From msc at freeswitch.org  Thu Aug  1 01:07:17 2013
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 31 Jul 2013 14:07:17 -0700
Subject: [Freeswitch-users] Just curious,
 is there any standard for FS API command output?
In-Reply-To: <CALtno09QGwV6nx_J7a+8V4sWtqoM4YTO9s8LePT-3T3oZZbZaA@mail.gmail.com>
References: <CALtno09QGwV6nx_J7a+8V4sWtqoM4YTO9s8LePT-3T3oZZbZaA@mail.gmail.com>
Message-ID: <CAKzWOxVBeQmkM9QWUcgbL+9mK=PPm7W=cB5Renp5+tmWmKYUzw@mail.gmail.com>

Not sure what the real issue is here. In all programming langs you have
functions that return various formats - you simply learn to use them. In FS
the APIs are easy to use. Many of them support different output formats
which gives you choices. You even have ESL which abstracts away a lot of
the boring and tedious work.

Are you facing a particularly difficult programming challenge? The gang
here loves to solve problems, so ask away.

-MC
On Jul 31, 2013 1:03 PM, "Rafal Gwizdala" <rafal.gwizdala at gmail.com> wrote:

> Guys, I'm communicating with FS through an inbound event socket and almost
> every command has some different standard for output data structure. Some
> responses are CSV, some (like 'sofia status profile internal reg') use a
>  'vertical name:value format , i've seen also different formats I don't
> remember now. What's worse, the event returned after executing a command
> (SOCKET_DATA api/response) doesn't indicate if the command succeeded or
> not. Some commands are nice enough to return +OK for success but not all of
> them.
> Am I missing something here? Are there more programmer-friendly methods of
> getting command results from FS?
> Thanks
> RG
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 rafal.gwizdala at gmail.com  Thu Aug  1 01:50:07 2013
From: rafal.gwizdala at gmail.com (Rafal Gwizdala)
Date: Wed, 31 Jul 2013 23:50:07 +0200
Subject: [Freeswitch-users] Just curious,
 is there any standard for FS API command output?
In-Reply-To: <CAKzWOxVBeQmkM9QWUcgbL+9mK=PPm7W=cB5Renp5+tmWmKYUzw@mail.gmail.com>
References: <CALtno09QGwV6nx_J7a+8V4sWtqoM4YTO9s8LePT-3T3oZZbZaA@mail.gmail.com>
	<CAKzWOxVBeQmkM9QWUcgbL+9mK=PPm7W=cB5Renp5+tmWmKYUzw@mail.gmail.com>
Message-ID: <CALtno0_EztOJOP80A1vBvRif3hh-NU7gwehRWU_J+7jz_XSMmw@mail.gmail.com>

Michael, there's no real issue here, I was just trying to parse the output
of several commands and was a little surprised that there's no common,
automatically-parsable format like json or xml (some commands support xml,
some csv only and there are totally custom formats too). But you're right,
there are much worse data formats than plain text so i won't complain about
it anymore.
R



On Wed, Jul 31, 2013 at 11:07 PM, Michael Collins <msc at freeswitch.org>wrote:

> Not sure what the real issue is here. In all programming langs you have
> functions that return various formats - you simply learn to use them. In FS
> the APIs are easy to use. Many of them support different output formats
> which gives you choices. You even have ESL which abstracts away a lot of
> the boring and tedious work.
>
> Are you facing a particularly difficult programming challenge? The gang
> here loves to solve problems, so ask away.
>
> -MC
> On Jul 31, 2013 1:03 PM, "Rafal Gwizdala" <rafal.gwizdala at gmail.com>
> wrote:
>
>> Guys, I'm communicating with FS through an inbound event socket and
>> almost every command has some different standard for output data structure.
>> Some responses are CSV, some (like 'sofia status profile internal reg') use
>> a  'vertical name:value format , i've seen also different formats I don't
>> remember now. What's worse, the event returned after executing a command
>> (SOCKET_DATA api/response) doesn't indicate if the command succeeded or
>> not. Some commands are nice enough to return +OK for success but not all of
>> them.
>> Am I missing something here? Are there more programmer-friendly methods
>> of getting command results from FS?
>> Thanks
>> RG
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 avi at avimarcus.net  Thu Aug  1 02:00:00 2013
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 31 Jul 2013 22:00:00 +0000
Subject: [Freeswitch-users] Just curious,
 is there any standard for FS API command output?
In-Reply-To: <CALtno0_EztOJOP80A1vBvRif3hh-NU7gwehRWU_J+7jz_XSMmw@mail.gmail.com>
References: <CALtno09QGwV6nx_J7a+8V4sWtqoM4YTO9s8LePT-3T3oZZbZaA@mail.gmail.com>
	<CAKzWOxVBeQmkM9QWUcgbL+9mK=PPm7W=cB5Renp5+tmWmKYUzw@mail.gmail.com>
	<CALtno0_EztOJOP80A1vBvRif3hh-NU7gwehRWU_J+7jz_XSMmw@mail.gmail.com>
Message-ID: <0000014036beae95-04a443c6-1341-4a6d-9385-e54cf74b23f9-000000@email.amazonses.com>

Most commands with lists you can specify "as json" or "as xml" just like
from fs_cli...

-Avi

On Thu, Aug 1, 2013 at 12:50 AM, Rafal Gwizdala <rafal.gwizdala at gmail.com>wrote:

> Michael, there's no real issue here, I was just trying to parse the output
> of several commands and was a little surprised that there's no common,
> automatically-parsable format like json or xml (some commands support xml,
> some csv only and there are totally custom formats too). But you're right,
> there are much worse data formats than plain text so i won't complain about
> it anymore.
>  R
>
>
>
> On Wed, Jul 31, 2013 at 11:07 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Not sure what the real issue is here. In all programming langs you have
>> functions that return various formats - you simply learn to use them. In FS
>> the APIs are easy to use. Many of them support different output formats
>> which gives you choices. You even have ESL which abstracts away a lot of
>> the boring and tedious work.
>>
>> Are you facing a particularly difficult programming challenge? The gang
>> here loves to solve problems, so ask away.
>>
>> -MC
>> On Jul 31, 2013 1:03 PM, "Rafal Gwizdala" <rafal.gwizdala at gmail.com>
>> wrote:
>>
>>> Guys, I'm communicating with FS through an inbound event socket and
>>> almost every command has some different standard for output data structure.
>>> Some responses are CSV, some (like 'sofia status profile internal reg') use
>>> a  'vertical name:value format , i've seen also different formats I don't
>>> remember now. What's worse, the event returned after executing a command
>>> (SOCKET_DATA api/response) doesn't indicate if the command succeeded or
>>> not. Some commands are nice enough to return +OK for success but not all of
>>> them.
>>> Am I missing something here? Are there more programmer-friendly methods
>>> of getting command results from FS?
>>> Thanks
>>> RG
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 karl at xtronics.com  Thu Aug  1 03:54:13 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Wed, 31 Jul 2013 18:54:13 -0500
Subject: [Freeswitch-users] What ports are really necessary?
In-Reply-To: <CE1E2BED.78469%krice@freeswitch.org>
References: <CE1E2BED.78469%krice@freeswitch.org>
Message-ID: <51F9A3A5.4000108@xtronics.com>

On 07/31/2013 02:50 AM, Ken Rice wrote:

>     On a stateful firewall you can also choose to only open the SIP ports. That'll depend on your
>     SIP profile settings. 5060 at least, and perhaps 5080 too.
>
>     The firewall could look at the SDP, mark the RTP ports as related traffic and automatically open
>     them for you too.

This should be possible with the shorewall package - ( still a learning curve, but you will be more 
likely to maintain your sanity than command line scripts.  shorewall helps avoid creating a 
miss-configured firewall).

I am a week or so away from doing this myself - when I do I will write it up.
I think the place to start is here: http://www.shorewall.net/Helpers.html

https://home.regit.org/netfilter-en/secure-use-of-helpers/
,.,.

I probably need to understand this as well:

http://www.shorewall.net/traffic_shaping.htm

http://shorewall.net/manpages/shorewall-tcrules.html

There are really three firewall issues - getting the right ports open - avoiding spoofing attacks - 
and traffic shaping..

I would also recommend a separate box for the firewall - not a virtual machine. ( I assume someone 
someday will find a way to crash the firewall - and I don't want any servers coming down with it. )

--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

History may not repeat itself, but it does rhyme a lot. -Mark Twain
--------------------------------------------------------------------------------


From eidevm5 at gmail.com  Thu Aug  1 06:33:06 2013
From: eidevm5 at gmail.com (Peter)
Date: Thu, 1 Aug 2013 12:33:06 +1000
Subject: [Freeswitch-users] No audio on either of an established call
Message-ID: <CABkS=xpqqwJBd=cHGzz_aj3r8kvuQssP4HcrYCyCj9oP5EWLTg@mail.gmail.com>

I currently have 2 SIP clients (Linphone) successfully calling each other,
but there is no audio on either end.

The set up is as follows:



Linphone1 (1000)  --> Kamailio 1   <-------> Freeswitch   <------>  Kamalio
2  <---- Linphone2 (2000)


Using Freeswitch 1.2.12 on CentOS (installed via RPM)

Freeswitch has two interfaces:

external - 10.1.1.206
internal -  10.10.10.206

Each of the Linphone clients are registered their respective Kamailio
instance and Kamailio is configured to route via the appropriate interface
on Freeswitch.

The SIP negotiation is working as I can call either Linphone client.

I've done a tcpdump on each side of  Freeswitch and can see the RTP traffic
between the Linphone and the appropriate interface on Freeswitch.

I've tried different codec combinations (mostly G711 and iLBC), and
different SIP clients but still get no audio.

Any pointers on how to track down the issue?

Thanks

Peter
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From blefko5361 at gmail.com  Thu Aug  1 06:39:56 2013
From: blefko5361 at gmail.com (Bruce Lefko)
Date: Wed, 31 Jul 2013 21:39:56 -0500
Subject: [Freeswitch-users] how to compress maked v1.2 into tarball and then
	make install
Message-ID: <CAJNDQt4HwKZoXP=mjWB8EP6wi13a3HyQoEnKYhpUqyVot-SWGQ@mail.gmail.com>

I would like to bootstrap, configure, and make v1.2, and then make the
compiled code into a tarball.

I can then use the tarball to install freeswitch on other machines with the
same OS, but much quicker.  This works with master, but on v1.2, it seems
that "make install" after extracting the tarball is recompiling some libs.

How can I avoid this?

Thanks!
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From ms at mstn.com  Thu Aug  1 06:58:23 2013
From: ms at mstn.com (Jose Suero)
Date: Wed, 31 Jul 2013 22:58:23 -0400
Subject: [Freeswitch-users] List registered users
Message-ID: <22a3e51118761cf1956c005d57cdd37c@mstn.com>

I've been searching on how to get a list of registered users, all I 
found was: "sofia status profile external" which says the amount of 
Registrations, but doesn't list the actual users.

How do I do that??

thanks in advance

Jose Suero


From nreis at wavecom.pt  Thu Aug  1 07:19:37 2013
From: nreis at wavecom.pt (Nuno Reis)
Date: Thu, 1 Aug 2013 04:19:37 +0100
Subject: [Freeswitch-users] List registered users
In-Reply-To: <22a3e51118761cf1956c005d57cdd37c@mstn.com>
References: <22a3e51118761cf1956c005d57cdd37c@mstn.com>
Message-ID: <CAE2+g+JpxtjdLrU-_7bO5tDUhnPx=BGe50QCVue9vWc4iG6N1A@mail.gmail.com>

you can use the following:

sofia status profile <profile_name> reg

or the XML version

sofia xmlstatus profile <profile_name> reg

eg.

sofia status profile internal reg

This will show the current registered users on profile internal (default
port 5060)

--

*Nuno Miguel Reis* | *Unified Communication** Systems*
M. +351 913907481 | nreis at wavecom.pt
WAVECOM-Solu??es R?dio, S.A.
Cacia Park | Rua do Progresso, Lote 15
3800-639 AVEIRO | Portugal
T. +351 309 700 225 | F. +351 234 919 191
*GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
www.wavecom.pt** <http://www.wavecom.pt/>*

[image: Description: Description:
WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php>

[image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>




On Thu, Aug 1, 2013 at 3:58 AM, Jose Suero <ms at mstn.com> wrote:

> I've been searching on how to get a list of registered users, all I
> found was: "sofia status profile external" which says the amount of
> Registrations, but doesn't list the actual users.
>
> How do I do that??
>
> thanks in advance
>
> Jose Suero
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nreis at wavecom.pt  Thu Aug  1 07:23:55 2013
From: nreis at wavecom.pt (Nuno Reis)
Date: Thu, 1 Aug 2013 04:23:55 +0100
Subject: [Freeswitch-users] Video switching
In-Reply-To: <1375267219620-7593439.post@n2.nabble.com>
References: <1375267219620-7593439.post@n2.nabble.com>
Message-ID: <CAE2+g++n9B4+gSi8Xu3abkgDvDjUHHuoaSZNS0s9f7av+aa=dw@mail.gmail.com>

Hi!

Your problem is most likely related to the way freeswitch is negotiating
SDP everytime you switch video/audio.

