>> it looks not so special resolve this issue...
>>
>> i would like to know what is use for "bind_server_ip=auto" and where and
>> when FS get the value of $${local_ip_v4}? and if it will changed when
>> physical IP changed?
>>
>> Thanks!
>>
>> Regards,
>> Charles
>>
>> 2011/4/30 Grmt <garmt.noname at gmail.com>
>>
>>>
>>> http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address
>>>
>>>
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
>>> Collins
>>> *Sent:* Friday, 29 April, 2011 19:35
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] FS - SIP profiles crashed
>>>
>>>
>>>
>>>
>>>
>>> On Fri, Apr 29, 2011 at 8:22 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>>>
>>> If your box is configured with a static IP do this.
>>>
>>> conf/autoload_configs/sofia.conf.xml --
>>>
>>>
>>>
>>> Jeff, was there more to this 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
>>>
>>>
>>
>
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2011-04-30 11:07:27.937500 [NOTICE] switch_loadable_module.c:185 Adding Codec PC
MA 8 G.711 alaw 8000hz 30ms 64000bps
2011-04-30 11:07:27.937500 [NOTICE] switch_loadable_module.c:185 Adding Codec PC
MA 8 G.711 alaw 8000hz 40ms 64000bps
2011-04-30 11:07:27.937500 [NOTICE] switch_loadable_module.c:185 Adding Codec PC
MA 8 G.711 alaw 8000hz 50ms 64000bps
2011-04-30 11:07:27.937500 [NOTICE] switch_loadable_module.c:185 Adding Codec PC
MA 8 G.711 alaw 8000hz 60ms 64000bps
2011-04-30 11:07:27.937500 [NOTICE] switch_loadable_module.c:185 Adding Codec PC
MA 8 G.711 alaw 8000hz 70ms 64000bps
2011-04-30 11:07:27.937500 [NOTICE] switch_loadable_module.c:185 Adding Codec PC
MA 8 G.711 alaw 8000hz 80ms 64000bps
2011-04-30 11:07:27.937500 [NOTICE] switch_loadable_module.c:185 Adding Codec PC
MA 8 G.711 alaw 8000hz 90ms 64000bps
2011-04-30 11:07:27.937500 [NOTICE] switch_loadable_module.c:185 Adding Codec PC
MA 8 G.711 alaw 8000hz 100ms 64000bps
2011-04-30 11:07:27.937500 [NOTICE] switch_loadable_module.c:185 Adding Codec PC
MA 8 G.711 alaw 8000hz 110ms 64000bps
2011-04-30 11:07:27.937500 [NOTICE] switch_loadable_module.c:185 Adding Codec PC
MA 8 G.711 alaw 8000hz 120ms 64000bps
2011-04-30 11:07:27.968750 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_console]
2011-04-30 11:07:27.968750 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_logfile]
2011-04-30 11:07:27.984375 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_xml_rpc]
2011-04-30 11:07:27.984375 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_event_socket]
2011-04-30 11:07:29.578125 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_sofia]
2011-04-30 11:07:29.593750 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_loopback]
2011-04-30 11:07:29.859375 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_commands]
2011-04-30 11:07:29.937500 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_conference]
2011-04-30 11:07:29.937500 [CONSOLE] sofia_presence.c:946 Event Thread Started
2011-04-30 11:07:29.937500 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_dptools]
2011-04-30 11:07:29.953125 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_esf]
2011-04-30 11:07:29.968750 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_expr]
2011-04-30 11:07:30.000000 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_spandsp]
2011-04-30 11:07:30.031250 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_fifo]
2011-04-30 11:07:30.046875 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_voicemail]
2011-04-30 11:07:30.078125 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_hash]
2011-04-30 11:07:30.078125 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_vmd]
2011-04-30 11:07:30.078125 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_valet_parking]
2011-04-30 11:07:30.093750 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_directory]
2011-04-30 11:07:30.093750 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_distributor]
2011-04-30 11:07:30.093750 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_dialplan_xml]
2011-04-30 11:07:30.109375 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_voipcodecs]
2011-04-30 11:07:30.125000 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_g723_1]
2011-04-30 11:07:30.125000 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_g729]
2011-04-30 11:07:30.125000 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_amr]
2011-04-30 11:07:30.140625 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_ilbc]
2011-04-30 11:07:30.140625 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_speex]
2011-04-30 11:07:30.140625 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_siren]
2011-04-30 11:07:30.156250 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_h26x]
2011-04-30 11:07:30.156250 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_sndfile]
2011-04-30 11:07:30.171875 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_native_file]
2011-04-30 11:07:30.171875 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_shout]
2011-04-30 11:07:30.187500 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_local_stream]
2011-04-30 11:07:30.187500 [CONSOLE] mod_local_stream.c:161 Can't open directory
: C:\FreeSWITCH\sounds/music/16000
2011-04-30 11:07:30.187500 [CONSOLE] mod_local_stream.c:161 Can't open directory
: C:\FreeSWITCH\sounds/music/32000
2011-04-30 11:07:30.187500 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_tone_stream]
2011-04-30 11:07:30.203125 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_lua]
2011-04-30 11:07:30.218750 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_flite]
2011-04-30 11:07:30.234375 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_pocketsphinx]
2011-04-30 11:07:30.234375 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_say_en]
2011-04-30 11:07:30.250000 [CONSOLE] switch_loadable_module.c:950 Successfully L
oaded [mod_xml_cdr]
2011-04-30 11:07:30.250000 [CONSOLE] switch_loadable_module.c:123 Starting runti
me thread for CORE_SOFTTIMER_MODULE
2011-04-30 11:07:30.250000 [CONSOLE] switch_loadable_module.c:123 Starting runti
me thread for mod_event_socket
2011-04-30 11:07:30.250000 [CONSOLE] switch_loadable_module.c:123 Starting runti
me thread for mod_xml_rpc
Created Mutex ok

2011-04-30 11:07:30.250000 [CONSOLE] switch_core.c:1111 Created ip list lan defa
ult (allow)
2011-04-30 11:07:30.250000 [CONSOLE] switch_core.c:1111 Created ip list domains
default (deny)
2011-04-30 11:07:30.250000 [CONSOLE] switch_core.c:1829
************************************************************

2011-04-30 11:07:30.250000 [CONSOLE] switch_core.c:1832
FreeSWITCH Version 1.0.head (git-0f22877 2011-04-04 00-00-08 +0800) Started.
Max Sessions[10]
Session Rate[30]
SQL [Enabled]

FreeSWITCH at ibm>

                     Name          Type                                       Da
ta      State
================================================================================
=================
                 external       profile         sip:mod_sofia at 192.168.200.101:50
80      RUNNING (0)
                 internal       profile         sip:mod_sofia at 192.168.200.101:50
60      RUNNING (0)
          192.168.200.101         alias                                   intern
al      ALIASED
================================================================================
=================
2 profiles 1 alias