--

*Nuno Miguel Reis* | *Unified Communication** Systems*
M. +351 913907481 | nreis at wavecom.pt
WAVECOM-Solu??es R?dio, S.A.
Cacia Park | Rua do Progresso, Lote 15
3800-639 AVEIRO | Portugal
T. +351 309 700 225 | F. +351 234 919 191
*GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
www.wavecom.pt** <http://www.wavecom.pt/>*

[image: Description: Description:
WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php>

[image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>




On Wed, Jul 31, 2013 at 11:40 AM, mehroz <mehroz.ashraf85 at gmail.com> wrote:

> Hi guys,
>
> I have Linphone SIP client and my UI allows me to switch call from audio to
> video and video to audio.
>
> After call establishment, when i switch to video, it do so ! and when i
> switch back to audio , it works. But after that when i switch to video , it
> doesn't!
>
> Any guess ? what could be wrong here?
> Thanks
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Video-switching-tp7593439.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ms at mstn.com  Thu Aug  1 07:34:13 2013
From: ms at mstn.com (Jose Suero)
Date: Wed, 31 Jul 2013 23:34:13 -0400
Subject: [Freeswitch-users] List registered users
In-Reply-To: <CAE2+g+JpxtjdLrU-_7bO5tDUhnPx=BGe50QCVue9vWc4iG6N1A@mail.gmail.com>
References: <22a3e51118761cf1956c005d57cdd37c@mstn.com>
	<CAE2+g+JpxtjdLrU-_7bO5tDUhnPx=BGe50QCVue9vWc4iG6N1A@mail.gmail.com>
Message-ID: <9dda3fd66a80541b7d94b4db1e3d6145@mstn.com>

Thanks exacly what I was looking for


On 2013-07-31 23:19, Nuno Reis wrote:
> you can use the following:
>
> sofia status profile <profile_name> reg
>
> or the XML version
>
> sofia xmlstatus profile <profile_name> reg
>
> eg.
>
> sofia status profile internal reg
>
> This will show the current registered users on profile internal
> (default port 5060)
>
> --
>
> NUNO MIGUEL REIS | UNIFIED COMMUNICATION SYSTEMS
>  M. +351 913907481 | nreis at wavecom.pt [11]
>  WAVECOM-Solu??es R?dio, S.A.
>  Cacia Park | Rua do Progresso, Lote 15
> 3800-639 AVEIRO | Portugal
>  T. +351 309 700 225 | F. +351 234 919 191
>  |
>
>  [12]
>
>  [13]
>
> On Thu, Aug 1, 2013 at 3:58 AM, Jose Suero <ms at mstn.com [14]> wrote:
>
>> Ive been searching on how to get a list of registered users, all I
>> found was: "sofia status profile external" which says the amount of
>> Registrations, but doesnt list the actual users.
>>
>> How do I do that??
>>
>> thanks in advance
>>
>> Jose Suero
>>
>>
> 
> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org [1]
>> http://www.freeswitchsolutions.com [2]
>>
>> 
>>  [3]
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org [4]
>> http://wiki.freeswitch.org [5]
>> http://www.cluecon.com [6]
>>
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org [7]
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users [8]
>>
> 
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> [9]
>> http://www.freeswitch.org [10]
>
>
>
> Links:
> ------
> [1] mailto:consulting at freeswitch.org
> [2] http://www.freeswitchsolutions.com
> [3] 
> [4] http://www.freeswitch.org
> [5] http://wiki.freeswitch.org
> [6] http://www.cluecon.com
> [7] mailto:FreeSWITCH-users at lists.freeswitch.org
> [8] http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> [9] http://lists.freeswitch.org/mailman/options/freeswitch-users
> [10] http://www.freeswitch.org
> [11] mailto:nreis at wavecom.pt
> [12] http://www.wavecom.pt/pt/wavecom/premios.php
> [13] http://www.wavecom.pt/pt/mail_eventos.php
> [14] mailto:ms at mstn.com



From lloyd.aloysius at sunteltech.ca  Thu Aug  1 07:37:59 2013
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Wed, 31 Jul 2013 23:37:59 -0400
Subject: [Freeswitch-users] List registered users
In-Reply-To: <CAE2+g+JpxtjdLrU-_7bO5tDUhnPx=BGe50QCVue9vWc4iG6N1A@mail.gmail.com>
References: <22a3e51118761cf1956c005d57cdd37c@mstn.com>
	<CAE2+g+JpxtjdLrU-_7bO5tDUhnPx=BGe50QCVue9vWc4iG6N1A@mail.gmail.com>
Message-ID: <CAGvuWYZte0aOF0kA9h7b5nqWYRzRHSE0wUDVYnoVXtGuixMC7g@mail.gmail.com>

show registrations

--

you can see the available show options - > show press the tab key
*  <http://www.sunteltech.ca/blog/>*

Lloyd

On Wed, Jul 31, 2013 at 11:19 PM, Nuno Reis <nreis at wavecom.pt> wrote:

> you can use the following:
>
> sofia status profile <profile_name> reg
>
> or the XML version
>
> sofia xmlstatus profile <profile_name> reg
>
> eg.
>
> sofia status profile internal reg
>
> This will show the current registered users on profile internal (default
> port 5060)
>
> --
>
> *Nuno Miguel Reis* | *Unified Communication** Systems*
> M. +351 913907481 | nreis at wavecom.pt
> WAVECOM-Solu??es R?dio, S.A.
> Cacia Park | Rua do Progresso, Lote 15
> 3800-639 AVEIRO | Portugal
> T. +351 309 700 225 | F. +351 234 919 191
> *GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
> www.wavecom.pt** <http://www.wavecom.pt/>*
>
> [image: Description: Description: WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php>
>
> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>
>
>
>
>
> On Thu, Aug 1, 2013 at 3:58 AM, Jose Suero <ms at mstn.com> wrote:
>
>> I've been searching on how to get a list of registered users, all I
>> found was: "sofia status profile external" which says the amount of
>> Registrations, but doesn't list the actual users.
>>
>> How do I do that??
>>
>> thanks in advance
>>
>> Jose Suero
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ms at mstn.com  Thu Aug  1 07:40:53 2013
From: ms at mstn.com (Jose Suero)
Date: Wed, 31 Jul 2013 23:40:53 -0400
Subject: [Freeswitch-users] Freeswitch behind opensips behind NAT
Message-ID: <bc6c45aea39c3fa37dc6bcb601539254@mstn.com>

I'm trying to setup freeswitch with opensips on EC2 I'm using opensips 
dispatcher to send registrations to freeswitch and Load Balancer to send 
calls to freeswitch this I took from 
http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS

I have opensips ext-sip-ip and ext-rtp-ip on external profile set to my 
external IP, and if I connect the phones directly to freeswitch I can 
hear sound. But when I go thru Opensips I can register the phone to 
freeswitch, but when I try calling the phone is trying to connect to RTP 
on my internal address. I'm using internal address to connect from one 
server to the other

Does anyone know how I can load balancer send the external address is 
getting from Freeswitch and not the internal IP is using to talk to the 
other server

thanks in advance


Jose Suero


From msc at freeswitch.org  Thu Aug  1 07:42:15 2013
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 31 Jul 2013 20:42:15 -0700
Subject: [Freeswitch-users] No audio on either of an established call
In-Reply-To: <CABkS=xpqqwJBd=cHGzz_aj3r8kvuQssP4HcrYCyCj9oP5EWLTg@mail.gmail.com>
References: <CABkS=xpqqwJBd=cHGzz_aj3r8kvuQssP4HcrYCyCj9oP5EWLTg@mail.gmail.com>
Message-ID: <CAKzWOxWxxasZFqtmEFyhUPN4GdSb1w1O5PFgzzeTaHXOwWe_jw@mail.gmail.com>

I'd start by getting a debug log of the call and posting it on
pastebin.freeswitch.org. Hopefully that will yield some clues.
-MC


On Wed, Jul 31, 2013 at 7:33 PM, Peter <eidevm5 at gmail.com> wrote:

> I currently have 2 SIP clients (Linphone) successfully calling each other,
> but there is no audio on either end.
>
> The set up is as follows:
>
>
>
> Linphone1 (1000)  --> Kamailio 1   <-------> Freeswitch   <------>
> Kamalio 2  <---- Linphone2 (2000)
>
>
> Using Freeswitch 1.2.12 on CentOS (installed via RPM)
>
> Freeswitch has two interfaces:
>
> external - 10.1.1.206
> internal -  10.10.10.206
>
> Each of the Linphone clients are registered their respective Kamailio
> instance and Kamailio is configured to route via the appropriate interface
> on Freeswitch.
>
> The SIP negotiation is working as I can call either Linphone client.
>
> I've done a tcpdump on each side of  Freeswitch and can see the RTP
> traffic between the Linphone and the appropriate interface on Freeswitch.
>
> I've tried different codec combinations (mostly G711 and iLBC), and
> different SIP clients but still get no audio.
>
> Any pointers on how to track down the issue?
>
> Thanks
>
> Peter
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From eidevm5 at gmail.com  Thu Aug  1 08:41:02 2013
From: eidevm5 at gmail.com (Peter)
Date: Thu, 1 Aug 2013 14:41:02 +1000
Subject: [Freeswitch-users] No audio on either of an established call
In-Reply-To: <CAKzWOxWxxasZFqtmEFyhUPN4GdSb1w1O5PFgzzeTaHXOwWe_jw@mail.gmail.com>
References: <CABkS=xpqqwJBd=cHGzz_aj3r8kvuQssP4HcrYCyCj9oP5EWLTg@mail.gmail.com>
	<CAKzWOxWxxasZFqtmEFyhUPN4GdSb1w1O5PFgzzeTaHXOwWe_jw@mail.gmail.com>
Message-ID: <CABkS=xrEtZ5BtSpDwQvBDAM=UHmSjSOf4ZFXLuK_KZkED2DS1Q@mail.gmail.com>

Here's the Freeswitch log showing the SIP traffic:

http://pastebin.freeswitch.org/21253

The summary of IP addresses are:

10.1.1.19 - External Kamailio
10.1.1.206 - External FS interface
10.10.10.206 - Internal FS interface
10.10.10.207 - Internal Kamailio

Call was made from 2005 (10.1.254.25) to 1008  (10.10.10.165).

Note that 10.1.254.0/24 is routable to 10.1.1.0/24



Here's the tcpdump sample taken on both interfaces on Freeswitch

http://pastebin.freeswitch.org/21254

Let me know if anymore logs/debugs are required.


On Thu, Aug 1, 2013 at 1:42 PM, Michael Collins <msc at freeswitch.org> wrote:

> I'd start by getting a debug log of the call and posting it on
> pastebin.freeswitch.org. Hopefully that will yield some clues.
> -MC
>
>
> On Wed, Jul 31, 2013 at 7:33 PM, Peter <eidevm5 at gmail.com> wrote:
>
>> I currently have 2 SIP clients (Linphone) successfully calling each
>> other, but there is no audio on either end.
>>
>> The set up is as follows:
>>
>>
>>
>> Linphone1 (1000)  --> Kamailio 1   <-------> Freeswitch   <------>
>> Kamalio 2  <---- Linphone2 (2000)
>>
>>
>> Using Freeswitch 1.2.12 on CentOS (installed via RPM)
>>
>> Freeswitch has two interfaces:
>>
>> external - 10.1.1.206
>> internal -  10.10.10.206
>>
>> Each of the Linphone clients are registered their respective Kamailio
>> instance and Kamailio is configured to route via the appropriate interface
>> on Freeswitch.
>>
>> The SIP negotiation is working as I can call either Linphone client.
>>
>> I've done a tcpdump on each side of  Freeswitch and can see the RTP
>> traffic between the Linphone and the appropriate interface on Freeswitch.
>>
>> I've tried different codec combinations (mostly G711 and iLBC), and
>> different SIP clients but still get no audio.
>>
>> Any pointers on how to track down the issue?
>>
>> Thanks
>>
>> Peter
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mehroz.ashraf85 at gmail.com  Thu Aug  1 09:04:14 2013
From: mehroz.ashraf85 at gmail.com (mehroz)
Date: Wed, 31 Jul 2013 22:04:14 -0700 (PDT)
Subject: [Freeswitch-users] Video switching
In-Reply-To: <CAE2+g++n9B4+gSi8Xu3abkgDvDjUHHuoaSZNS0s9f7av+aa=dw@mail.gmail.com>
References: <1375267219620-7593439.post@n2.nabble.com>
	<CAE2+g++n9B4+gSi8Xu3abkgDvDjUHHuoaSZNS0s9f7av+aa=dw@mail.gmail.com>
Message-ID: <1375333454405-7593468.post@n2.nabble.com>

yes exactly  Nuno Reis, 
I am confused as how can I debug this issue? What and where something is
wrong.
While looking at logs of freeswitch each time when switched, SDPs are same
are I dont even see any error.
Please comment...



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Video-switching-tp7593439p7593468.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From eidevm5 at gmail.com  Thu Aug  1 09:12:55 2013
From: eidevm5 at gmail.com (Peter)
Date: Thu, 1 Aug 2013 15:12:55 +1000
Subject: [Freeswitch-users] No audio on either of an established call
In-Reply-To: <CABkS=xpqqwJBd=cHGzz_aj3r8kvuQssP4HcrYCyCj9oP5EWLTg@mail.gmail.com>
References: <CABkS=xpqqwJBd=cHGzz_aj3r8kvuQssP4HcrYCyCj9oP5EWLTg@mail.gmail.com>
Message-ID: <CABkS=xqFknFok1NWgoB+tBhrDB+GzrQNA-wA+CjnftpmSmuFHA@mail.gmail.com>

After checking everything for the umpteenth time, I discovered I was
missing a route back to 10.1.254.0/24

Once I put that in, bingo, audio!


On Thu, Aug 1, 2013 at 12:33 PM, Peter <eidevm5 at gmail.com> wrote:

> I currently have 2 SIP clients (Linphone) successfully calling each other,
> but there is no audio on either end.
>
> The set up is as follows:
>
>
>
> Linphone1 (1000)  --> Kamailio 1   <-------> Freeswitch   <------>
> Kamalio 2  <---- Linphone2 (2000)
>
>
> Using Freeswitch 1.2.12 on CentOS (installed via RPM)
>
> Freeswitch has two interfaces:
>
> external - 10.1.1.206
> internal -  10.10.10.206
>
> Each of the Linphone clients are registered their respective Kamailio
> instance and Kamailio is configured to route via the appropriate interface
> on Freeswitch.
>
> The SIP negotiation is working as I can call either Linphone client.
>
> I've done a tcpdump on each side of  Freeswitch and can see the RTP
> traffic between the Linphone and the appropriate interface on Freeswitch.
>
> I've tried different codec combinations (mostly G711 and iLBC), and
> different SIP clients but still get no audio.
>
> Any pointers on how to track down the issue?
>
> Thanks
>
> Peter
>
>
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From rafal.gwizdala at gmail.com  Thu Aug  1 09:44:34 2013
From: rafal.gwizdala at gmail.com (Rafal Gwizdala)
Date: Thu, 1 Aug 2013 07:44:34 +0200
Subject: [Freeswitch-users] Just curious,
 is there any standard for FS API command output?
In-Reply-To: <0000014036beae95-04a443c6-1341-4a6d-9385-e54cf74b23f9-000000@email.amazonses.com>
References: <CALtno09QGwV6nx_J7a+8V4sWtqoM4YTO9s8LePT-3T3oZZbZaA@mail.gmail.com>
	<CAKzWOxVBeQmkM9QWUcgbL+9mK=PPm7W=cB5Renp5+tmWmKYUzw@mail.gmail.com>
	<CALtno0_EztOJOP80A1vBvRif3hh-NU7gwehRWU_J+7jz_XSMmw@mail.gmail.com>
	<0000014036beae95-04a443c6-1341-4a6d-9385-e54cf74b23f9-000000@email.amazonses.com>
Message-ID: <CALtno08zWGeUUU2xgqK-hn6pH4gWsSUF3eFM6GK4s_Nk6yzHgQ@mail.gmail.com>

I'll check this. thanks.
R


On Thu, Aug 1, 2013 at 12:00 AM, Avi Marcus <avi at avimarcus.net> wrote:

> Most commands with lists you can specify "as json" or "as xml" just like
> from fs_cli...
>
> -Avi
>
> On Thu, Aug 1, 2013 at 12:50 AM, Rafal Gwizdala <rafal.gwizdala at gmail.com>wrote:
>
>> Michael, there's no real issue here, I was just trying to parse the
>> output of several commands and was a little surprised that there's no
>> common, automatically-parsable format like json or xml (some commands
>> support xml, some csv only and there are totally custom formats too). But
>> you're right, there are much worse data formats than plain text so i won't
>> complain about it anymore.
>>  R
>>
>>
>>
>> On Wed, Jul 31, 2013 at 11:07 PM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Not sure what the real issue is here. In all programming langs you have
>>> functions that return various formats - you simply learn to use them. In FS
>>> the APIs are easy to use. Many of them support different output formats
>>> which gives you choices. You even have ESL which abstracts away a lot of
>>> the boring and tedious work.
>>>
>>> Are you facing a particularly difficult programming challenge? The gang
>>> here loves to solve problems, so ask away.
>>>
>>> -MC
>>> On Jul 31, 2013 1:03 PM, "Rafal Gwizdala" <rafal.gwizdala at gmail.com>
>>> wrote:
>>>
>>>> Guys, I'm communicating with FS through an inbound event socket and
>>>> almost every command has some different standard for output data structure.
>>>> Some responses are CSV, some (like 'sofia status profile internal reg') use
>>>> a  'vertical name:value format , i've seen also different formats I don't
>>>> remember now. What's worse, the event returned after executing a command
>>>> (SOCKET_DATA api/response) doesn't indicate if the command succeeded or
>>>> not. Some commands are nice enough to return +OK for success but not all of
>>>> them.
>>>> Am I missing something here? Are there more programmer-friendly methods
>>>> of getting command results from FS?
>>>> Thanks
>>>> RG
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 gerald.weber at besharp.at  Thu Aug  1 10:37:55 2013
From: gerald.weber at besharp.at (Gerald Weber)
Date: Thu, 1 Aug 2013 06:37:55 +0000
Subject: [Freeswitch-users] List registered users
In-Reply-To: <CAGvuWYZte0aOF0kA9h7b5nqWYRzRHSE0wUDVYnoVXtGuixMC7g@mail.gmail.com>
References: <22a3e51118761cf1956c005d57cdd37c@mstn.com>
	<CAE2+g+JpxtjdLrU-_7bO5tDUhnPx=BGe50QCVue9vWc4iG6N1A@mail.gmail.com>
	<CAGvuWYZte0aOF0kA9h7b5nqWYRzRHSE0wUDVYnoVXtGuixMC7g@mail.gmail.com>
Message-ID: <D1925DECD316894DAADB84076C1B7D827E406F17@exchange.besharp.at>

or list_users

Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Lloyd Aloysius
Gesendet: Donnerstag, 01. August 2013 05:38
An: FreeSWITCH Users Help
Cc: comunica??es unificadas
Betreff: Re: [Freeswitch-users] List registered users

show registrations

--

you can see the available show options - > show press the tab key


Lloyd

On Wed, Jul 31, 2013 at 11:19 PM, Nuno Reis <nreis at wavecom.pt<mailto:nreis at wavecom.pt>> wrote:
you can use the following:

sofia status profile <profile_name> reg

or the XML version

sofia xmlstatus profile <profile_name> reg

eg.

sofia status profile internal reg

This will show the current registered users on profile internal (default port 5060)

--

Nuno Miguel Reis | Unified Communication Systems
M. +351 913907481 | nreis at wavecom.pt<mailto:nreis at wavecom.pt>
WAVECOM-Solu??es R?dio, S.A.
Cacia Park | Rua do Progresso, Lote 15
3800-639 AVEIRO | Portugal
T. +351 309 700 225 | F. +351 234 919 191
GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88> | www.wavecom.pt<http://www.wavecom.pt/>

[Description: Description: WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php>

<http://www.wavecom.pt/pt/mail_eventos.php>

 <http://www.wavecom.pt/pt/mail_eventos.php>
 <http://www.wavecom.pt/pt/mail_eventos.php>
On Thu, Aug 1, 2013 at 3:58 AM, Jose Suero <ms at mstn.com> wrote:<http://www.wavecom.pt/pt/mail_eventos.php>
I've been searching on how to get a list of registered users, all I
found was: "sofia status profile external" which says the amount of
Registrations, but doesn't list the actual users.

How do I do that??

thanks in advance

Jose Suero

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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<http://www.wavecom.pt/pt/mail_eventos.php>
 <http://www.wavecom.pt/pt/mail_eventos.php>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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<http://www.wavecom.pt/pt/mail_eventos.php>
 <http://www.wavecom.pt/pt/mail_eventos.php>
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From mayamatakeshi at gmail.com  Thu Aug  1 11:32:13 2013
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 1 Aug 2013 16:32:13 +0900
Subject: [Freeswitch-users] WebSocket for proxied outgoing calls
Message-ID: <CABaNFCaQBmLDr97c=JKMFQvY3fxrNkoQk7d5UBVxJCLdQteSjw@mail.gmail.com>

Hello,
I have kamailio SIP proxy with WebSocket support in front of FS using plain
UDP transport (not using ws-binding).
It works fine for incoming calls:
http://sipml5.org --> SIP over WebSocket --> kamailio --> SIP over UDP -->
FS plays some prompt.

Then, the WebSocket app registers with kamailio and when a call arrives to
it at FS, FS sends the call to kamailio and the call is sent to the browser.
However, when I try to answer the call, the web app refuses the call with
this:

SIP/2.0 603 Failed to get local SDP.
From: "displayname.user3"<sip:user3 at test1.com>;tag=6197r4g0Kar9H.
To: <sip:user2 at 127.0.0.1>;tag=okDK7cd5n34HgaEM9dgs.
Call-ID: 2d1ab230-7516-1231-969f-5254000fd208.
CSeq: 47333138 INVITE.
Content-Length: 0.

I suppose this is happening because an SDP for a WebRTC call cannot be a
plain one that we send on ordinary calls. Is this correct?
So is it possible to instruct FS (some parameter when calling originate or
bridge) to prepare an SDP for WebRTC?
If not, I was thinking if asking FS to not send SDP in the INVITE would
work (late negotiation) as it would receive the SDP in the "200 OK" and
send its SDP in the ACK. But i could not find a way to force FS to do this
either.

regards,
Takeshi
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From mayamatakeshi at gmail.com  Thu Aug  1 12:01:30 2013
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 1 Aug 2013 17:01:30 +0900
Subject: [Freeswitch-users] WebSocket for proxied outgoing calls
In-Reply-To: <CABaNFCaQBmLDr97c=JKMFQvY3fxrNkoQk7d5UBVxJCLdQteSjw@mail.gmail.com>
References: <CABaNFCaQBmLDr97c=JKMFQvY3fxrNkoQk7d5UBVxJCLdQteSjw@mail.gmail.com>
Message-ID: <CABaNFCZqzCPrn6qkmw-KOq+iVGJbEB1AzyB-U3VCMnmV9UBing@mail.gmail.com>

On Thu, Aug 1, 2013 at 4:32 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> Hello,
> I have kamailio SIP proxy with WebSocket support in front of FS using
> plain UDP transport (not using ws-binding).
> It works fine for incoming calls:
> http://sipml5.org --> SIP over WebSocket --> kamailio --> SIP over UDP
> --> FS plays some prompt.
>
> Then, the WebSocket app registers with kamailio and when a call arrives to
> it at FS, FS sends the call to kamailio and the call is sent to the browser.
> However, when I try to answer the call, the web app refuses the call with
> this:
>
> SIP/2.0 603 Failed to get local SDP.
> From: "displayname.user3"<sip:user3 at test1.com>;tag=6197r4g0Kar9H.
> To: <sip:user2 at 127.0.0.1>;tag=okDK7cd5n34HgaEM9dgs.
> Call-ID: 2d1ab230-7516-1231-969f-5254000fd208.
> CSeq: 47333138 INVITE.
> Content-Length: 0.
>
> I suppose this is happening because an SDP for a WebRTC call cannot be a
> plain one that we send on ordinary calls. Is this correct?
> So is it possible to instruct FS (some parameter when calling originate or
> bridge) to prepare an SDP for WebRTC?
>

I have found the answer on this thread:
http://lists.freeswitch.org/pipermail/freeswitch-users/2013-July/097369.html

Set

*media_webrtc=true*

when originating the call.

Sorry for the noise.
I will document the var at the wiki.


If not, I was thinking if asking FS to not send SDP in the INVITE would
> work (late negotiation) as it would receive the SDP in the "200 OK" and
> send its SDP in the ACK. But i could not find a way to force FS to do this
> either.
>
> regards,
> Takeshi
>
>
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From martyn at magiccow.co.uk  Thu Aug  1 15:20:33 2013
From: martyn at magiccow.co.uk (Martyn Davies)
Date: Thu, 1 Aug 2013 12:20:33 +0100
Subject: [Freeswitch-users] Sending NOTIFY using sendevent / ESL
In-Reply-To: <CA++4AS4s2g9obztJU4bA06FJ=3UfnnLohxTrGaGXuiHsHgDmYQ@mail.gmail.com>
References: <CA++4AS4s2g9obztJU4bA06FJ=3UfnnLohxTrGaGXuiHsHgDmYQ@mail.gmail.com>
Message-ID: <CA++4AS7ePd2nd6s35wNbEQ06t9EZihzQmjkYs31w6ZS_hkJdNg@mail.gmail.com>

Regarding my "sendevent' not working: can anyone suggest useful
diagnostics/traces I could switch on to find out why no NOTIFY results from
my API request?

Regards,
Martyn


On 23 July 2013 20:45, Martyn Davies <martyn at magiccow.co.uk> wrote:

> I've been trying to generate a NOTIFY using sendevent, following some of
> the examples in http://wiki.freeswitch.org/wiki/Mod_event_socket and also
> various stuff found in Jira and historic message trails.
>
> I can't get it to do anything.  If I switch on events for notify, I can
> see that the sendevent command generates some kind of message (and the
> fields seem to have appropriate URIs and so on), but at the SIP level
> nothing at all happens (I've got wireshark at the client end). I've
> switched on all events on fs_cli, and sofia trace but see nothing to
> suggest that a notify is sent.
>
> I'd appreciate any suggestions that could make a message flow. Failing
> that, what logs can I switch on to give a clue about where/why the message
> is discarded before making it to SIP?
>
> I was using FS 1.5.1, just changed to 1.5.3 today (no difference). Client
> software is brand-new Mac Blink client bought last week.
>
> Regards,
> Martyn
>
>
>
>
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From wstephen80 at gmail.com  Thu Aug  1 18:09:44 2013
From: wstephen80 at gmail.com (Stephen Wilde)
Date: Thu, 1 Aug 2013 16:09:44 +0200
Subject: [Freeswitch-users] switch_channel_wait_for_state_timeout
Message-ID: <CAO4LeU3kyO1sNpQkBp4gJJcSVvnS5fDkaNmr5NVqnaqCn9KMrQ@mail.gmail.com>

the call to:

switch_channel_wait_for_state_timeout(channel, CS_REPORTING, 5000);


in "switch_ivr_originate.c"


is correct?


What I mean is that:


CS_REPORTING >= CS_HANGUP


then the "switch_channel_wait_for_state_timeout" doesn't wait for
CS_REPORTING state: I'm wrong?


Stephen
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From randhawaay at gmail.com  Thu Aug  1 12:22:34 2013
From: randhawaay at gmail.com (Shan Randhawa)
Date: Thu, 1 Aug 2013 01:22:34 -0700
Subject: [Freeswitch-users] PROBLEM: Making Concurrent Calls
Message-ID: <CAECKtzK3DJOtOBg05QhcMSwv3rhYRwQe+013Sxb_T0biupD19g@mail.gmail.com>

The problem, i m confronting right now, is that

When I make 2 calls simultaneously from 4 phones,it acts like a Conference
Call means everyone is listening to everyone,
For Example if cell A calls cell B,AND cell C calls cell D,then cell C can
hear the conversation between A and B. and similarly vice verse is
happening.