+OK console log level set to DEBUG

2011-04-30 11:08:23.296875 [INFO] mod_sofia.c:4785 EVENT_TRAP: IP change detecte
d
2011-04-30 11:08:23.296875 [INFO] mod_sofia.c:4786 IP change detected [192.168.2
00.101]->[0.0.0.0] []->[]
2011-04-30 11:08:23.671875 [NOTICE] sofia_glue.c:5119 Reload XML [Success]
2011-04-30 11:08:23.671875 [INFO] mod_pocketsphinx.c:482 PocketSphinx Reloaded
2011-04-30 11:08:23.671875 [INFO] switch_time.c:999 Timezone reloaded 530 defini
tions
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:1661 Write lock external
2011-04-30 11:08:24.156250 [NOTICE] sofia.c:1668 Waiting for worker thread
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:1725 Write unlock external
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 debug [0]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 sip-trace [no]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 rfc2833-pt [101]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 sip-port [5080]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 dialplan [XML]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:1661 Write lock internal
2011-04-30 11:08:24.156250 [NOTICE] sofia.c:1668 Waiting for worker thread
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 context [public]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 dtmf-duration [2000]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 inbound-codec-prefs [PCMA,PCMU,G
729,G723,G722,GSM]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 outbound-codec-prefs [PCMA,PCMU,
G729,G723,G722,GSM]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 hold-music [local_stream://moh]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 rtp-timer-name [soft]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 local-network-acl [localnet.auto
]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 manage-presence [false]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:1725 Write unlock internal
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 context [default]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 sip-port [5060]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 rtp-ip [0.0.0.0]
2011-04-30 11:08:24.156250 [WARNING] sofia.c:3378 Invalid IP 0.0.0.0 replaced wi
th 192.168.200.101
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 sip-ip [0.0.0.0]
2011-04-30 11:08:24.156250 [WARNING] sofia.c:3391 Invalid IP 0.0.0.0 replaced wi
th 192.168.200.101
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 apply-inbound-acl [172.28.0.0/16
]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 inbound-bypass-media [false]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 inbound-proxy-media [true]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 NDLB-received-in-nat-reg-contact
 [true]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 NDLB-force-rport [true]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 user-agent-string [FreeSWITCH]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 debug [0]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 inbound-codec-negotiation [gener
ous]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 nonce-ttl [60]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 auth-calls [false]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 rtp-ip [0.0.0.0]
2011-04-30 11:08:24.156250 [WARNING] sofia.c:3378 Invalid IP 0.0.0.0 replaced wi
th 192.168.200.101
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 sip-ip [0.0.0.0]
2011-04-30 11:08:24.156250 [WARNING] sofia.c:3391 Invalid IP 0.0.0.0 replaced wi
th 192.168.200.101
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 ext-rtp-ip [auto-nat]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 ext-sip-ip [auto-nat]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 rtp-timeout-sec [300]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 rtp-hold-timeout-sec [1800]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 tls [false]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 tls-bind-params [transport=tls]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 tls-sip-port [5081]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 tls-cert-dir [C:\FreeSWITCH/conf/ss
l]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 tls-version [tlsv1]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 sip-trace [no]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 watchdog-enabled [no]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 watchdog-step-timeout [30000]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 watchdog-event-timeout [30000]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 log-auth-failures [true]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 forward-unsolicited-mwi-notify [
false]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 rfc2833-pt [101]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 dtmf-duration [2000]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 dialplan [XML]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 inbound-codec-prefs [PCMA,PCMU,G
722,GSM]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 outbound-codec-prefs [PCMA,PCMU,
G722,GSM]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 rtp-timer-name [soft]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 hold-music [local_stream://moh]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 apply-nat-acl [nat.auto]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 local-network-acl [localnet.auto
]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 record-path [C:\FreeSWITCH\recordin
gs]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 record-template [${caller_id_num
ber}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:3087 manage-presence [false]
2011-04-30 11:08:24.156250 [DEBUG] sofia.c:1476 Creating agent for external
2011-04-30 11:08:24.156250 [NOTICE] sofia.c:4009 Started Profile external [sofia
_reg_external]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 inbound-codec-negotiation [gener
ous]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 tls [false]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 tls-bind-params [transport=tls]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 tls-sip-port [5061]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 tls-cert-dir [C:\FreeSWITCH/conf/ss
l]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 tls-version [tlsv1]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 nonce-ttl [60]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 auth-calls [true]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 inbound-reg-force-matching-usern
ame [true]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 auth-all-packets [false]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 rtp-timeout-sec [300]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 rtp-hold-timeout-sec [1800]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 force-register-domain [0.0.0.0]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 force-subscription-domain [0.0.0
.0]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 force-register-db-domain [0.0.0.
0]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 challenge-realm [auto_from]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 send-message-query-on-register [
false]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:3087 all-reg-options-ping [true]
2011-04-30 11:08:24.171875 [NOTICE] sofia.c:2265 Adding Alias [0.0.0.0] for prof
ile [internal]
2011-04-30 11:08:24.171875 [NOTICE] sofia.c:3994 Adding Alias [internal] for pro
file [internal]
2011-04-30 11:08:24.171875 [NOTICE] sofia.c:4009 Started Profile internal [sofia
_reg_internal]
2011-04-30 11:08:24.171875 [DEBUG] sofia.c:1476 Creating agent for internal
2011-04-30 11:08:24.187500 [ERR] sofia.c:1539 Error Creating SIP UA for profile:
 external
2011-04-30 11:08:24.203125 [ERR] sofia.c:1539 Error Creating SIP UA for profile:
 internal


                     Name          Type                                       Da
ta      State
================================================================================
=================
================================================================================
=================
0 profiles 0 aliases

2011-04-30 11:09:23.296875 [INFO] mod_sofia.c:4785 EVENT_TRAP: IP change detecte
d
2011-04-30 11:09:23.296875 [INFO] mod_sofia.c:4786 IP change detected [0.0.0.0]-
>[192.168.200.101] []->[]
2011-04-30 11:09:23.671875 [NOTICE] sofia_glue.c:5119 Reload XML [Success]
2011-04-30 11:09:23.671875 [INFO] mod_pocketsphinx.c:482 PocketSphinx Reloaded
2011-04-30 11:09:23.671875 [INFO] switch_time.c:999 Timezone reloaded 530 defini
tions


                     Name          Type                                       Da
ta      State
================================================================================
=================
================================================================================
=================
0 profiles 0 aliases


From boris at tagnet.ru  Sat Apr 30 08:50:56 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 30 Apr 2011 10:50:56 +0600
Subject: [Freeswitch-users] rfc2833-pt
In-Reply-To: <BANLkTi=J2FwbXWV_BKy634JpRviCp6JKFA@mail.gmail.com>
References: <4DBACE54.30809@tagnet.ru>	<BANLkTikAzag1bwhF1OMq3Rv6pnedZ7pJow@mail.gmail.com>	<4DBAD281.8070505@tagnet.ru>
	<BANLkTi=J2FwbXWV_BKy634JpRviCp6JKFA@mail.gmail.com>
Message-ID: <4DBB9530.6000104@tagnet.ru>

Hello!

     Thank You, Kristian!
> According to the IANA payload types:
>
> http://www.iana.org/assignments/rtp-parameters
>
> 96-127 are dynamic.  Anything is available to use them.  The
> "limitations" you are describing come from Audiocodes.  In my
> experience with a wide variety of endpoints 101 is fairly
> typical/standard for RFC 2833.
>
>
> On Fri, Apr 29, 2011 at 11:00 AM, Boris Kovalenko<boris at tagnet.ru>  wrote:
>> Hello!
>>
>> Ok. And why FS uses 101 while valid are 96 to 99? Or this means valid
>> only for MP-118?
>>
>>> You don't have to change it.  SDP will handle it for you.
>>>
>>> I've worked with many AudioCodes gateways without every change the
>>> RFC2833 payload type.
>>>
>>> On Fri, Apr 29, 2011 at 10:42 AM, Boris Kovalenko<boris at tagnet.ru>    wrote:
>>>> Hello!
>>>>
>>>>       In the Audiocodecs MP-118 User's Manual "The RFC 2833 DTMF relay
>>>> dynamic payload type. The valid range is 96 to 99, and 106 to 127. The
>>>> default is 96. The 100, 102 to 105 range is allocated for proprietary
>>>> usage." But FS uses 101 by default. Is this ok or should I change it to
>>>> 96 for example?
>>>>
>>>> --
>>>> 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
>>>>
>>>
>>
>> --
>> ? ?????????,
>>    ????? ?????????
>>    ??? "??????"
>>    ???. +7 (3435) 230001
>>    ???? +7 (3435) 230005
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>


-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



From boris at tagnet.ru  Sat Apr 30 09:00:01 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 30 Apr 2011 11:00:01 +0600
Subject: [Freeswitch-users] RFC2833 and beeps
Message-ID: <4DBB9751.6050208@tagnet.ru>

Hello!

     My network configuration:
Linksys SPA8000 -> FS -> Cisco 5350 -> PSTN
FS: FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)
Sometimes, when an user (female) on the Linksys speaks, on the other end 
(PSTN) there are "beeps" like DTMF. After googling I found that this 
problem is reproducable on some types of hardware where the female 
russian voice is incorretly recognized as DTMF. So may I do  some 
workaround with FS? For example to disable RFC2833 on Linksys and use 
start_dtmf before bridge?