Here is the extension  that i m using to make calls,
-------------------------------------------
<extension name="local_call">
<condition field="${python(VBTS_DB_Get
name|callerid|${destination_number})}" expression="IMSI\d{15}"/>
<condition field="${python(VBTS_DB_Get callerid|name|${username})}"
expression="\d{7,10}">
<action application="set" data="target=${python(VBTS_DB_Get
name|callerid|${destination_number})}"/>
<action application="set" data="port=${python(VBTS_DB_Get
port|callerid|${destination_number})}"/>
<action application="set" data="host=${python(VBTS_DB_Get
ipaddr|callerid|${destination_number})}"/>
<action application="set"
data="effective_caller_id_number=${python(VBTS_DB_Get
callerid|name|${username})}"/>
<action application="set" data="hangup_after_bridge=true"/><action
application="set" data="continue_on_fail=true"/>
<action application="bridge" data="sofia/internal/${target}@
${host}:${port}"/>
--------------------------------------------

Does any one has any feedback on this why this is happening.

Thanks in Advance
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From osharoiko at gmail.com  Thu Aug  1 03:40:13 2013
From: osharoiko at gmail.com (Oleg Sharoyko)
Date: Thu, 1 Aug 2013 00:40:13 +0100
Subject: [Freeswitch-users] Consecutive bridge actions vs multiple
 targets in a single bridge action.
In-Reply-To: <CAKzWOxV4JE9FjVs86qPjnDUBmNbF9XHg96VxAdQLBVswZuV25Q@mail.gmail.com>
References: <CAOSKQLH=cS11DHqNMwQzj1J0oAhf+zRZx2vCjEfjxr4oUw37+w@mail.gmail.com>
	<CAKzWOxV4JE9FjVs86qPjnDUBmNbF9XHg96VxAdQLBVswZuV25Q@mail.gmail.com>
Message-ID: <CAOSKQLFAavhoy4=BJWneuWK0XvTnX1a-rhv+dNdyU5yW4TV1pg@mail.gmail.com>

Hi Michael,

On 30 July 2013 16:30, Michael Collins <msc at freeswitch.org> wrote:

> Try experimenting with
> https://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail.

Thanks for pointing me to this variable. Unfortunately it don't seem
to help. I've tried setting it to false and to some hangup codes other
than what I receive when user rejects a call but FS still executes
second action. Now searching for continue_on_fail I've found FS-4232
which seems to correspond to my problem and I also found references to
fail_on_single_reject which I haven't checked yet. I will also try to
look in the source code - maybe I'll read something there.

Regards,
Oleg.

> On Tue, Jul 30, 2013 at 7:13 AM, Oleg Sharoyko <osharoiko at gmail.com> wrote:
>>
>> Hi,
>>
>> I was wondering if there exists a simple way of implementing behavior
>> equivalent to
>>
>> <action application="bridge"
>> data="sofia/gateway/gw1/$1|sofia/gateway/gw2/$1"/>
>>
>> with two consecutive bridge actions?
>>
>> I've tried this:
>>
>> <action application="set" data="hangup_after_bridge=true"/>
>> <action application="bridge" data="sofia/gateway/gw1/$1"/>
>> <action application="bridge" data="sofia/gateway/gw2/$1"/>
>>
>> and it results in almost the same behavior except that if first call
>> is rejected by destination (say I was calling a mobile phone and on
>> that phone reject button was pushed) then second action will still be
>> executed. I think with a single bridge action call is considered
>> successful if early media state is reached while with two consecutive
>> actions hangup_after_bridge=true only acts on a fully established
>> call. I couldn't find any options to change this. Will appreciate your
>> help. Thank you!
>>
>> Best regards,
>> --
>> Oleg
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Oleg


From kris at kriskinc.com  Thu Aug  1 18:24:22 2013
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 1 Aug 2013 10:24:22 -0400
Subject: [Freeswitch-users] how to compress maked v1.2 into tarball and
 then make install
In-Reply-To: <CAJNDQt4HwKZoXP=mjWB8EP6wi13a3HyQoEnKYhpUqyVot-SWGQ@mail.gmail.com>
References: <CAJNDQt4HwKZoXP=mjWB8EP6wi13a3HyQoEnKYhpUqyVot-SWGQ@mail.gmail.com>
Message-ID: <CAKDfjgd0Yk8xBDyxj4ubCogqm1WsXzbLdqbpPU8fbXOebiEA=g@mail.gmail.com>

There are certainly better ways to do this but if you really want to
hack it, you can just tar up an installed FS directory and copy it to
another system.  After that a simple "copy the init script and enable
it" should be all you need.

On Wed, Jul 31, 2013 at 10:39 PM, Bruce Lefko <blefko5361 at gmail.com> wrote:
> I would like to bootstrap, configure, and make v1.2, and then make the
> compiled code into a tarball.
>
> I can then use the tarball to install freeswitch on other machines with the
> same OS, but much quicker.  This works with master, but on v1.2, it seems
> that "make install" after extracting the tarball is recompiling some libs.
>
> How can I avoid this?
>
> Thanks!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From mike at jerris.com  Thu Aug  1 18:43:40 2013
From: mike at jerris.com (Michael Jerris)
Date: Thu, 1 Aug 2013 10:43:40 -0400
Subject: [Freeswitch-users] how to compress maked v1.2 into tarball and
	then make install
In-Reply-To: <CAJNDQt4HwKZoXP=mjWB8EP6wi13a3HyQoEnKYhpUqyVot-SWGQ@mail.gmail.com>
References: <CAJNDQt4HwKZoXP=mjWB8EP6wi13a3HyQoEnKYhpUqyVot-SWGQ@mail.gmail.com>
Message-ID: <BF1FDBD8-75BF-4039-BC5D-CC572025F8DD@jerris.com>

What OS? This is what packages are.

Mike

On Jul 31, 2013, at 10:39 PM, Bruce Lefko <blefko5361 at gmail.com> wrote:

> I would like to bootstrap, configure, and make v1.2, and then make the compiled code into a tarball.
> 
> I can then use the tarball to install freeswitch on other machines with the same OS, but much quicker.  This works with master, but on v1.2, it seems that "make install" after extracting the tarball is recompiling some libs.
> 
> How can I avoid this?
> 
> Thanks!
> 


From krice at freeswitch.org  Thu Aug  1 19:08:30 2013
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 01 Aug 2013 10:08:30 -0500
Subject: [Freeswitch-users] FreeSWITCH Utilization in Government and
 Municipalities
In-Reply-To: <0AB0A933E0F9D24EAD7873D7B2241546011ECE42@EOCEXVS01.brevardco.int>
Message-ID: <CE1FE41E.7873F%krice@freeswitch.org>

Hi Jeff,

There are several public institutions that are currently either deploying or
have deployed FreeSWITCH.
A couple that come to mind right away.

Missouri University of Science and Technology is currently in a pilot for
deployment. 
Bryant School District, Bryant, AR has FS in production.
There is a Public Housing Authority in GA (which one I cant remember). I am
sure there are others also.
I have also heard of people using FreeSWITCH in the PSAP for the E911
dispatchers.

As far as Cluecon, Look forward to seeing you there!

K


On 7/31/13 2:27 PM, "McKnight, Jeffrey" <Jeffrey.McKnight at brevardcounty.us>
wrote:

> I am implementing FreeSWITCH in my county in the very near future.
>  
> I have an approved preliminary plan to roll out a couple of hundred VOIP
> phones, but need case studies of FreeSWITCH utilized in government or
> municipal applications in order to proceed on a larger scale. Fire
> departments, law enforcement, emergency operations, and 911 applications would
> also be helpful.  It appears that some feel that government is so different
> than commercial applications, that it warrants further investigation.
>  
> If you are involved in or are aware of specific FreeSWITCH applications in
> these areas, I would appreciate any details that you could provide.
>  
> I will be attending ClueCon next week in Chicago and will continue to solicit
> contacts while there. Hope to see you there.
>  
> I appreciate your time and consideration.
>  
> Regards,
>  
> Jeff
>  
>  
>  
> Jeff McKnight, PMP?
> IT Telecommunications Manager
> Brevard County BoCC
> 321 633-2135 
> Jeffrey.McKnight at brevardcounty.us <mailto:Jeffrey.McKnight at brevardcounty.us>
>  
>  <javascript:ClickThumbnail(37)>
>  
> 
> 
> Under Florida Law, email addresses are Public Records. If you do not want your
> e-mail address released in response to public record requests, do not send
> electronic mail to this entity. Instead, contact this office by phone or in
> writing.  
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From lloyd.aloysius at sunteltech.ca  Thu Aug  1 19:09:34 2013
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Thu, 1 Aug 2013 11:09:34 -0400
Subject: [Freeswitch-users] PROBLEM: Making Concurrent Calls
In-Reply-To: <CAECKtzK3DJOtOBg05QhcMSwv3rhYRwQe+013Sxb_T0biupD19g@mail.gmail.com>
References: <CAECKtzK3DJOtOBg05QhcMSwv3rhYRwQe+013Sxb_T0biupD19g@mail.gmail.com>
Message-ID: <CAGvuWYZgzAak9WmKc-_3_+UOmhTrpzp1SzeKjvsZpztUa+stmg@mail.gmail.com>

may be problem with your bridge data.

<action application="bridge" data="sofia/internal/${target}
@${host}:${port}"/>

why you use port in the bridge? assume Phone A - Register with user A =>
bridge sofia/internal/userA

Lloyd
*
*
*  <http://www.sunteltech.ca/blog/>*


On Thu, Aug 1, 2013 at 4:22 AM, Shan Randhawa <randhawaay at gmail.com> wrote:

> The problem, i m confronting right now, is that
>
> When I make 2 calls simultaneously from 4 phones,it acts like a Conference
> Call means everyone is listening to everyone,
> For Example if cell A calls cell B,AND cell C calls cell D,then cell C can
> hear the conversation between A and B. and similarly vice verse is
> happening.
>
> Here is the extension  that i m using to make calls,
> -------------------------------------------
> <extension name="local_call">
> <condition field="${python(VBTS_DB_Get
> name|callerid|${destination_number})}" expression="IMSI\d{15}"/>
> <condition field="${python(VBTS_DB_Get callerid|name|${username})}"
> expression="\d{7,10}">
> <action application="set" data="target=${python(VBTS_DB_Get
> name|callerid|${destination_number})}"/>
> <action application="set" data="port=${python(VBTS_DB_Get
> port|callerid|${destination_number})}"/>
> <action application="set" data="host=${python(VBTS_DB_Get
> ipaddr|callerid|${destination_number})}"/>
> <action application="set"
> data="effective_caller_id_number=${python(VBTS_DB_Get
> callerid|name|${username})}"/>
> <action application="set" data="hangup_after_bridge=true"/><action
> application="set" data="continue_on_fail=true"/>
> <action application="bridge" data="sofia/internal/${target}@
> ${host}:${port}"/>
> --------------------------------------------
>
> Does any one has any feedback on this why this is happening.
>
> Thanks in Advance
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 khorsmann at gmail.com  Thu Aug  1 19:11:17 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Thu, 1 Aug 2013 17:11:17 +0200
Subject: [Freeswitch-users] Codec Settings and some other WebRTC FreeSWITCH
	v1.4beta questions
Message-ID: <CAFArqsb686M=cfvPZU+ce-UhE7s_O3CFxTwFYV7mU54XT+20JQ@mail.gmail.com>

Hello all,

branch  v1.4.beta
version 8ca6fe728d34cf4d7431b44c4ff6cada639b8a98
Centos 6.4 64bit
OpenSSL 1.0.0-fips

What are the recommended settings of "global_codec_prefs" and
"outbound_codec_prefs" for webrtc use?

Could you show me an example dialplan snippet how to by-pass-media for
webrtc video from browser to browser?

Is it possible to make an condition if video is send or not?

Thanks!

-- 
Cheers

*Karsten Horsmann* aka witchdoc on irc :)
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From Jeffrey.McKnight at brevardcounty.us  Thu Aug  1 19:13:41 2013
From: Jeffrey.McKnight at brevardcounty.us (McKnight, Jeffrey)
Date: Thu, 1 Aug 2013 11:13:41 -0400
Subject: [Freeswitch-users] FreeSWITCH Utilization in Government and
	Municipalities
Message-ID: <0AB0A933E0F9D24EAD7873D7B2241546011ECFBC@EOCEXVS01.brevardco.int>

Thanks Ken!

 

I will follow up with your recommendations.

 

See you next week.

 

Regards,

 

Jeff

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
Rice
Sent: Thursday, August 01, 2013 11:09 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] FreeSWITCH Utilization in Government and
Municipalities

 

Hi Jeff,

There are several public institutions that are currently either
deploying or have deployed FreeSWITCH.
A couple that come to mind right away.

Missouri University of Science and Technology is currently in a pilot
for deployment. 
Bryant School District, Bryant, AR has FS in production.
There is a Public Housing Authority in GA (which one I cant remember). I
am sure there are others also.
I have also heard of people using FreeSWITCH in the PSAP for the E911
dispatchers.

As far as Cluecon, Look forward to seeing you there!

K


On 7/31/13 2:27 PM, "McKnight, Jeffrey" <
Jeffrey.McKnight at brevardcounty.us> wrote:

I am implementing FreeSWITCH in my county in the very near future.
 
I have an approved preliminary plan to roll out a couple of hundred VOIP
phones, but need case studies of FreeSWITCH utilized in government or
municipal applications in order to proceed on a larger scale. Fire
departments, law enforcement, emergency operations, and 911 applications
would also be helpful.  It appears that some feel that government is so
different than commercial applications, that it warrants further
investigation.
 
If you are involved in or are aware of specific FreeSWITCH applications
in these areas, I would appreciate any details that you could provide.
 
I will be attending ClueCon next week in Chicago and will continue to
solicit contacts while there. Hope to see you there.
 
I appreciate your time and consideration.
 
Regards,
 
Jeff
 
 
 
Jeff McKnight, PMP(r)
IT Telecommunications Manager
Brevard County BoCC
321 633-2135 
Jeffrey.McKnight at brevardcounty.us <
mailto:Jeffrey.McKnight at brevardcounty.us> 

 <javascript:ClickThumbnail(37)> 
 

________________________________


Under Florida Law, email addresses are Public Records. If you do not
want your e-mail address released in response to public record requests,
do not send electronic mail to this entity. Instead, contact this office
by phone or in writing.  

________________________________

________________________________________________________________________
_
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch



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

Under Florida Law, email addresses are Public Records. If you do
not want your e-mail address released in response to public record
requests, do not send electronic mail to this entity. Instead,
contact this office by phone or in writing.  
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From lloyd.aloysius at gmail.com  Thu Aug  1 19:14:28 2013
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Thu, 1 Aug 2013 11:14:28 -0400
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
Message-ID: <CAGvuWYbS4+aZJaW18fRw_oV13Wwo81+H5bAoMTwrYDzQV=UFmA@mail.gmail.com>

Hi All

Does any one have experience with FreeSWITCH Core and Cassandra Database.
Sounds like Cassandra (http://cassandra.apache.org/) have a better
replication system. This will help to distribute freeswitch across multiple
data centers and provide a better fail over system.

Any feedback ? Thank you in advance.

Thanks
Lloyd
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From intralanman at freeswitch.org  Thu Aug  1 19:24:47 2013
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 01 Aug 2013 11:24:47 -0400
Subject: [Freeswitch-users] FreeSWITCH Utilization in Government and
	Municipalities
In-Reply-To: <0AB0A933E0F9D24EAD7873D7B2241546011ECE42@EOCEXVS01.brevardco.int>
References: <0AB0A933E0F9D24EAD7873D7B2241546011ECE42@EOCEXVS01.brevardco.int>
Message-ID: <51FA7DBF.7040006@freeswitch.org>

On 07/31/2013 03:27 PM, McKnight, Jeffrey wrote:
>
> I am implementing FreeSWITCH in my county in the very near future.
>
Happy to hear that. Especially as a former Florida resident :-)

> I have an approved preliminary plan to roll out a couple of hundred 
> VOIP phones, but need case studies of FreeSWITCH utilized in 
> government or municipal applications in order to proceed on a larger 
> scale. Fire departments, law enforcement, emergency operations, and 
> 911 applications would also be helpful.  It appears that some feel 
> that government is so different than commercial applications, that it 
> warrants further investigation.
>
>
There are some cases in which that is true.  When you start talking 
about dealing with 911, police, fire, etc then it's worth noting to keep 
in mind those with disabilities that would require TDD (which FreeSWITCH 
has support for) and other accomodations.  Aside from that, there aren't 
too many things that I can think of that are much different than normal 
enterprise installations with thousands of users.

Looking forward to seeing you at ClueCon


-Ray
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From bdfoster at davri.com  Thu Aug  1 19:39:28 2013
From: bdfoster at davri.com (Brian Foster)
Date: Thu, 1 Aug 2013 11:39:28 -0400
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <CAGvuWYbS4+aZJaW18fRw_oV13Wwo81+H5bAoMTwrYDzQV=UFmA@mail.gmail.com>
References: <CAGvuWYbS4+aZJaW18fRw_oV13Wwo81+H5bAoMTwrYDzQV=UFmA@mail.gmail.com>
Message-ID: <CAHgk4BWe0yWsFru86SZGEDExHZ8FE9ZJiVxJ_vZri3woqcma_Q@mail.gmail.com>

This question comes up alot. Cassandra/CouchDB/MongoDB/etc are NoSQL. From
what I've read there's not a lot of evidence supporting that move. RDBMS
are more than capable of replication and tend to be faster. You're probably
beating a dead horse at this point ;)

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 1, 2013 11:24 AM, "Lloyd Aloysius" <lloyd.aloysius at gmail.com> wrote:

> Hi All
>
> Does any one have experience with FreeSWITCH Core and Cassandra Database.
> Sounds like Cassandra (http://cassandra.apache.org/) have a better
> replication system. This will help to distribute freeswitch across multiple
> data centers and provide a better fail over system.
>
> Any feedback ? Thank you in advance.
>
> Thanks
> Lloyd
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at davri.com  Thu Aug  1 19:42:44 2013
From: bdfoster at davri.com (Brian Foster)
Date: Thu, 1 Aug 2013 11:42:44 -0400
Subject: [Freeswitch-users] how to compress maked v1.2 into tarball and
 then make install
In-Reply-To: <BF1FDBD8-75BF-4039-BC5D-CC572025F8DD@jerris.com>
References: <CAJNDQt4HwKZoXP=mjWB8EP6wi13a3HyQoEnKYhpUqyVot-SWGQ@mail.gmail.com>
	<BF1FDBD8-75BF-4039-BC5D-CC572025F8DD@jerris.com>
Message-ID: <CAHgk4BVex-PSBd=e4==J+=HBrw=2fuaL80Hy4TEDVP3u0i+jaA@mail.gmail.com>

Isn't there tarballs on files.freeswitch.org?

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 1, 2013 10:48 AM, "Michael Jerris" <mike at jerris.com> wrote:

> What OS? This is what packages are.
>
> Mike
>
> On Jul 31, 2013, at 10:39 PM, Bruce Lefko <blefko5361 at gmail.com> wrote:
>
> > I would like to bootstrap, configure, and make v1.2, and then make the
> compiled code into a tarball.
> >
> > I can then use the tarball to install freeswitch on other machines with
> the same OS, but much quicker.  This works with master, but on v1.2, it
> seems that "make install" after extracting the tarball is recompiling some
> libs.
> >
> > How can I avoid this?
> >
> > Thanks!
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jleung at v10networks.ca  Thu Aug  1 20:04:20 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Thu, 1 Aug 2013 09:04:20 -0700
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <CAGvuWYbS4+aZJaW18fRw_oV13Wwo81+H5bAoMTwrYDzQV=UFmA@mail.gmail.com>
References: <CAGvuWYbS4+aZJaW18fRw_oV13Wwo81+H5bAoMTwrYDzQV=UFmA@mail.gmail.com>
Message-ID: <008101ce8ed0$c96def90$5c49ceb0$@v10networks.ca>

I'm not sure of Cassandra database will work as a database for the FreeSWICH
core since the syntax looks SQL but isn't.

 

If It provides ODBC connectivity, you may as well as give it a shot.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lloyd
Aloysius
Sent: Thursday, August 1, 2013 8:14 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database

 

Hi All

 

Does any one have experience with FreeSWITCH Core and Cassandra Database.
Sounds like Cassandra (http://cassandra.apache.org/) have a better
replication system. This will help to distribute freeswitch across multiple
data centers and provide a better fail over system.

 

Any feedback ? Thank you in advance.

 

Thanks

Lloyd

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From robert.hadley at teotech.com  Thu Aug  1 20:15:42 2013
From: robert.hadley at teotech.com (Robert Hadley)
Date: Thu, 1 Aug 2013 16:15:42 +0000
Subject: [Freeswitch-users] Sending NOTIFY using sendevent / ESL
In-Reply-To: <CA++4AS7ePd2nd6s35wNbEQ06t9EZihzQmjkYs31w6ZS_hkJdNg@mail.gmail.com>
References: <CA++4AS4s2g9obztJU4bA06FJ=3UfnnLohxTrGaGXuiHsHgDmYQ@mail.gmail.com>
	<CA++4AS7ePd2nd6s35wNbEQ06t9EZihzQmjkYs31w6ZS_hkJdNg@mail.gmail.com>
Message-ID: <2e4997fca1114f68b19d8b86eed1241b@BN1PR04MB027.namprd04.prod.outlook.com>

Hi Martyn,

A couple suggestions.
1. sendevent is not an api command, do not prefix with api or bgapi
2. sofia debugging: http://wiki.freeswitch.org/wiki/Sofia-SIP#Debugging_Sofia-SIP, re: sofia loglevel all

We send this text to the event socket which sends a check-sync notify to the phone.  The wiki example is missing the to/from-uri lines, please update it.

sendevent NOTIFY
profile: internal
content-type: application/simple-message-summary
event-string: check-sync
user: 1200@<domain>
host: <domain>
from-uri: sip:1200@<domain>
to-uri: sip:1200@<device IP>


Regards,
Robert

From: Martyn Davies [mailto:martyn at magiccow.co.uk]
Sent: Thursday, August 01, 2013 4:21 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Sending NOTIFY using sendevent / ESL

Regarding my "sendevent' not working: can anyone suggest useful diagnostics/traces I could switch on to find out why no NOTIFY results from my API request?

Regards,
Martyn

On 23 July 2013 20:45, Martyn Davies <martyn at magiccow.co.uk<mailto:martyn at magiccow.co.uk>> wrote:
I've been trying to generate a NOTIFY using sendevent, following some of the examples in http://wiki.freeswitch.org/wiki/Mod_event_socket and also various stuff found in Jira and historic message trails.

I can't get it to do anything.  If I switch on events for notify, I can see that the sendevent command generates some kind of message (and the fields seem to have appropriate URIs and so on), but at the SIP level nothing at all happens (I've got wireshark at the client end). I've switched on all events on fs_cli, and sofia trace but see nothing to suggest that a notify is sent.

I'd appreciate any suggestions that could make a message flow. Failing that, what logs can I switch on to give a clue about where/why the message is discarded before making it to SIP?

I was using FS 1.5.1, just changed to 1.5.3 today (no difference). Client software is brand-new Mac Blink client bought last week.

Regards,
Martyn




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From lloyd.aloysius at sunteltech.ca  Thu Aug  1 20:23:08 2013
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Thu, 1 Aug 2013 12:23:08 -0400
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <008101ce8ed0$c96def90$5c49ceb0$@v10networks.ca>
References: <CAGvuWYbS4+aZJaW18fRw_oV13Wwo81+H5bAoMTwrYDzQV=UFmA@mail.gmail.com>
	<008101ce8ed0$c96def90$5c49ceb0$@v10networks.ca>
Message-ID: <CAGvuWYajFde+WmiPZDHkF7rwYKBBY8Uk7QcxAbxGM22JiDR7DQ@mail.gmail.com>

if you use MySQL  replication is painful. It is not faster. Binary logs are
killing the cpu and memory. Lots of interest in this direction. I saw
OpenSIP using Casendra Driver for distribution.

LLoyd
*  <http://www.sunteltech.ca/blog/>*


On Thu, Aug 1, 2013 at 12:04 PM, Jeff Leung <jleung at v10networks.ca> wrote:

> I?m not sure of Cassandra database will work as a database for the
> FreeSWICH core since the syntax looks SQL but isn?t.****
>
> ** **
>
> If It provides ODBC connectivity, you may as well as give it a shot.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Lloyd
> Aloysius
> *Sent:* Thursday, August 1, 2013 8:14 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] FreeSWITCH Core - Cassandra Database****
>
> ** **
>
> Hi All****
>
> ** **
>
> Does any one have experience with FreeSWITCH Core and Cassandra Database.
> Sounds like Cassandra (http://cassandra.apache.org/) have a better
> replication system. This will help to distribute freeswitch across multiple
> data centers and provide a better fail over system.****
>
> ** **
>
> Any feedback ? Thank you in advance.****
>
> ** **
>
> Thanks****
>
> Lloyd****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 rnbrady at gmail.com  Thu Aug  1 20:29:36 2013
From: rnbrady at gmail.com (Richard Brady)
Date: Thu, 1 Aug 2013 17:29:36 +0100
Subject: [Freeswitch-users] originate loopback
Message-ID: <CAN5esQ+kM-Du3hQ1K2efH7cnnNHaKBT2hDjUnA8x=Uohyshb5w@mail.gmail.com>

Guys

Please help with this mind bender from the wiki at
http://wiki.freeswitch.org/wiki/Loopback_endpoint:

Example 4: loopback with A-leg inline dialplan

originate loopback/set:job=1234\,answer\,park/default/inline lua:loop.lua inline

Executes *set* app, then *answer* app, then *park* app on the A leg then
bridges to the *lua* app on the B leg

Is the para at the end correct? Isn't it the A-leg that executes the lua
and the B-leg that executes set,answer,park?

Richard
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From ashwinrkjain at gmail.com  Thu Aug  1 20:40:44 2013
From: ashwinrkjain at gmail.com (Ashwin Jain)
Date: Thu, 1 Aug 2013 22:10:44 +0530
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <CAGvuWYajFde+WmiPZDHkF7rwYKBBY8Uk7QcxAbxGM22JiDR7DQ@mail.gmail.com>
References: <CAGvuWYbS4+aZJaW18fRw_oV13Wwo81+H5bAoMTwrYDzQV=UFmA@mail.gmail.com>
	<008101ce8ed0$c96def90$5c49ceb0$@v10networks.ca>
	<CAGvuWYajFde+WmiPZDHkF7rwYKBBY8Uk7QcxAbxGM22JiDR7DQ@mail.gmail.com>
Message-ID: <CAM9SNVSNHV-E93XGro=z7tpGAksnFLsqXiv8Bi-OJB+jSv6otA@mail.gmail.com>

Hi,

I am also looking for similar kind of solution.

I have a web application for which I am using backend as Cassandra.
My web application also controls Freeswitch (as in it contains the UI for
setting up bridging for calls). So, right now, for that I wrote a sync
service which fetches data from Cassandra and puts it in MySQL.