-- 
? ?????????,
   ????? ?????????
   ??? "??????"
   ???. +7 (3435) 230001
   ???? +7 (3435) 230005



From fieldpeak at gmail.com  Sat Apr 30 09:34:37 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sat, 30 Apr 2011 13:34:37 +0800
Subject: [Freeswitch-users] FS - SIP profiles crashed
In-Reply-To: <BANLkTinhWi3Q7x4B3hM60w+1F96htTVpvA@mail.gmail.com>
References: <BANLkTi=402RnR54vQmVx7_fh8z1iaX3aFw@mail.gmail.com>
	<1304090527690-6317287.post@n2.nabble.com>
	<BANLkTinqHZb1A0Fqz1z+BK+UeM2vUfWG9A@mail.gmail.com>
	<4dbb2f69.81bf0e0a.4bad.2812@mx.google.com>
	<BANLkTimYon6k0qvZtUnQWY0MVMS6AYEhmw@mail.gmail.com>
	<BANLkTikXtcGECS_tZPMcS+4Kq_mvASgnXA@mail.gmail.com>
	<BANLkTinhWi3Q7x4B3hM60w+1F96htTVpvA@mail.gmail.com>
Message-ID: <BANLkTim1Bp=t-XMQJmi9ABrvfY=fr+CnGg@mail.gmail.com>

it looks the same issue as
http://jira.freeswitch.org/browse/FS-933?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel

however, where to add the new parameter: PFLAG_SKIP_RESTART
?

2011/4/30 fieldpeak <fieldpeak at gmail.com>

> oh, looks i found the root cause,
>
> after i unplug the network cable and wait around 20 seconds, FS show ' IP
> address changed to '0.0.0.0', then try to load profiles, however, due to
> wrong IP address, load failure.
> and then i re-plugin the cable, FS detected the IP change back, and
> reloadxml, however, did not restart the profiles, so caused profiles
> crashed... attached is the log for throughout the procedure...
>
> the resolution is set
>
> in the sofia.conf.xml file:
>
> <param name="auto-restart" value="false"/>
>
> welcome to any comment,
>
> thanks.
>
> Regards,
> Charles
>
>
>
> 2011/4/30 fieldpeak <fieldpeak at gmail.com>
>
>> really want to know what caused that the IP address changed to '0.0.0.0',
>> very odd...
>> i tried even unplug and plugin the network cable, it will not change the
>> IP...let alone 0.0.0.0...
>>
>>
>> 2011/4/30 fieldpeak <fieldpeak at gmail.com>
>>
>>> Thanks all for information, it definitely help a lot. i will try...
>>>
>>> meanwhile, i found this link
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-May/003086.html
>>>
>>> it looks not so special resolve this issue...
>>>
>>> i would like to know what is use for "bind_server_ip=auto" and where and
>>> when FS get the value of $${local_ip_v4}? and if it will changed when
>>> physical IP changed?
>>>
>>> Thanks!
>>>
>>> Regards,
>>> Charles
>>>
>>> 2011/4/30 Grmt <garmt.noname at gmail.com>
>>>
>>>>
>>>> http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address
>>>>
>>>>
>>>>
>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
>>>> Collins
>>>> *Sent:* Friday, 29 April, 2011 19:35
>>>> *To:* FreeSWITCH Users Help
>>>> *Subject:* Re: [Freeswitch-users] FS - SIP profiles crashed
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> On Fri, Apr 29, 2011 at 8:22 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>>>>
>>>> If your box is configured with a static IP do this.
>>>>
>>>> conf/autoload_configs/sofia.conf.xml --
>>>>
>>>>
>>>>
>>>> Jeff, was there more to this 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
>>>>
>>>>
>>>
>>
>
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From fieldpeak at gmail.com  Sat Apr 30 09:39:18 2011
From: fieldpeak at gmail.com (fieldpeak)
Date: Sat, 30 Apr 2011 13:39:18 +0800
Subject: [Freeswitch-users] FS - SIP profiles crashed
In-Reply-To: <BANLkTim1Bp=t-XMQJmi9ABrvfY=fr+CnGg@mail.gmail.com>
References: <BANLkTi=402RnR54vQmVx7_fh8z1iaX3aFw@mail.gmail.com>
	<1304090527690-6317287.post@n2.nabble.com>
	<BANLkTinqHZb1A0Fqz1z+BK+UeM2vUfWG9A@mail.gmail.com>
	<4dbb2f69.81bf0e0a.4bad.2812@mx.google.com>
	<BANLkTimYon6k0qvZtUnQWY0MVMS6AYEhmw@mail.gmail.com>
	<BANLkTikXtcGECS_tZPMcS+4Kq_mvASgnXA@mail.gmail.com>
	<BANLkTinhWi3Q7x4B3hM60w+1F96htTVpvA@mail.gmail.com>
	<BANLkTim1Bp=t-XMQJmi9ABrvfY=fr+CnGg@mail.gmail.com>
Message-ID: <BANLkTikfsdWxJR5NnzqxFj6JfajcDxRF-A@mail.gmail.com>

sorry, my misunderstanding, it looks this issue fixed by FS-933 on
21/Apr/10, however, my GIT is April 4, 2001, it still exist... strange...

2011/4/30 fieldpeak <fieldpeak at gmail.com>

> it looks the same issue as
> http://jira.freeswitch.org/browse/FS-933?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel
>
> however, where to add the new parameter: PFLAG_SKIP_RESTART
>
> ?
>
> 2011/4/30 fieldpeak <fieldpeak at gmail.com>
>
>> oh, looks i found the root cause,
>>
>> after i unplug the network cable and wait around 20 seconds, FS show ' IP
>> address changed to '0.0.0.0', then try to load profiles, however, due to
>> wrong IP address, load failure.
>> and then i re-plugin the cable, FS detected the IP change back, and
>> reloadxml, however, did not restart the profiles, so caused profiles
>> crashed... attached is the log for throughout the procedure...
>>
>> the resolution is set
>>
>> in the sofia.conf.xml file:
>>
>> <param name="auto-restart" value="false"/>
>>
>> welcome to any comment,
>>
>> thanks.
>>
>> Regards,
>> Charles
>>
>>
>>
>> 2011/4/30 fieldpeak <fieldpeak at gmail.com>
>>
>>> really want to know what caused that the IP address changed to '0.0.0.0',
>>> very odd...
>>> i tried even unplug and plugin the network cable, it will not change the
>>> IP...let alone 0.0.0.0...
>>>
>>>
>>> 2011/4/30 fieldpeak <fieldpeak at gmail.com>
>>>
>>>> Thanks all for information, it definitely help a lot. i will try...
>>>>
>>>> meanwhile, i found this link
>>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-May/003086.html
>>>>
>>>> it looks not so special resolve this issue...
>>>>
>>>> i would like to know what is use for "bind_server_ip=auto" and where and
>>>> when FS get the value of $${local_ip_v4}? and if it will changed when
>>>> physical IP changed?
>>>>
>>>> Thanks!
>>>>
>>>> Regards,
>>>> Charles
>>>>
>>>> 2011/4/30 Grmt <garmt.noname at gmail.com>
>>>>
>>>>>
>>>>> http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP_address
>>>>>
>>>>>
>>>>>
>>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
>>>>> Collins
>>>>> *Sent:* Friday, 29 April, 2011 19:35
>>>>> *To:* FreeSWITCH Users Help
>>>>> *Subject:* Re: [Freeswitch-users] FS - SIP profiles crashed
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Fri, Apr 29, 2011 at 8:22 AM, Jeff Lenk <jeff at jefflenk.com> wrote:
>>>>>
>>>>> If your box is configured with a static IP do this.
>>>>>
>>>>> conf/autoload_configs/sofia.conf.xml --
>>>>>
>>>>>
>>>>>
>>>>> Jeff, was there more to this 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
>>>>>
>>>>>
>>>>
>>>
>>
>
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From kbdfck at gmail.com  Sat Apr 30 10:03:28 2011
From: kbdfck at gmail.com (Dmitry Sytchev)
Date: Sat, 30 Apr 2011 10:03:28 +0400
Subject: [Freeswitch-users] RFC2833 and beeps
In-Reply-To: <4DBB9751.6050208@tagnet.ru>
References: <4DBB9751.6050208@tagnet.ru>
Message-ID: <BANLkTi=iXzpfWpMRiV2tvqAcsuqm0n0A1g@mail.gmail.com>

This is called talkoff and there are countermeasures in hardware like
Linksys PAP2T/SPA8000. DTMF mode can be set to 'Strict', and Strict Holdoff
TIme to 80-90.
After that DTMF will not be triggered by voice, but inband digits muting to
convert to RFC2833 will work less reliable, so there can be DTMF artifacts
as in our case.