On Thu, Aug 1, 2013 at 9:53 PM, Lloyd Aloysius <lloyd.aloysius at sunteltech.ca
> wrote:

>
> if you use MySQL  replication is painful. It is not faster. Binary logs
> are killing the cpu and memory. Lots of interest in this direction. I saw
> OpenSIP using Casendra Driver for distribution.
>
> LLoyd
> *  <http://www.sunteltech.ca/blog/>*
>
>
> On Thu, Aug 1, 2013 at 12:04 PM, Jeff Leung <jleung at v10networks.ca> wrote:
>
>> I?m not sure of Cassandra database will work as a database for the
>> FreeSWICH core since the syntax looks SQL but isn?t.****
>>
>> ** **
>>
>> If It provides ODBC connectivity, you may as well as give it a shot.****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Lloyd
>> Aloysius
>> *Sent:* Thursday, August 1, 2013 8:14 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] FreeSWITCH Core - Cassandra Database****
>>
>> ** **
>>
>> Hi All****
>>
>> ** **
>>
>> Does any one have experience with FreeSWITCH Core and Cassandra Database.
>> Sounds like Cassandra (http://cassandra.apache.org/) have a better
>> replication system. This will help to distribute freeswitch across multiple
>> data centers and provide a better fail over system.****
>>
>> ** **
>>
>> Any feedback ? Thank you in advance.****
>>
>> ** **
>>
>> Thanks****
>>
>> Lloyd****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Thanks and Regards,
Ashwin Jain
Director (Engineering)
MetroGuild
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From martyn at magiccow.co.uk  Thu Aug  1 20:52:27 2013
From: martyn at magiccow.co.uk (Martyn Davies)
Date: Thu, 1 Aug 2013 17:52:27 +0100
Subject: [Freeswitch-users] Sending NOTIFY using sendevent / ESL
In-Reply-To: <2e4997fca1114f68b19d8b86eed1241b@BN1PR04MB027.namprd04.prod.outlook.com>
References: <CA++4AS4s2g9obztJU4bA06FJ=3UfnnLohxTrGaGXuiHsHgDmYQ@mail.gmail.com>
	<CA++4AS7ePd2nd6s35wNbEQ06t9EZihzQmjkYs31w6ZS_hkJdNg@mail.gmail.com>
	<2e4997fca1114f68b19d8b86eed1241b@BN1PR04MB027.namprd04.prod.outlook.com>
Message-ID: <CA++4AS51f0_mjHBPqPe_d+a4j7zJxMq6KSp=kUzbDWt8j29+Og@mail.gmail.com>

Thanks Robert, I've got the NOTIFY coming out now!  Those from-uri and
to-uri fields are both really needed.


On 1 August 2013 17:15, Robert Hadley <robert.hadley at teotech.com> wrote:

>  Hi Martyn,****
>
> ** **
>
> A couple suggestions.****
>
> 1. sendevent is not an api command, do not prefix with api or bgapi****
>
> 2. sofia debugging:
> http://wiki.freeswitch.org/wiki/Sofia-SIP#Debugging_Sofia-SIP, re: sofia
> loglevel all****
>
> ** **
>
> We send this text to the event socket which sends a check-sync notify to
> the phone.  The wiki example is missing the to/from-uri lines, please
> update it.****
>
> ** **
>
> sendevent NOTIFY****
>
> profile: internal****
>
> content-type: application/simple-message-summary****
>
> event-string: check-sync****
>
> user: 1200@<domain>****
>
> host: <domain>****
>
> from-uri: sip:1200@<domain>****
>
> to-uri: sip:1200@<device IP>****
>
> ** **
>
> ** **
>
> Regards,****
>
> Robert****
>
> ** **
>
> *From:* Martyn Davies [mailto:martyn at magiccow.co.uk]
> *Sent:* Thursday, August 01, 2013 4:21 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Sending NOTIFY using sendevent / ESL****
>
> ** **
>
> Regarding my "sendevent' not working: can anyone suggest useful
> diagnostics/traces I could switch on to find out why no NOTIFY results from
> my API request?****
>
> ** **
>
> Regards,****
>
> Martyn****
>
> ** **
>
> On 23 July 2013 20:45, Martyn Davies <martyn at magiccow.co.uk> wrote:****
>
> I've been trying to generate a NOTIFY using sendevent, following some of
> the examples in http://wiki.freeswitch.org/wiki/Mod_event_socket and also
> various stuff found in Jira and historic message trails. ****
>
> ** **
>
> I can't get it to do anything.  If I switch on events for notify, I can
> see that the sendevent command generates some kind of message (and the
> fields seem to have appropriate URIs and so on), but at the SIP level
> nothing at all happens (I've got wireshark at the client end). I've
> switched on all events on fs_cli, and sofia trace but see nothing to
> suggest that a notify is sent.****
>
> ** **
>
> I'd appreciate any suggestions that could make a message flow. Failing
> that, what logs can I switch on to give a clue about where/why the message
> is discarded before making it to SIP?****
>
> ** **
>
> I was using FS 1.5.1, just changed to 1.5.3 today (no difference). Client
> software is brand-new Mac Blink client bought last week.****
>
> ** **
>
> Regards,****
>
> Martyn****
>
> ** **
>
> ** **
>
> ** **
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 Aug  1 20:52:50 2013
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 1 Aug 2013 11:52:50 -0500
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <CAM9SNVSNHV-E93XGro=z7tpGAksnFLsqXiv8Bi-OJB+jSv6otA@mail.gmail.com>
References: <CAGvuWYbS4+aZJaW18fRw_oV13Wwo81+H5bAoMTwrYDzQV=UFmA@mail.gmail.com>
	<008101ce8ed0$c96def90$5c49ceb0$@v10networks.ca>
	<CAGvuWYajFde+WmiPZDHkF7rwYKBBY8Uk7QcxAbxGM22JiDR7DQ@mail.gmail.com>
	<CAM9SNVSNHV-E93XGro=z7tpGAksnFLsqXiv8Bi-OJB+jSv6otA@mail.gmail.com>
Message-ID: <9ACC40C7-180E-442D-A86A-2D6ADE09EA0A@freeswitch.org>

the core of freeswtich requires a sql database. changing it to use a nosql database is not a trivial task. 

Postgres is probably the most recommended database to use and is used in many high volume production environments. (its that the developers of freeswitch primarily use)
Ken
Sent from my iPad

On Aug 1, 2013, at 11:40, Ashwin Jain <ashwinrkjain at gmail.com> wrote:

> Hi,
> 
> I am also looking for similar kind of solution.
> 
> I have a web application for which I am using backend as Cassandra. 
> My web application also controls Freeswitch (as in it contains the UI for setting up bridging for calls). So, right now, for that I wrote a sync service which fetches data from Cassandra and puts it in MySQL.
> 
> 
> 
> On Thu, Aug 1, 2013 at 9:53 PM, Lloyd Aloysius <lloyd.aloysius at sunteltech.ca> wrote:
>> 
>> if you use MySQL  replication is painful. It is not faster. Binary logs are killing the cpu and memory. Lots of interest in this direction. I saw OpenSIP using Casendra Driver for distribution.
>> 
>> LLoyd
>>  
>> 
>> 
>> On Thu, Aug 1, 2013 at 12:04 PM, Jeff Leung <jleung at v10networks.ca> wrote:
>>> I?m not sure of Cassandra database will work as a database for the FreeSWICH core since the syntax looks SQL but isn?t.
>>> 
>>>  
>>> 
>>> If It provides ODBC connectivity, you may as well as give it a shot.
>>> 
>>>  
>>> 
>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lloyd Aloysius
>>> Sent: Thursday, August 1, 2013 8:14 AM
>>> To: FreeSWITCH Users Help
>>> Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
>>> 
>>>  
>>> 
>>> Hi All
>>> 
>>>  
>>> 
>>> Does any one have experience with FreeSWITCH Core and Cassandra Database. Sounds like Cassandra (http://cassandra.apache.org/) have a better replication system. This will help to distribute freeswitch across multiple data centers and provide a better fail over system.
>>> 
>>>  
>>> 
>>> Any feedback ? Thank you in advance.
>>> 
>>>  
>>> 
>>> Thanks
>>> 
>>> Lloyd
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> 
> 
> -- 
> Thanks and Regards,
> Ashwin Jain
> Director (Engineering)
> MetroGuild
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lloyd.aloysius at sunteltech.ca  Thu Aug  1 21:10:51 2013
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Thu, 1 Aug 2013 13:10:51 -0400
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <9ACC40C7-180E-442D-A86A-2D6ADE09EA0A@freeswitch.org>
References: <CAGvuWYbS4+aZJaW18fRw_oV13Wwo81+H5bAoMTwrYDzQV=UFmA@mail.gmail.com>
	<008101ce8ed0$c96def90$5c49ceb0$@v10networks.ca>
	<CAGvuWYajFde+WmiPZDHkF7rwYKBBY8Uk7QcxAbxGM22JiDR7DQ@mail.gmail.com>
	<CAM9SNVSNHV-E93XGro=z7tpGAksnFLsqXiv8Bi-OJB+jSv6otA@mail.gmail.com>
	<9ACC40C7-180E-442D-A86A-2D6ADE09EA0A@freeswitch.org>
Message-ID: <CAGvuWYanBamOztJ=FmKWZqLDPvD5v8c2_BAJz40aacQ0C-RavA@mail.gmail.com>

Ken,

Thank you for the information. Can you please tell me how can we scale with
postgres with multiple Data centers. I think postgres - replication is not
possible?


Thanks
Lloyd
*
*
*  <http://www.sunteltech.ca/blog/>*


On Thu, Aug 1, 2013 at 12:52 PM, Ken Rice <krice at freeswitch.org> wrote:

> the core of freeswtich requires a sql database. changing it to use a nosql
> database is not a trivial task.
>
> Postgres is probably the most recommended database to use and is used in
> many high volume production environments. (its that the developers of
> freeswitch primarily use)
> Ken
> Sent from my iPad
>
> On Aug 1, 2013, at 11:40, Ashwin Jain <ashwinrkjain at gmail.com> wrote:
>
> Hi,
>
> I am also looking for similar kind of solution.
>
> I have a web application for which I am using backend as Cassandra.
> My web application also controls Freeswitch (as in it contains the UI for
> setting up bridging for calls). So, right now, for that I wrote a sync
> service which fetches data from Cassandra and puts it in MySQL.
>
>
>
> On Thu, Aug 1, 2013 at 9:53 PM, Lloyd Aloysius <
> lloyd.aloysius at sunteltech.ca> wrote:
>
>>
>> if you use MySQL  replication is painful. It is not faster. Binary logs
>> are killing the cpu and memory. Lots of interest in this direction. I saw
>> OpenSIP using Casendra Driver for distribution.
>>
>> LLoyd
>> *  <http://www.sunteltech.ca/blog/>*
>>
>>
>> On Thu, Aug 1, 2013 at 12:04 PM, Jeff Leung <jleung at v10networks.ca>wrote:
>>
>>> I?m not sure of Cassandra database will work as a database for the
>>> FreeSWICH core since the syntax looks SQL but isn?t.****
>>>
>>> ** **
>>>
>>> If It provides ODBC connectivity, you may as well as give it a shot.****
>>>
>>> ** **
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Lloyd
>>> Aloysius
>>> *Sent:* Thursday, August 1, 2013 8:14 AM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* [Freeswitch-users] FreeSWITCH Core - Cassandra Database****
>>>
>>> ** **
>>>
>>> Hi All****
>>>
>>> ** **
>>>
>>> Does any one have experience with FreeSWITCH Core and Cassandra
>>> Database. Sounds like Cassandra (http://cassandra.apache.org/) have a
>>> better replication system. This will help to distribute freeswitch across
>>> multiple data centers and provide a better fail over system.****
>>>
>>> ** **
>>>
>>> Any feedback ? Thank you in advance.****
>>>
>>> ** **
>>>
>>> Thanks****
>>>
>>> Lloyd****
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Thanks and Regards,
> Ashwin Jain
> Director (Engineering)
> MetroGuild
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 Aug  1 21:38:41 2013
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 01 Aug 2013 12:38:41 -0500
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <CAGvuWYanBamOztJ=FmKWZqLDPvD5v8c2_BAJz40aacQ0C-RavA@mail.gmail.com>
Message-ID: <CE200751.7876B%krice@freeswitch.org>

Postgresql offers streaming replication in PG9 versions... Its a
master/(warm)standby system but it works quite well (I use the native pgsql
replication for high volume calling). PostgreSQL is BSD licensed so this has
also lead to several addons that use various methods for replication... Some
are opensourced others are not...

On 8/1/13 12:10 PM, "Lloyd Aloysius" <lloyd.aloysius at sunteltech.ca> wrote:

> Ken,
> 
> Thank you for the information. Can you please tell me how can we scale with
> postgres with multiple Data centers. I think postgres -?replication?is not
> possible?
> 
> 
> Thanks
> Lloyd
> 
> ? <http://www.sunteltech.ca/blog/>
> 
> 
> On Thu, Aug 1, 2013 at 12:52 PM, Ken Rice <krice at freeswitch.org> wrote:
>> the core of freeswtich requires a sql database. changing it to use a nosql
>> database is not a trivial task.?
>> 
>> Postgres is probably the most recommended database to use and is used in many
>> high volume production environments. (its that the developers of freeswitch
>> primarily use)
>> Ken
>> Sent from my iPad
>> 
>> On Aug 1, 2013, at 11:40, Ashwin Jain <ashwinrkjain at gmail.com> wrote:
>> 
>>> Hi,
>>> 
>>> I am also looking for similar kind of solution.
>>> 
>>> I have a web application for which I am using backend as Cassandra.?
>>> My web application also controls Freeswitch (as in it contains the UI for
>>> setting up bridging for calls). So, right now, for that I wrote a sync
>>> service which fetches data from Cassandra and puts it in MySQL.
>>> 
>>> 
>>> 
>>> On Thu, Aug 1, 2013 at 9:53 PM, Lloyd Aloysius
>>> <lloyd.aloysius at sunteltech.ca> wrote:
>>>> 
>>>> if you use MySQL ?replication is?painful. It is not faster. Binary logs are
>>>> killing the cpu and memory. Lots of interest in this direction. I saw
>>>> OpenSIP using Casendra Driver for distribution.
>>>> 
>>>> LLoyd
>>>> ? <http://www.sunteltech.ca/blog/>
>>>> 
>>>> 
>>>> On Thu, Aug 1, 2013 at 12:04 PM, Jeff Leung <jleung at v10networks.ca> wrote:
>>>>> I?m not sure of Cassandra database will work as a database for the
>>>>> FreeSWICH core since the syntax looks SQL but isn?t.
>>>>> ?
>>>>> If It provides ODBC connectivity, you may as well as give it a shot.
>>>>> ?
>>>>> 
>>>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lloyd
>>>>> Aloysius
>>>>> Sent: Thursday, August 1, 2013 8:14 AM
>>>>> To: FreeSWITCH Users Help
>>>>> Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
>>>>> 
>>>>> ?
>>>>> 
>>>>> Hi All
>>>>> 
>>>>> ?
>>>>> 
>>>>> Does any one have experience with FreeSWITCH Core and Cassandra Database.
>>>>> Sounds like Cassandra (http://cassandra.apache.org/) have a better
>>>>> replication system. This will help to distribute freeswitch across
>>>>> multiple data centers and provide a better fail over system.
>>>>> 
>>>>> ?
>>>>> 
>>>>> Any feedback ? Thank you in advance.
>>>>> 
>>>>> ?
>>>>> 
>>>>> Thanks
>>>>> 
>>>>> Lloyd
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>> 
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>> 
>>> 
>>> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From yudha2008 at gmail.com  Thu Aug  1 22:29:11 2013
From: yudha2008 at gmail.com (baskar)
Date: Thu, 1 Aug 2013 11:29:11 -0700 (PDT)
Subject: [Freeswitch-users] DTMF issues
Message-ID: <1375381751492-7593499.post@n2.nabble.com>

Hi All,

Our current existing set up has three different SIP service providers and
all three were working fine, but all of a sudden on one of our SIP service
provider we started to have DTMF issues. We followed up with the problematic
service provider and they kept saying that there is no issue at their end,
now we have connected the non working SIP link to our test server and we are
not having any DTMF issues while using it on the test server. I have
attached log on pastebin.

Freeswitch Live server log
http://pastebin.com/mQ27rDts

Freeswitch test server log
http://pastebin.com/4BEeUcSu

Please some one help me to resolve this issues.

Thanks,
N.Baskar




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/DTMF-issues-tp7593499.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From msc at freeswitch.org  Thu Aug  1 22:33:16 2013
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Aug 2013 11:33:16 -0700
Subject: [Freeswitch-users] originate loopback
In-Reply-To: <CAN5esQ+kM-Du3hQ1K2efH7cnnNHaKBT2hDjUnA8x=Uohyshb5w@mail.gmail.com>
References: <CAN5esQ+kM-Du3hQ1K2efH7cnnNHaKBT2hDjUnA8x=Uohyshb5w@mail.gmail.com>
Message-ID: <CAKzWOxVH8F08OJEEF=kTi0TmGUBSAP=AFc_4uRO6HSqU7tU3Og@mail.gmail.com>

On Thu, Aug 1, 2013 at 9:29 AM, Richard Brady <rnbrady at gmail.com> wrote:

> Guys
>
> Please help with this mind bender from the wiki at
> http://wiki.freeswitch.org/wiki/Loopback_endpoint:
>
> Example 4: loopback with A-leg inline dialplan
>
> originate loopback/set:job=1234\,answer\,park/default/inline lua:loop.lua inline
>
> Executes *set* app, then *answer* app, then *park* app on the A leg then
> bridges to the *lua* app on the B leg
>
> Is the para at the end correct? Isn't it the A-leg that executes the lua
> and the B-leg that executes set,answer,park?
>

I'm pretty sure that the description is correct. It's equivalent to this:

originate loopback/set:job=1234\,answer\,park/default/inline &lua(loop.lua)


The Lua stuff is on the B leg and the set, answer, park stuff is on the
loopback/A leg.
-MC


> Richard
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From arsen.semionov at gmail.com  Thu Aug  1 22:39:25 2013
From: arsen.semionov at gmail.com (Arsen)
Date: Thu, 1 Aug 2013 21:39:25 +0300
Subject: [Freeswitch-users] Freeswitch behind opensips behind NAT
In-Reply-To: <bc6c45aea39c3fa37dc6bcb601539254@mstn.com>
References: <bc6c45aea39c3fa37dc6bcb601539254@mstn.com>
Message-ID: <CABoraY5DOU+0z6YAoAuiLU43+VmzbTwaiBMLd+jHq97WOLfvJw@mail.gmail.com>

Hi Jose,

Your problem most likely related to opensips nat traversal. It is not
(only) freeswitch issue.
You can try to use opensips nathelper module in combination with rtpproxy.
Take your attention on on_reply routes in opensips.

Regards,
Arsen

On Thursday, August 1, 2013, Jose Suero wrote:

> I'm trying to setup freeswitch with opensips on EC2 I'm using opensips
> dispatcher to send registrations to freeswitch and Load Balancer to send
> calls to freeswitch this I took from
> http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS
>
> I have opensips ext-sip-ip and ext-rtp-ip on external profile set to my
> external IP, and if I connect the phones directly to freeswitch I can
> hear sound. But when I go thru Opensips I can register the phone to
> freeswitch, but when I try calling the phone is trying to connect to RTP
> on my internal address. I'm using internal address to connect from one
> server to the other
>
> Does anyone know how I can load balancer send the external address is
> getting from Freeswitch and not the internal IP is using to talk to the
> other server
>
> thanks in advance
>
>
> Jose Suero
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


-- 
Regards,
Arsen.
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From tru083 at yahoo.com  Thu Aug  1 23:02:06 2013
From: tru083 at yahoo.com (D D)
Date: Thu, 1 Aug 2013 12:02:06 -0700 (PDT)
Subject: [Freeswitch-users] Best way to play a message to the first member
	of a conference?
In-Reply-To: <CABoraY5DOU+0z6YAoAuiLU43+VmzbTwaiBMLd+jHq97WOLfvJw@mail.gmail.com>
References: <bc6c45aea39c3fa37dc6bcb601539254@mstn.com>
	<CABoraY5DOU+0z6YAoAuiLU43+VmzbTwaiBMLd+jHq97WOLfvJw@mail.gmail.com>
Message-ID: <1375383726.60030.YahooMailNeo@web120701.mail.ne1.yahoo.com>

Hi,

I would like to play a custom message when someone joins a conference.
I know I can play a message to each joiner individually, but would rather
play the message into the conference as a whole.

I have noticed that when the first member joins the conference, there is a
delay of up to a few seconds after they join before they can hear the
announcement, so the initial part of the announcement is missed. ?It doesn't
seem to happen for subsequent joiners.

I have tried adding a delay before playing the message, but it doesn't reliably
work.

Any ideas why this initial audio missing occurs, and if there is a workaround?

Thanks,
David
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From jpyle at fidelityvoice.com  Thu Aug  1 23:35:43 2013
From: jpyle at fidelityvoice.com (Jeff Pyle)
Date: Thu, 1 Aug 2013 15:35:43 -0400
Subject: [Freeswitch-users] removing FS from media path after transfer
Message-ID: <CACYJG3Ja5o26GCkUWFrg=70YUG1e1jTQZ8tSbBpM5DVDH1UaAg@mail.gmail.com>

Hello,

This is on version 1.2.10 installed from apt repo onto Debian wheezy.

I have two sofia profiles configured, both with inbound-bypass-media set
true.  When a call is bridged across them FS stays out of the media path.
 The SDP from one side to the other is untouched.  Excellent.  I need to
keep FS out of the media path unless it's actually generating media, for
example, ringback after a REFER.  When one side sends a REFER, FS handles
it properly accordingly to the dialplan, including ringback generation.  FS
keeps itself in the media path after the new endpoint answers.  How can I
configure it to remove itself?

Both sofia profiles have:

    <param name="inbound-bypass-media" value="true"/>
    <param name="media-option" value="resume-media-on-hold"/>
    <param name="media-option" value="bypass-media-after-att-xfer"/>

It doesn't seem to matter, FS stays in the media path.  What might I be
missing?


- Jeff
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From jpyle at fidelityvoice.com  Thu Aug  1 23:38:54 2013
From: jpyle at fidelityvoice.com (Jeff Pyle)
Date: Thu, 1 Aug 2013 15:38:54 -0400
Subject: [Freeswitch-users] Codec order flipped in proxy-media mode
	re-INVITE
In-Reply-To: <CAKbxfG9m1xU8t2X7iZ60WLQu5JCWZCit=xtKeGGzomDDaG840Q@mail.gmail.com>
References: <CACYJG3+M+KVH30Roj_MB2zo7a3hxQOw+NkBsSWkNZe=OS-jqPg@mail.gmail.com>
	<CAKbxfG9m1xU8t2X7iZ60WLQu5JCWZCit=xtKeGGzomDDaG840Q@mail.gmail.com>
Message-ID: <CACYJG3+dgHd2MMhGdt02BC64ym3X9=wF6KDSyfQwvQTodMouzQ@mail.gmail.com>

This seems to break other things.  I lose media, possibly when the far end
sends 200 OK.  I didn't have an opportunity to do much troubleshooting.

I'm working a different approach that includes removing FS from the media
path altogether.


- Jeff


On Thu, Jul 25, 2013 at 8:49 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Try enable-soa=false in the Sofia profile to disable any auto parsing.
> On Jul 25, 2013 6:41 AM, "Jeff Pyle" <jpyle at fidelityvoice.com> wrote:
>
>> Hello,
>>
>> FreeSWITCH Version 1.3.17+git~20130318T211211Z~2dc3b47db1.  The config is
>> a basic SBC to handle NAT traversal between public and private interfaces.
>>  Proxy-media mode is configured.
>>
>> Inbound call from carrier hits us with G711u and G729 in the SDP, in that
>> order.  Inside PBX chooses G711u and call is established.  30 minutes later
>> the carrier does a session refresh with a re-INVITE with an identical SDP
>> as the initial INVITE.  When FreeSWITCH the re-INVITE leaves FreeSWITCH
>> towards the inside PBX, the attached SDP has G711u and G729 are flipped
>> such that G729 is now preferred.  The PBX accepts the G729 request and the
>> session switches to G729.
>>
>> There are no SDP smarts in the dialplan.  Codecs are not handled at all
>> by design.
>>
>> Is this something that could be caused by a misconfiguration?  Or, is my
>> next step a jira?
>>
>>
>>
>> - Jeff
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 john at millican.us  Thu Aug  1 23:45:45 2013
From: john at millican.us (john at millican.us)
Date: Thu, 01 Aug 2013 15:45:45 -0400
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <CE200751.7876B%krice@freeswitch.org>
References: <CE200751.7876B%krice@freeswitch.org>
Message-ID: <51FABAE9.8070709@millican.us>

Hello,
If you are looking for a master-master replication I can say that 
Percona Xtradb cluster works very well.  There are a few extra settings 
that need to be in my.cnf pertaining to wsrep but not too bad to set up, 
certainly not painful.  I am currently running a 3 master cluster for an 
eCommerce site.   I can not speak to cross data center replication 
though as all three are racked above one another.  MMySQL is pretty 
quick but can it not compare to MongoDB or CouchBase.  I have used many 
SQL db and a few NoSQL and the NoSQL are much faster, easier to scale, 
and just plain more fun. My opinion and of course YMMV
JohnM