2011/4/30 Boris Kovalenko <boris at tagnet.ru>

> Hello!
>
>     My network configuration:
> Linksys SPA8000 -> FS -> Cisco 5350 -> PSTN
> FS: FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50 -0300)
> Sometimes, when an user (female) on the Linksys speaks, on the other end
> (PSTN) there are "beeps" like DTMF. After googling I found that this
> problem is reproducable on some types of hardware where the female
> russian voice is incorretly recognized as DTMF. So may I do  some
> workaround with FS? For example to disable RFC2833 on Linksys and use
> start_dtmf before bridge?
>
> --
> ? ?????????,
>   ????? ?????????
>   ??? "??????"
>   ???. +7 (3435) 230001
>   ???? +7 (3435) 230005
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at 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 gmaruzz at gmail.com  Sat Apr 30 12:31:39 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 30 Apr 2011 10:31:39 +0200
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
Message-ID: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>

Dear FreeSWITCHers,

after a fair amount of effort, I ended up with a new way to install
and use mod_skypopen on Linux.

No more looking around the internet for the lost 2.0.0.72 Skype client for ALSA.

First, we can use the readily available Skype client for OSS.

(added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS
driver, that's very easy to compile and install, and do not need to
mess with the operating system installation.)

Second, I wrote an installer that automatically do all the tedious
work for you: download and install the skype client, create the config
directory for Skype clients, create the config file for mod_skypopen,
create the script that launches the Skype clients.

I hope those improvements will lower the barriers for Skype calls on FreeSWITCH.

Actually is ludicrously simple now, and after you compile FreeSWITCH,
mod_skypopen and the skypopen.ko OSS driver it will take like less
than one minute to have a complete installation of mod_skypopen ready
to make and receive calls.

All automatic, no more need to fiddle around with the Skype client
download, configurations, authorization, etc.

Is all well tested, but maybe there are still some bugs, and maybe the
docs are not clear/easy enough.

Please have a look at the new and improved wiki page and let me know
what do you think about (and maybe test the procedures).

http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk
http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux
http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR

You must update to the latest git to have all the goodies.

Thank you all for your support,

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From boris at tagnet.ru  Sat Apr 30 14:03:28 2011
From: boris at tagnet.ru (Boris Kovalenko)
Date: Sat, 30 Apr 2011 16:03:28 +0600
Subject: [Freeswitch-users] RFC2833 and beeps
In-Reply-To: <BANLkTi=iXzpfWpMRiV2tvqAcsuqm0n0A1g@mail.gmail.com>
References: <4DBB9751.6050208@tagnet.ru>
	<BANLkTi=iXzpfWpMRiV2tvqAcsuqm0n0A1g@mail.gmail.com>
Message-ID: <4DBBDE70.7010000@tagnet.ru>

Thank You, Dmitry!

> This is called talkoff and there are countermeasures in hardware like 
> Linksys PAP2T/SPA8000. DTMF mode can be set to 'Strict', and Strict 
> Holdoff TIme to 80-90.
> After that DTMF will not be triggered by voice, but inband digits 
> muting to convert to RFC2833 will work less reliable, so there can be 
> DTMF artifacts as in our case.
>
> 2011/4/30 Boris Kovalenko <boris at tagnet.ru <mailto:boris at tagnet.ru>>
>
>     Hello!
>
>         My network configuration:
>     Linksys SPA8000 -> FS -> Cisco 5350 -> PSTN
>     FS: FreeSWITCH Version 1.0.head (git-1c95ad9 2011-01-20 22-43-50
>     -0300)
>     Sometimes, when an user (female) on the Linksys speaks, on the
>     other end
>     (PSTN) there are "beeps" like DTMF. After googling I found that this
>     problem is reproducable on some types of hardware where the female
>     russian voice is incorretly recognized as DTMF. So may I do  some
>     workaround with FS? For example to disable RFC2833 on Linksys and use
>     start_dtmf before bridge?
>
>     --
>     ? ?????????,
>       ????? ?????????
>       ??? "??????"
>       ???. +7 (3435) 230001
>       ???? +7 (3435) 230005
>
>
>     _______________________________________________
>     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
>
>
>
>
> -- 
> 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


-- 
Regards,
Boris


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From christian.loeschenkohl at xpirio.com  Sat Apr 30 14:45:53 2011
From: christian.loeschenkohl at xpirio.com (=?UTF-8?B?Q2hyaXN0aWFuIEzDtnNjaGVua29obA==?=)
Date: Sat, 30 Apr 2011 12:45:53 +0200
Subject: [Freeswitch-users] Trunking between Lync and FreeSWITCH
In-Reply-To: <3120C73F-FFD6-4979-87FD-A61AA1A2E14C@ipeva.fr>
References: <3120C73F-FFD6-4979-87FD-A61AA1A2E14C@ipeva.fr>
Message-ID: <4DBBE861.9050906@xpirio.com>

hi

we have kind of the same issue in our lync test setup

- outbound calls work with two-way audio and with no disconnection
- inbound calls work only with one-way audio and the disconnection from
the lync side after exact 32 seconds

at the moment it looks like a firewall/nat problem to us (firewall is now a isa server).
the next test will be done with an iptables firewall

if something new comes up, i'll post it here

br


On 2011-04-28 23:24, David Ponzone wrote:

> Well, I thought I could start a thread about that.
> I am trying to accomplish this, and I guess I am not the only one.
> Lync 2010 seems to have quite some success (compared to OCS), so there are probably business opportunities around this.
>
> My first attempt is a half success, as I got calls from Lync to FreeSWITCH working (need some extensive testing, though), and calls from FreeSWITCH to Lync are connecting ok, but they get disconnected after circa 30 seconds.
>
> I am trying to pinpoint the reason for this disconnection, but untll now, I had no luck.
> RTCP requirement was disabled on the Lync side (RTCPActiveCalls), but anyway I have enabled RTCP on the FreeSWITCH, so it should not be that.
> RTP is flowing normally, it's not a call on hold or muted.
> Refer was disabled on the Lync side.
> SessionTimer is enabled on the Lync side (but I dont think I ever saw a REINVITE caused by the session timer).
> I just suddenly receive a BYE from Lync, after 30-35 seconds.
> On the Lync side, as you can guess, it's not easy to access any meaningful logs. I think our partner managing the Lync will have to escalate that to the SVP Product Engineering to get the right command to enable the interesting debug mode :)
>
> I've been told that some folks at MS were claiming a such trunking was not possible.
> Well, I would tend to say otherwise, as I am not far to get this working, except of course if they just hardcoded some forbidden Owner Usernames in the SDP, like FreeSWITCH, in order to save this business from the mean OpenSource world and leave it to commercial SBC vendors.
> I can hardly imagine they would dare to do that in 2011, so the question remains opened: what may be closing that call after 30 seconds ?
>
> I would be glad to discuss the subject, here or privately, with anyone who is involved on a such project, or planning to be soon.
>
> David Ponzone Direction Technique
> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
> tel: 01 74 03 18 97
> gsm: 06 66 98 76 34
>
> Service ClientIPeva
> tel: 0811 46 26 26
> www.ipeva.fr <BLOCKED::http://www.ipeva.fr/>- www.ipeva-studio.com <BLOCKED::http://www.ipeva-studio.com/>
>
> /Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. /*/IPeva/*/d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur./
> /
> /
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


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

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

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


From kerem.erciyes at gmail.com  Sat Apr 30 17:34:52 2011
From: kerem.erciyes at gmail.com (Kerem Erciyes)
Date: Sat, 30 Apr 2011 16:34:52 +0300
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
Message-ID: <BANLkTimdDaTA_vaUrzFxavbVWAyCrSu94A@mail.gmail.com>

Hi Giovanni,

You released it just as I finished out testing ang translating the old way.
Well, no worries, I will get on to it right away and try to see if it works
out of the box.

Keep up the hard work,
Kerem

On Sat, Apr 30, 2011 at 11:31 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> Dear FreeSWITCHers,
>
> after a fair amount of effort, I ended up with a new way to install
> and use mod_skypopen on Linux.
>
> No more looking around the internet for the lost 2.0.0.72 Skype client for
> ALSA.
>
> First, we can use the readily available Skype client for OSS.
>
> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS
> driver, that's very easy to compile and install, and do not need to
> mess with the operating system installation.)
>
> Second, I wrote an installer that automatically do all the tedious
> work for you: download and install the skype client, create the config
> directory for Skype clients, create the config file for mod_skypopen,
> create the script that launches the Skype clients.
>
> I hope those improvements will lower the barriers for Skype calls on
> FreeSWITCH.
>
> Actually is ludicrously simple now, and after you compile FreeSWITCH,
> mod_skypopen and the skypopen.ko OSS driver it will take like less
> than one minute to have a complete installation of mod_skypopen ready
> to make and receive calls.
>
> All automatic, no more need to fiddle around with the Skype client
> download, configurations, authorization, etc.
>
> Is all well tested, but maybe there are still some bugs, and maybe the
> docs are not clear/easy enough.
>
> Please have a look at the new and improved wiki page and let me know
> what do you think about (and maybe test the procedures).
>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux
>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR
>
> You must update to the latest git to have all the goodies.
>
> Thank you all for your support,
>
> -giovanni
>
> --
> 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
>



-- 
Kerem Erciyes - Sistem Danismani
http://keremerciyes.com
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From anton.vazir at gmail.com  Sat Apr 30 18:31:29 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 30 Apr 2011 19:31:29 +0500
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTimdDaTA_vaUrzFxavbVWAyCrSu94A@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<BANLkTimdDaTA_vaUrzFxavbVWAyCrSu94A@mail.gmail.com>
Message-ID: <BANLkTimW3X=D0FDiWmoHuONhnTtiZu+VWQ@mail.gmail.com>

Waw. Great! Thanks so much Giovanni!

2011/4/30 Kerem Erciyes <kerem.erciyes at gmail.com>:
> Hi Giovanni,
>
> You released it just as I finished out testing ang translating the old way.
> Well, no worries, I will get on to it right away and try to see if it works
> out of the box.
>
> Keep up the hard work,
> Kerem
>
> On Sat, Apr 30, 2011 at 11:31 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:
>>
>> Dear FreeSWITCHers,
>>
>> after a fair amount of effort, I ended up with a new way to install
>> and use mod_skypopen on Linux.
>>
>> No more looking around the internet for the lost 2.0.0.72 Skype client for
>> ALSA.
>>
>> First, we can use the readily available Skype client for OSS.
>>
>> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS
>> driver, that's very easy to compile and install, and do not need to
>> mess with the operating system installation.)
>>
>> Second, I wrote an installer that automatically do all the tedious
>> work for you: download and install the skype client, create the config
>> directory for Skype clients, create the config file for mod_skypopen,
>> create the script that launches the Skype clients.
>>
>> I hope those improvements will lower the barriers for Skype calls on
>> FreeSWITCH.
>>
>> Actually is ludicrously simple now, and after you compile FreeSWITCH,
>> mod_skypopen and the skypopen.ko OSS driver it will take like less
>> than one minute to have a complete installation of mod_skypopen ready
>> to make and receive calls.
>>
>> All automatic, no more need to fiddle around with the Skype client
>> download, configurations, authorization, etc.
>>
>> Is all well tested, but maybe there are still some bugs, and maybe the
>> docs are not clear/easy enough.
>>
>> Please have a look at the new and improved wiki page and let me know
>> what do you think about (and maybe test the procedures).
>>
>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk
>>
>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux
>>
>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR
>>
>> You must update to the latest git to have all the goodies.
>>
>> Thank you all for your support,
>>
>> -giovanni
>>
>> --
>> 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
>
>
>
> --
> Kerem Erciyes - Sistem Danismani
> http://keremerciyes.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 anton.vazir at gmail.com  Sat Apr 30 18:43:13 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 30 Apr 2011 19:43:13 +0500
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTimW3X=D0FDiWmoHuONhnTtiZu+VWQ@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<BANLkTimdDaTA_vaUrzFxavbVWAyCrSu94A@mail.gmail.com>
	<BANLkTimW3X=D0FDiWmoHuONhnTtiZu+VWQ@mail.gmail.com>
Message-ID: <BANLkTinB=0ZUW8X47_gtpBrShVxUGm-NCA@mail.gmail.com>

Giovanni,

the kernel module provides just a sound device? So it means version of
skype needed is the same, just OSS one can be used instead?
Any other pros, like less CPU usage, or other for oss version?

btw, any info on the biggest known skypopen installation, like 10
running clients, 100 clients, or more?


2011/4/30 Anton VG <anton.vazir at gmail.com>:
> Waw. Great! Thanks so much Giovanni!
>
> 2011/4/30 Kerem Erciyes <kerem.erciyes at gmail.com>:
>> Hi Giovanni,
>>
>> You released it just as I finished out testing ang translating the old way.
>> Well, no worries, I will get on to it right away and try to see if it works
>> out of the box.
>>
>> Keep up the hard work,
>> Kerem
>>
>> On Sat, Apr 30, 2011 at 11:31 AM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:
>>>
>>> Dear FreeSWITCHers,
>>>
>>> after a fair amount of effort, I ended up with a new way to install
>>> and use mod_skypopen on Linux.
>>>
>>> No more looking around the internet for the lost 2.0.0.72 Skype client for
>>> ALSA.
>>>
>>> First, we can use the readily available Skype client for OSS.
>>>
>>> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS
>>> driver, that's very easy to compile and install, and do not need to
>>> mess with the operating system installation.)
>>>
>>> Second, I wrote an installer that automatically do all the tedious
>>> work for you: download and install the skype client, create the config
>>> directory for Skype clients, create the config file for mod_skypopen,
>>> create the script that launches the Skype clients.
>>>
>>> I hope those improvements will lower the barriers for Skype calls on
>>> FreeSWITCH.
>>>
>>> Actually is ludicrously simple now, and after you compile FreeSWITCH,
>>> mod_skypopen and the skypopen.ko OSS driver it will take like less
>>> than one minute to have a complete installation of mod_skypopen ready
>>> to make and receive calls.
>>>
>>> All automatic, no more need to fiddle around with the Skype client
>>> download, configurations, authorization, etc.
>>>
>>> Is all well tested, but maybe there are still some bugs, and maybe the
>>> docs are not clear/easy enough.
>>>
>>> Please have a look at the new and improved wiki page and let me know
>>> what do you think about (and maybe test the procedures).
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux
>>>
>>> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR
>>>
>>> You must update to the latest git to have all the goodies.
>>>
>>> Thank you all for your support,
>>>
>>> -giovanni
>>>
>>> --
>>> 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
>>
>>
>>
>> --
>> Kerem Erciyes - Sistem Danismani
>> http://keremerciyes.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 david.ponzone at ipeva.fr  Sat Apr 30 19:16:37 2011
From: david.ponzone at ipeva.fr (David Ponzone)
Date: Sat, 30 Apr 2011 17:16:37 +0200
Subject: [Freeswitch-users] Trunking between Lync and FreeSWITCH
In-Reply-To: <4DBBE861.9050906@xpirio.com>
References: <3120C73F-FFD6-4979-87FD-A61AA1A2E14C@ipeva.fr>
	<4DBBE861.9050906@xpirio.com>
Message-ID: <761372A6-B060-42ED-8553-9EB3FFF13F7B@ipeva.fr>

Christian,

that's interesting, and I think I can help you to solve your one-way audio issue with inbound calls :)
You have to force the outbound codec from FS to Lync to be PCMU.
If you send it both PCMA and PCMU or PCMA only, the stupid Lync agrees on PCMA, but will send you PCMU anyway.
And those are the guys who are saying they follow the RFCs to the letter....

That should solve it.
And at the same time, this will probably eliminate your NAT/FW hypothesis, and I am sorry for that :)
With a normal audio,  and if you enable it, a normal RTCP flow, it becomes more difficult to understand why the call is disconnected, even before there is one REINVITE caused by session-timers.

I got some info from MS.
I got some web links with completely useless information (the currently certified equipements and the lengthy and painful process to be come certified):
http://technet.microsoft.com/en-us/lync/gg131938

Also some interesting info:
http://technet.microsoft.com/en-us/library/gg398619.aspx

What is interesting in that last document is that they never talk about a NAT topology. They only show examples without NAT.
So the question is: is it because they have issues with NAT that even FS can't circumvent, or is it because they wan't to discourage people to run SIP trunk to Lync over an unmanaged public Internet ?
As the outbound calls are working fine, I would tend to think the NAT is not an issue, and that's just MS being overcautious.

On the Lync side, it seems the amount of useful debugging info that can be obtained is close to 0, but I don't have access to the Lync, so I could be wrong.
You could probably confirm that point.

David Ponzone  Direction Technique
email: david.ponzone at ipeva.fr
tel:      01 74 03 18 97
gsm:   06 66 98 76 34

Service Client IPeva
tel:      0811 46 26 26
www.ipeva.fr  -   www.ipeva-studio.com

Ce message et toutes les pi?ces jointes sont confidentiels et ?tablis ? l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autoris?e est interdite. Tout message ?lectronique est susceptible d'alt?ration. IPeva d?cline toute responsabilit? au titre de ce message s'il a ?t? alt?r?, d?form? ou falsifi?. Si vous n'?tes pas destinataire de ce message, merci de le d?truire imm?diatement et d'avertir l'exp?diteur.



Le 30/04/2011 ? 12:45, Christian L?schenkohl a ?crit :

> hi
> 
> we have kind of the same issue in our lync test setup
> 
> - outbound calls work with two-way audio and with no disconnection
> - inbound calls work only with one-way audio and the disconnection from
> the lync side after exact 32 seconds
> 
> at the moment it looks like a firewall/nat problem to us (firewall is now a isa server).
> the next test will be done with an iptables firewall
> 
> if something new comes up, i'll post it here
> 
> br
> 
> 
> On 2011-04-28 23:24, David Ponzone wrote:
> 
>> Well, I thought I could start a thread about that.
>> I am trying to accomplish this, and I guess I am not the only one.
>> Lync 2010 seems to have quite some success (compared to OCS), so there are probably business opportunities around this.
>> 
>> My first attempt is a half success, as I got calls from Lync to FreeSWITCH working (need some extensive testing, though), and calls from FreeSWITCH to Lync are connecting ok, but they get disconnected after circa 30 seconds.
>> 
>> I am trying to pinpoint the reason for this disconnection, but untll now, I had no luck.
>> RTCP requirement was disabled on the Lync side (RTCPActiveCalls), but anyway I have enabled RTCP on the FreeSWITCH, so it should not be that.
>> RTP is flowing normally, it's not a call on hold or muted.
>> Refer was disabled on the Lync side.
>> SessionTimer is enabled on the Lync side (but I dont think I ever saw a REINVITE caused by the session timer).
>> I just suddenly receive a BYE from Lync, after 30-35 seconds.
>> On the Lync side, as you can guess, it's not easy to access any meaningful logs. I think our partner managing the Lync will have to escalate that to the SVP Product Engineering to get the right command to enable the interesting debug mode :)
>> 
>> I've been told that some folks at MS were claiming a such trunking was not possible.
>> Well, I would tend to say otherwise, as I am not far to get this working, except of course if they just hardcoded some forbidden Owner Usernames in the SDP, like FreeSWITCH, in order to save this business from the mean OpenSource world and leave it to commercial SBC vendors.
>> I can hardly imagine they would dare to do that in 2011, so the question remains opened: what may be closing that call after 30 seconds ?
>> 
>> I would be glad to discuss the subject, here or privately, with anyone who is involved on a such project, or planning to be soon.
>> 
>> David Ponzone Direction Technique
>> email: david.ponzone at ipeva.fr <mailto:david.ponzone at ipeva.fr>
>> tel: 01 74 03 18 97
>> gsm: 06 66 98 76 34

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From Nabble at slickdeals.endjunk.com  Sat Apr 30 19:58:46 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 30 Apr 2011 08:58:46 -0700 (PDT)
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
Message-ID: <1304179126964-6319822.post@n2.nabble.com>


Giovanni Maruzzelli-2 wrote:
> No more looking around the internet for the lost 2.0.0.72 Skype client for
> ALSA.
> 
> First, we can use the readily available Skype client for OSS.
Argh ..., a Skype client is still needed. I thought you had hacked Skype and
came up with a mod_skypopen that already includes Skype protocol. 
-----FreeSWITCH hosted on a Seagate DockStar with OpenWRT.--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/NEW-skypopen-installer-and-easy-Skype-client-download-Skype-calls-on-FreeSWITCH-in-one-minute-tp6319154p6319822.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From gmaruzz at celliax.org  Sat Apr 30 20:28:02 2011
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sat, 30 Apr 2011 18:28:02 +0200
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTimdDaTA_vaUrzFxavbVWAyCrSu94A@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<BANLkTimdDaTA_vaUrzFxavbVWAyCrSu94A@mail.gmail.com>
Message-ID: <BANLkTino7Yfk2aoBDQH9YoaBC6_4JffKCg@mail.gmail.com>

On Sat, Apr 30, 2011 at 3:34 PM, Kerem Erciyes <kerem.erciyes at gmail.com> wrote:
> You released it just as I finished out testing ang translating the old way.
> Well, no worries, I will get on to it right away and try to see if it works
> out of the box.

Ooops! :)
Let us know how it goes for you and where the new translation will be located.
Thanks a lot Kerem!
-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Sat Apr 30 20:40:37 2011
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sat, 30 Apr 2011 18:40:37 +0200
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTinB=0ZUW8X47_gtpBrShVxUGm-NCA@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<BANLkTimdDaTA_vaUrzFxavbVWAyCrSu94A@mail.gmail.com>
	<BANLkTimW3X=D0FDiWmoHuONhnTtiZu+VWQ@mail.gmail.com>
	<BANLkTinB=0ZUW8X47_gtpBrShVxUGm-NCA@mail.gmail.com>
Message-ID: <BANLkTinFLeiU6T220h0NCTtRLu-jDRjqOg@mail.gmail.com>

On Sat, Apr 30, 2011 at 4:43 PM, Anton VG <anton.vazir at gmail.com> wrote:
> the kernel module provides just a sound device? So it means version of

The kernel module is a fake OSS audio driver and device.
> skype needed is the same, just OSS one can be used instead?

No, you need the Skype client for OSS. Is downloaded and installed
automatically by the installer. (or you can look into the installer
and see where you can download it).

> Any other pros, like less CPU usage, or other for oss version?

I found less CPU usage, better stability, and no overhead from ALSA.

> btw, any info on the biggest known skypopen installation, like 10
> running clients, 100 clients, or more?

I always test with 20 concurrent clients (I registered 20
skypeusernames before the registration procedure has become so
burdening as today). I will test soon or later with more (maybe also
many more) but the very boring thing is that you need many different
Skype username for testing. I mean: you can have a server that has 100
instances of the same skypeusername (let's say Bob), but the incoming
calls must come from Alice, Adam, etc...

So I will need to find the patience to register with the skype network
let's say 100 skypeusernames to test 100 concurrent calls.

Another way to test would be to have a skypeusername with some credit
for skypeout, and use let's say 100 instances to call skypeout on
PSTN.

Any other ideas on how to proceed for scalability tests?

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at celliax.org  Sat Apr 30 20:43:49 2011
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sat, 30 Apr 2011 18:43:49 +0200
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <1304179126964-6319822.post@n2.nabble.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<1304179126964-6319822.post@n2.nabble.com>
Message-ID: <BANLkTinYd64kyirHuOKV64Hc_SVsmO+pzA@mail.gmail.com>

On Sat, Apr 30, 2011 at 5:58 PM, mazilo <Nabble at slickdeals.endjunk.com> wrote:
> Argh ..., a Skype client is still needed. I thought you had hacked Skype and
> came up with a mod_skypopen that already includes Skype protocol.

Hahaha, poor Mazilo, sorry to have you disilluded!

Hacking Skype, reverse engineer the protocol, and any other access to
the Skype network that does not use a binary officially developed and
distributed and authorized by Skype corporation is explicitly illegal
and can bring crimilan charges.

mod_skypopen is completely legal and completely abiding to Skype
corporation policies etc.

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From anton.vazir at gmail.com  Sat Apr 30 20:52:18 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 30 Apr 2011 21:52:18 +0500
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTinFLeiU6T220h0NCTtRLu-jDRjqOg@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<BANLkTimdDaTA_vaUrzFxavbVWAyCrSu94A@mail.gmail.com>
	<BANLkTimW3X=D0FDiWmoHuONhnTtiZu+VWQ@mail.gmail.com>
	<BANLkTinB=0ZUW8X47_gtpBrShVxUGm-NCA@mail.gmail.com>
	<BANLkTinFLeiU6T220h0NCTtRLu-jDRjqOg@mail.gmail.com>
Message-ID: <BANLkTim7aXhnJEZHHpWFo_kNXYinfhpKTg@mail.gmail.com>

> Any other ideas on how to proceed for scalability tests?
>

Hm, since 2.0.0.72 still allows simultaneous calls, you do not have to
register lots of usernames, just 2 of them for 2 PC's
than make 100 calls from 1 machine to another


From gmaruzz at gmail.com  Sat Apr 30 21:01:54 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 30 Apr 2011 19:01:54 +0200
Subject: [Freeswitch-users] skypopen scalability tests
Message-ID: <BANLkTinMytjFaMc4zA0pfmgV6dMqEHu7WQ@mail.gmail.com>

On Sat, Apr 30, 2011 at 6:52 PM, Anton VG <anton.vazir at gmail.com> wrote:
>> Any other ideas on how to proceed for scalability tests?
>>
>
> Hm, since 2.0.0.72 still allows simultaneous calls, you do not have to
> register lots of usernames, just 2 of them for 2 PC's
> than make 100 calls from 1 machine to another

I was going that way, couple years ago, but the Skype client was
scaling kind of badly for me. No more than a bunch of simultaneus
calls.
But at that time I was testing using a standard ALSA sound driver, and
Skype client for ALSA.
Maybe using Skype for OSS and skypopen.ko it will scale better.

Good hint, at least I will check into it in the future.

Other ideas?

-giovanni

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From gmaruzz at gmail.com  Sat Apr 30 21:09:50 2011
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sat, 30 Apr 2011 19:09:50 +0200
Subject: [Freeswitch-users] skypopen scalability tests
In-Reply-To: <BANLkTinMytjFaMc4zA0pfmgV6dMqEHu7WQ@mail.gmail.com>
References: <BANLkTinMytjFaMc4zA0pfmgV6dMqEHu7WQ@mail.gmail.com>
Message-ID: <BANLkTimnJ1wRTmVRJ8sw1ed57HdKUZxDZA@mail.gmail.com>

Ooops, I was wrong in my previous mail (my memory no more what was
used to be :) ).

You cannot have simultaneus calls from many instances of Bob to many
instances of Alice. It just does not works reliably.

I was testing simultaneus calls from one client skypeusername (Bob) to
a server with many skypeusernames (Alice, Adam, etc).

So, we're back in the situation I described (need to register many
different skypeusernames) with the added drawback that simultaneus
calls from one Skype client does not work well.

Any other ideas?
-giovanni

On Sat, Apr 30, 2011 at 7:01 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> On Sat, Apr 30, 2011 at 6:52 PM, Anton VG <anton.vazir at gmail.com> wrote:
>>> Any other ideas on how to proceed for scalability tests?
>>>
>>
>> Hm, since 2.0.0.72 still allows simultaneous calls, you do not have to
>> register lots of usernames, just 2 of them for 2 PC's
>> than make 100 calls from 1 machine to another
>
> I was going that way, couple years ago, but the Skype client was
> scaling kind of badly for me. No more than a bunch of simultaneus
> calls.
> But at that time I was testing using a standard ALSA sound driver, and
> Skype client for ALSA.
> Maybe using Skype for OSS and skypopen.ko it will scale better.
>
> Good hint, at least I will check into it in the future.
>
> Other ideas?
>
> -giovanni
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>



-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From Nabble at slickdeals.endjunk.com  Sat Apr 30 21:13:08 2011
From: Nabble at slickdeals.endjunk.com (mazilo)
Date: Sat, 30 Apr 2011 10:13:08 -0700 (PDT)
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTinYd64kyirHuOKV64Hc_SVsmO+pzA@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<1304179126964-6319822.post@n2.nabble.com>
	<BANLkTinYd64kyirHuOKV64Hc_SVsmO+pzA@mail.gmail.com>
Message-ID: <1304183588768-6319970.post@n2.nabble.com>


Giovanni Maruzzelli wrote:
> 
> On Sat, Apr 30, 2011 at 5:58 PM, mazilo
> &lt;Nabble at slickdeals.endjunk.com&gt; wrote:
>> Argh ..., a Skype client is still needed. I thought you had hacked Skype
>> and
>> came up with a mod_skypopen that already includes Skype protocol.
> 
> Hahaha, poor Mazilo, sorry to have you disilluded!
Oh well, until I can find an ARM embedded Skype client that will run on a
Seagate DockStar powered by OpenWRT, this is not going to happen in my
little DockStar world.
-----FreeSWITCH hosted on a Seagate DockStar with OpenWRT.--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/NEW-skypopen-installer-and-easy-Skype-client-download-Skype-calls-on-FreeSWITCH-in-one-minute-tp6319154p6319970.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From chat2jesse at gmail.com  Sat Apr 30 21:45:49 2011
From: chat2jesse at gmail.com (jesse)
Date: Sat, 30 Apr 2011 10:45:49 -0700
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <1304183588768-6319970.post@n2.nabble.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<1304179126964-6319822.post@n2.nabble.com>
	<BANLkTinYd64kyirHuOKV64Hc_SVsmO+pzA@mail.gmail.com>
	<1304183588768-6319970.post@n2.nabble.com>
Message-ID: <BANLkTinbWcMX2jP5TdL0FOp_khCrNJpwcA@mail.gmail.com>

You don't like debian on dockstar?
On Apr 30, 2011 10:13 AM, "mazilo" <Nabble at slickdeals.endjunk.com> wrote:
>
> Giovanni Maruzzelli wrote:
>>
>> On Sat, Apr 30, 2011 at 5:58 PM, mazilo
>> &lt;Nabble at slickdeals.endjunk.com&gt; wrote:
>>> Argh ..., a Skype client is still needed. I thought you had hacked Skype
>>> and
>>> came up with a mod_skypopen that already includes Skype protocol.
>>
>> Hahaha, poor Mazilo, sorry to have you disilluded!
> Oh well, until I can find an ARM embedded Skype client that will run on a
> Seagate DockStar powered by OpenWRT, this is not going to happen in my
> little DockStar world.
> -----FreeSWITCH hosted on a Seagate DockStar with OpenWRT.--
> View this message in context:
http://freeswitch-users.2379917.n2.nabble.com/NEW-skypopen-installer-and-easy-Skype-client-download-Skype-calls-on-FreeSWITCH-in-one-minute-tp6319154p6319970.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 gmaruzz at celliax.org  Sat Apr 30 21:50:24 2011
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sat, 30 Apr 2011 19:50:24 +0200
Subject: [Freeswitch-users] NEW skypopen installer and easy Skype client
 download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTinbWcMX2jP5TdL0FOp_khCrNJpwcA@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
	<1304179126964-6319822.post@n2.nabble.com>
	<BANLkTinYd64kyirHuOKV64Hc_SVsmO+pzA@mail.gmail.com>
	<1304183588768-6319970.post@n2.nabble.com>
	<BANLkTinbWcMX2jP5TdL0FOp_khCrNJpwcA@mail.gmail.com>
Message-ID: <BANLkTinKmrDQGmXMyoK5MVn8Ttq14zJPww@mail.gmail.com>

On Sat, Apr 30, 2011 at 7:45 PM, jesse <chat2jesse at gmail.com> wrote:
> You don't like debian on dockstar?

mazilo was disilluded because for mod_skypopen you still need a Skype
client, and Skype client does not exists for Debian on Dockstar. So,
he cannot use mod_skypopen.

-giovanni


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From garmt.noname at gmail.com  Sat Apr 30 22:21:30 2011
From: garmt.noname at gmail.com (Grmt)
Date: Sat, 30 Apr 2011 20:21:30 +0200
Subject: [Freeswitch-users] FS - SIP profiles crashed
In-Reply-To: <BANLkTikfsdWxJR5NnzqxFj6JfajcDxRF-A@mail.gmail.com>
References: <BANLkTi=402RnR54vQmVx7_fh8z1iaX3aFw@mail.gmail.com>
	<1304090527690-6317287.post@n2.nabble.com>
	<BANLkTinqHZb1A0Fqz1z+BK+UeM2vUfWG9A@mail.gmail.com>
	<4dbb2f69.81bf0e0a.4bad.2812@mx.google.com>
	<BANLkTimYon6k0qvZtUnQWY0MVMS6AYEhmw@mail.gmail.com>
	<BANLkTikXtcGECS_tZPMcS+4Kq_mvASgnXA@mail.gmail.com>
	<BANLkTinhWi3Q7x4B3hM60w+1F96htTVpvA@mail.gmail.com>
	<BANLkTim1Bp=t-XMQJmi9ABrvfY=fr+CnGg@mail.gmail.com>
	<BANLkTikfsdWxJR5NnzqxFj6JfajcDxRF-A@mail.gmail.com>
Message-ID: <4dbc5332.c9860e0a.668b.ffffa003@mx.google.com>

I assume you mean GIT april 4th, 2011 ?  I'm almost certain that I also
experienced this problem after apr 21st, 2010.

And indeed you have to restart/scan sofia sip profiles.

 

I suspect it is related to 

-          Having a multihomed system (multiple network cards, i.e. wifi +
fixed Ethernet + virtual adaptors)

o   Possibly related to running virtual machines (with virtual adaptors) on
the same host (e.g. any vmware installation)

-          Using ipv4 and ipv6 simultaneously

 

On the other hand, I have not seen this problem recently.  I changed my
vmware player network card installation and also disabled ipv6 and updated
the firmware of my router,  while still using DHCP (and NAT). Changes to the
FS source code in the mean time may also have fixed the issue.  As this
problem occurred on my dev system, I did not bother going to the bottom of
it . In a production system I would recommend using a fixed IP address and
the settings as noted in the wiki.

 

 

Garmt

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
fieldpeak
Sent: Saturday, 30 April, 2011 07:39
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FS - SIP profiles crashed

 

sorry, my misunderstanding, it looks this issue fixed by FS-933 on
21/Apr/10, however, my GIT is April 4, 2001, it still exist... strange...


2011/4/30 fieldpeak <fieldpeak at gmail.com>

it looks the same issue as
http://jira.freeswitch.org/browse/FS-933?page=com.atlassian.jira.plugin.syst
em.issuetabpanels%3Aall-tabpanel

however, where to add the new parameter: PFLAG_SKIP_RESTART


?

2011/4/30 fieldpeak <fieldpeak at gmail.com>

oh, looks i found the root cause, 

after i unplug the network cable and wait around 20 seconds, FS show ' IP
address changed to '0.0.0.0', then try to load profiles, however, due to
wrong IP address, load failure.
and then i re-plugin the cable, FS detected the IP change back, and
reloadxml, however, did not restart the profiles, so caused profiles
crashed... attached is the log for throughout the procedure...

the resolution is set 

in the sofia.conf.xml file: 

<param name="auto-restart" value="false"/>

welcome to any comment,  

thanks.

Regards,
Charles

 

2011/4/30 fieldpeak <fieldpeak at gmail.com>

really want to know what caused that the IP address changed to '0.0.0.0',
very odd...
i tried even unplug and plugin the network cable, it will not change the
IP...let alone 0.0.0.0...

 

2011/4/30 fieldpeak <fieldpeak at gmail.com>

Thanks all for information, it definitely help a lot. i will try...

meanwhile, i found this link
http://lists.freeswitch.org/pipermail/freeswitch-users/2008-May/003086.html

it looks not so special resolve this issue...

i would like to know what is use for "bind_server_ip=auto" and where and
when FS get the value of $${local_ip_v4}? and if it will changed when
physical IP changed?

Thanks!

Regards,
Charles

2011/4/30 Grmt <garmt.noname at gmail.com>

http://wiki.freeswitch.org/wiki/Sofia#Forcing_SIP_profile_to_use_a_static_IP
_address

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Friday, 29 April, 2011 19:35
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FS - SIP profiles crashed

 

 

On Fri, Apr 29, 2011 at 8:22 AM, Jeff Lenk <jeff at jefflenk.com> wrote:

If your box is configured with a static IP do this.

conf/autoload_configs/sofia.conf.xml --

 

Jeff, was there more to this 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

 

 

 

 

 

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From anton.vazir at gmail.com  Sat Apr 30 22:41:15 2011
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 30 Apr 2011 23:41:15 +0500
Subject: [Freeswitch-users] skypopen scalability tests
In-Reply-To: <BANLkTimnJ1wRTmVRJ8sw1ed57HdKUZxDZA@mail.gmail.com>
References: <BANLkTinMytjFaMc4zA0pfmgV6dMqEHu7WQ@mail.gmail.com>
	<BANLkTimnJ1wRTmVRJ8sw1ed57HdKUZxDZA@mail.gmail.com>
Message-ID: <BANLkTi=RqvGKXvDTnhr1acoaTyNgSUsOVw@mail.gmail.com>

Than we just limited in options :) not ideas, and have to register
this bunch of usernames.

I can confirm that regular ALSA driver is quite heavy, on ATOM 330
(dual core) - it did not support 10 simultaneously running clients,
with 2 simultaneous call, sound was itchy. After switching to your
patched DUMMY - situation recovers.
I may support over 5 calls with moderate load. Pentium dualcore cound
keep atleast 2xAtom330 load
Will test your new kernel module also.


2011/4/30 Giovanni Maruzzelli <gmaruzz at gmail.com>:
> Ooops, I was wrong in my previous mail (my memory no more what was
> used to be :) ).
>
> You cannot have simultaneus calls from many instances of Bob to many
> instances of Alice. It just does not works reliably.
>
> I was testing simultaneus calls from one client skypeusername (Bob) to
> a server with many skypeusernames (Alice, Adam, etc).
>
> So, we're back in the situation I described (need to register many
> different skypeusernames) with the added drawback that simultaneus
> calls from one Skype client does not work well.
>
> Any other ideas?
> -giovanni
>
> On Sat, Apr 30, 2011 at 7:01 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
>> On Sat, Apr 30, 2011 at 6:52 PM, Anton VG <anton.vazir at gmail.com> wrote:
>>>> Any other ideas on how to proceed for scalability tests?
>>>>
>>>
>>> Hm, since 2.0.0.72 still allows simultaneous calls, you do not have to
>>> register lots of usernames, just 2 of them for 2 PC's
>>> than make 100 calls from 1 machine to another
>>
>> I was going that way, couple years ago, but the Skype client was
>> scaling kind of badly for me. No more than a bunch of simultaneus
>> calls.
>> But at that time I was testing using a standard ALSA sound driver, and
>> Skype client for ALSA.
>> Maybe using Skype for OSS and skypopen.ko it will scale better.
>>
>> Good hint, at least I will check into it in the future.
>>
>> Other ideas?
>>
>> -giovanni
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From tayeb.meftah at gmail.com  Sat Apr 30 23:01:34 2011
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 30 Apr 2011 21:01:34 +0200
Subject: [Freeswitch-users] [Freeswitch-dev] NEW skypopen installer and
 easy Skype client download, Skype calls on FreeSWITCH in one minute
In-Reply-To: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
References: <BANLkTi=SdCGB_OHGHpDEujmar=PNtAhGUA@mail.gmail.com>
Message-ID: <4DBC5C8E.6080506@gmail.com>

thank you Giovanni for your facilitys
you did a very great Job
On 30/04/2011 10:31, Giovanni Maruzzelli wrote:
> Dear FreeSWITCHers,
>
> after a fair amount of effort, I ended up with a new way to install
> and use mod_skypopen on Linux.
>
> No more looking around the internet for the lost 2.0.0.72 Skype client for ALSA.
>
> First, we can use the readily available Skype client for OSS.
>
> (added benefit: no more need for ALSA driver. I wrote skypopen.ko OSS
> driver, that's very easy to compile and install, and do not need to
> mess with the operating system installation.)
>
> Second, I wrote an installer that automatically do all the tedious
> work for you: download and install the skype client, create the config
> directory for Skype clients, create the config file for mod_skypopen,
> create the script that launches the Skype clients.
>
> I hope those improvements will lower the barriers for Skype calls on FreeSWITCH.
>
> Actually is ludicrously simple now, and after you compile FreeSWITCH,
> mod_skypopen and the skypopen.ko OSS driver it will take like less
> than one minute to have a complete installation of mod_skypopen ready
> to make and receive calls.
>
> All automatic, no more need to fiddle around with the Skype client
> download, configurations, authorization, etc.
>
> Is all well tested, but maybe there are still some bugs, and maybe the
> docs are not clear/easy enough.
>
> Please have a look at the new and improved wiki page and let me know
> what do you think about (and maybe test the procedures).
>
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Linux
> http://wiki.freeswitch.org/wiki/Mod_skypopen_Skype_Endpoint_and_Trunk#Interactive_INSTALLER_and_CONFIGURATOR
>
> You must update to the latest git to have all the goodies.
>
> Thank you all for your support,
>
> -giovanni
>
>    


-- 
Meftah Tayeb
inum: +883510001288000
phone: +13477595883