On 8/1/2013 1:38 PM, Ken Rice wrote:
> Re: [Freeswitch-users] FreeSWITCH Core - Cassandra Database Postgresql 
> offers streaming replication in PG9 versions... Its a 
> master/(warm)standby system but it works quite well (I use the native 
> pgsql replication for high volume calling). PostgreSQL is BSD licensed 
> so this has also lead to several addons that use various methods for 
> replication... Some are opensourced others are not...
>
> On 8/1/13 12:10 PM, "Lloyd Aloysius" <lloyd.aloysius at sunteltech.ca> wrote:
>
>     Ken,
>
>     Thank you for the information. Can you please tell me how can we
>     scale with postgres with multiple Data centers. I think postgres
>     - replication is not possible?
>
>
>     Thanks
>     Lloyd
>     *
>       <http://www.sunteltech.ca/blog/>
>     *
>
>     On Thu, Aug 1, 2013 at 12:52 PM, Ken Rice <krice at freeswitch.org>
>     wrote:
>
>         the core of freeswtich requires a sql database. changing it to
>         use a nosql database is not a trivial task.
>
>         Postgres is probably the most recommended database to use and
>         is used in many high volume production environments. (its that
>         the developers of freeswitch primarily use)
>         Ken
>         Sent from my iPad
>
>         On Aug 1, 2013, at 11:40, Ashwin Jain <ashwinrkjain at gmail.com>
>         wrote:
>
>             Hi,
>
>             I am also looking for similar kind of solution.
>
>             I have a web application for which I am using backend as
>             Cassandra.
>             My web application also controls Freeswitch (as in it
>             contains the UI for setting up bridging for calls). So,
>             right now, for that I wrote a sync service which fetches
>             data from Cassandra and puts it in MySQL.
>
>
>
>             On Thu, Aug 1, 2013 at 9:53 PM, Lloyd Aloysius
>             <lloyd.aloysius at sunteltech.ca> wrote:
>
>
>                 if you use MySQL  replication is painful. It is not
>                 faster. Binary logs are killing the cpu and memory.
>                 Lots of interest in this direction. I saw OpenSIP
>                 using Casendra Driver for distribution.
>
>                 LLoyd
>                 *  <http://www.sunteltech.ca/blog/>
>                 *
>
>                 On Thu, Aug 1, 2013 at 12:04 PM, Jeff Leung
>                 <jleung at v10networks.ca> wrote:
>
>                     I'm not sure of Cassandra database will work as a
>                     database for the FreeSWICH core since the syntax
>                     looks SQL but isn't.
>
>                     If It provides ODBC connectivity, you may as well
>                     as give it a shot.
>
>
>                     *From:*
>                     freeswitch-users-bounces at lists.freeswitch.org
>                     [mailto:freeswitch-users-bounces at lists.freeswitch.org]
>                     *On Behalf Of *Lloyd Aloysius
>                     *Sent:* Thursday, August 1, 2013 8:14 AM
>                     *To:* FreeSWITCH Users Help
>                     *Subject:* [Freeswitch-users] FreeSWITCH Core -
>                     Cassandra Database
>
>
>
>                     Hi All
>
>
>
>                     Does any one have experience with FreeSWITCH Core
>                     and Cassandra Database. Sounds like Cassandra
>                     (http://cassandra.apache.org/) have a better
>                     replication system. This will help to distribute
>                     freeswitch across multiple data centers and
>                     provide a better fail over system.
>
>
>
>                     Any feedback ? Thank you in advance.
>
>
>
>                     Thanks
>
>                     Lloyd
>
>                     _________________________________________________________________________
>                     Professional FreeSWITCH Consulting Services:
>                     consulting at freeswitch.org
>                     http://www.freeswitchsolutions.com
>
>                     FreeSWITCH-powered IP PBX: The CudaTel
>                     Communication Server
>                     
>
>                     Official FreeSWITCH Sites
>                     http://www.freeswitch.org
>                     http://wiki.freeswitch.org
>                     http://www.cluecon.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
>
>
>
>                 _________________________________________________________________________
>                 Professional FreeSWITCH Consulting Services:
>                 consulting at freeswitch.org
>                 http://www.freeswitchsolutions.com
>
>                 FreeSWITCH-powered IP PBX: The CudaTel Communication
>                 Server
>                 
>
>                 Official FreeSWITCH Sites
>                 http://www.freeswitch.org
>                 http://wiki.freeswitch.org
>                 http://www.cluecon.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
>
>
>
>
> -- 
> Ken
> _http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> _irc.freenode.net #freeswitch
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 freeswitch-list at puzzled.xs4all.nl  Fri Aug  2 00:19:18 2013
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Thu, 01 Aug 2013 22:19:18 +0200
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <51FABAE9.8070709@millican.us>
References: <CE200751.7876B%krice@freeswitch.org>
	<51FABAE9.8070709@millican.us>
Message-ID: <51FAC2C6.4030107@puzzled.xs4all.nl>

On 08/01/2013 09:45 PM, john at millican.us wrote:
[snip]
> I have used many
> SQL db and a few NoSQL and the NoSQL are much faster, easier to scale,
> and just plain more fun. My opinion and of course YMMV

Doesn't MongoDB acknowledge a write when it still has the data in mem as 
opposed to written on disk? So when the power fails and both FreeSWITCH 
server 1 and MongoDB server 1 are down there was data in memory that has 
not been replicated to MongoDB server 2. How is that going to result in 
proper failover? How can you fire up FreeSWITCH server 2 with all call 
data when MongoDB server 2 does not have all the call data? I'm no NoSQL 
expert so would love to hear how a NoSQL based solution could do the job 
reliably.

Regards,
Patrick


From oak.hidden at gmail.com  Fri Aug  2 00:14:36 2013
From: oak.hidden at gmail.com (woot root)
Date: Thu, 1 Aug 2013 15:14:36 -0500
Subject: [Freeswitch-users] conference add-member event delay
Message-ID: <CABnA6+aB2F430ZvTwdLHpw5rMsMxXPXtL+hkhkQjwrnijLym5g@mail.gmail.com>

Recently i noticed we have this strange issue:

what we are doing is:
first we create a conference for AGENT (because several party needed for
the phone call)
then we start dialing LEAD
once LEAD answered the phone, I'll transfer this LEAD into AGENT's
conference by using
> uuid_transfer  LEAD 'conference:AGENT' inline

if everything goes well, I'll receive this event very quick (usually less
than one second)
 >Event-Subclass:conference::maintenance
 >Action:add-member
so i could get the LEAD's member_id (to do mute/deaf with member_id)

But, now, sometimes -- not always, we could not receive the
conference::maintenance event with member_id after uuid_transfer command,
it will take around 7 seconds to receive!

I spend a lot of time on this issue, but still could not find any clue, and
don't know how to reproduce it, but it still happens from time to time...

I hope somebody could help me out.

Thanks!!

Eric
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From pasha at prosperity4ever.com  Fri Aug  2 00:31:30 2013
From: pasha at prosperity4ever.com (Paul)
Date: Thu, 01 Aug 2013 20:24:30 -0007
Subject: [Freeswitch-users] No audio on either of an established call
In-Reply-To: <CABkS=xqFknFok1NWgoB+tBhrDB+GzrQNA-wA+CjnftpmSmuFHA@mail.gmail.com>
References: <CABkS=xpqqwJBd=cHGzz_aj3r8kvuQssP4HcrYCyCj9oP5EWLTg@mail.gmail.com>
	<CABkS=xqFknFok1NWgoB+tBhrDB+GzrQNA-wA+CjnftpmSmuFHA@mail.gmail.com>
Message-ID: <20130801203132.0612157E002@mail.mydcs.ca>

Where on your diagram is the Router? I assume that your Kamailio 1 and 
Kamailio 2 are on two different networks?

Paul

On Wed, 31 Jul, 2013 at 10:12 PM, Peter <eidevm5 at gmail.com> wrote:
> After checking everything for the umpteenth time, I discovered I was 
> missing a route back to 10.1.254.0/24
> 
> Once I put that in, bingo, audio!
> 
> 
> On Thu, Aug 1, 2013 at 12:33 PM, Peter <eidevm5 at gmail.com> wrote:
>> I currently have 2 SIP clients (Linphone) successfully calling each 
>> other, but there is no audio on either end.
>> 
>> The set up is as follows:
>> 
>> 
>> 
>> Linphone1 (1000)? --> Kamailio 1?? <-------> Freeswitch?? 
>> <------>? Kamalio 2? <---- Linphone2 (2000)
>> 
>> 
>> Using Freeswitch 1.2.12 on CentOS (installed via RPM)
>> 
>> Freeswitch has two interfaces:
>> 
>> external - 10.1.1.206
>> internal -? 10.10.10.206
>> 
>> Each of the Linphone clients are registered their respective 
>> Kamailio instance and Kamailio is configured to route via the 
>> appropriate interface on Freeswitch.
>> 
>> The SIP negotiation is working as I can call either Linphone 
>> client.?? 
>> 
>> I've done a tcpdump on each side of? Freeswitch and can see the RTP 
>> traffic between the Linphone and the appropriate interface on 
>> Freeswitch.
>> 
>> I've tried different codec combinations (mostly G711 and iLBC), and 
>> different SIP clients but still get no audio.?? 
>> 
>> Any pointers on how to track down the issue?
>> 
>> Thanks
>> 
>> Peter
>> 
>> 
> 
> 
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From pasha at prosperity4ever.com  Fri Aug  2 01:13:02 2013
From: pasha at prosperity4ever.com (Paul)
Date: Thu, 01 Aug 2013 21:06:02 -0007
Subject: [Freeswitch-users] FreeSWITCH -- 30 Second call drop
In-Reply-To: <20130801204721.GC20191@0rdior.com>
References: <20130801204721.GC20191@0rdior.com>
Message-ID: <20130801211304.35C9957E002@mail.mydcs.ca>



On Thu, 1 Aug, 2013 at 1:47 PM, E.J.C. Lindner <office at fordior.net> 
wrote:
> Hi Paul,
> 
> I saw you emails regarding 30 seconds call drop. I have exactly the
> same issue when I want to connect my client over OpenVPN to my FS
> server, which is also the vpn server at the same time (server is
> on a public IP). 
> 
> All my calls are dropped after 32 seconds... very annoying and I can't
> find the solution. So I'm constantly switching config for standard vs 
> vpn 
> for my client and FS (the standard config is just connecting to the
> public ip which is unencrypted).
> 
> Did you find a solution in the mean time? Please inform me, cause it's
> really killing me. :=|
> 
> I hope you can inform me on a short term; really appreciated!
> 
> -- 
> Yours sincerely, 
> E.J.C. Lindner
> 
Yes I found a solution to my problem.

My case differs a little bit from yours in that my openvpn server is on 
the router, not on the PBX, in your case it's probably even a little 
simpler.

So here is what I can advise you,

1. Ensure NAT is working properly over openVPN, so to test that all you 
need to do is have 2 phones on the same PBX able to call each other 
internally (extension 100 to extension 101 for instance).

If that works then your VPN NAT is probably good.

2. Ensure that your PBX is reporting the correct external address to 
your vendor upstream (this is where it was messing me up).

For example, for me my PBX was 10.0.0.40 (itnernal address) which the 
router was NATing to 24.38.231.4 (just a sample external address) and 
the same on the way back, when a request would come in for 24.38.231.4 
it would translate it to 10.0.0.40.

Because my PBX was not aware of 24.38.231.4 as it's external address, 
it was only listening on 10.0.0.40, it wasn't responding to the SIP ACK 
packets correctly, and the call was being perceived as finished and 
therefore hung up.

To fix it in FreeSwitch I had to edit external.xml and insert the 
proper IP address for sip_ext_ip and rtp_ext_ip (this is from memory)

I hope this helps you out cuz it cost me hours of frustration :)

Paul
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From juanjo at comellas.org  Fri Aug  2 02:13:25 2013
From: juanjo at comellas.org (Juan Jose Comellas)
Date: Thu, 1 Aug 2013 19:13:25 -0300
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <51FAC2C6.4030107@puzzled.xs4all.nl>
References: <CE200751.7876B%krice@freeswitch.org>
	<51FABAE9.8070709@millican.us> <51FAC2C6.4030107@puzzled.xs4all.nl>
Message-ID: <CAOLqyqwJ1S3hz6cxWVOL=3Eu_1EAxz7z=4W5KDx5Mcfq2_tguw@mail.gmail.com>

You can modify this behavior in MongoDB by changing the "write concern" [1]
to ensure that the MongoDB client only returns to the caller when at least
one of the replicas has acknowledged the data. The problem is that if you
do that the performance drops dramatically. In my opinion, MongoDB is only
suitable for storing non-critical data. I would never use it for a failover
solution with FreeSWITCH.

BTW, moving from a relational database to a NoSQL one is not trivial at
all, especially if your application requires any kind of transactionality
from the DB.

[1] http://docs.mongodb.org/manual/core/write-concern/


On Thu, Aug 1, 2013 at 5:19 PM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 08/01/2013 09:45 PM, john at millican.us wrote:
> [snip]
> > I have used many
> > SQL db and a few NoSQL and the NoSQL are much faster, easier to scale,
> > and just plain more fun. My opinion and of course YMMV
>
> Doesn't MongoDB acknowledge a write when it still has the data in mem as
> opposed to written on disk? So when the power fails and both FreeSWITCH
> server 1 and MongoDB server 1 are down there was data in memory that has
> not been replicated to MongoDB server 2. How is that going to result in
> proper failover? How can you fire up FreeSWITCH server 2 with all call
> data when MongoDB server 2 does not have all the call data? I'm no NoSQL
> expert so would love to hear how a NoSQL based solution could do the job
> reliably.
>
> Regards,
> Patrick
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jmesquita at freeswitch.org  Fri Aug  2 02:32:24 2013
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 1 Aug 2013 19:32:24 -0300
Subject: [Freeswitch-users] FreeSWITCH Utilization in Government and
	Municipalities
In-Reply-To: <51FA7DBF.7040006@freeswitch.org>
References: <0AB0A933E0F9D24EAD7873D7B2241546011ECE42@EOCEXVS01.brevardco.int>
	<51FA7DBF.7040006@freeswitch.org>
Message-ID: <CABH8QX2j+YCW0KhDn+8rtZ5Eru8pYJM=oothid_bO95jaM1tZQ@mail.gmail.com>

Jeff, I am going to ClueCon and I have a big installation of fs on the gov
I south america. I know, not the US but still might be able to help. I have
1500 extensions across 2 server sets. On the airport now flying to the us.
I will also give a 15 min talk on the subject but not sure when. Lol.
On Aug 1, 2013 12:28 PM, "Raymond Chandler" <intralanman at freeswitch.org>
wrote:

>  On 07/31/2013 03:27 PM, McKnight, Jeffrey wrote:
>
>  I am implementing FreeSWITCH in my county in the very near future.****
>
> Happy to hear that. Especially as a former Florida resident :-)
>
>  ** **
>
> I have an approved preliminary plan to roll out a couple of hundred VOIP
> phones, but need case studies of FreeSWITCH utilized in government or
> municipal applications in order to proceed on a larger scale. Fire
> departments, law enforcement, emergency operations, and 911 applications
> would also be helpful.  It appears that some feel that government is so
> different than commercial applications, that it warrants further
> investigation.****
>
>  There are some cases in which that is true.  When you start talking
> about dealing with 911, police, fire, etc then it's worth noting to keep in
> mind those with disabilities that would require TDD (which FreeSWITCH has
> support for) and other accomodations.  Aside from that, there aren't too
> many things that I can think of that are much different than normal
> enterprise installations with thousands of users.
>
> Looking forward to seeing you at ClueCon
>
>
> -Ray
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 avi at avimarcus.net  Fri Aug  2 02:32:25 2013
From: avi at avimarcus.net (Avi Marcus)
Date: Thu, 1 Aug 2013 22:32:25 +0000
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <CAOLqyqwJ1S3hz6cxWVOL=3Eu_1EAxz7z=4W5KDx5Mcfq2_tguw@mail.gmail.com>
References: <CE200751.7876B%krice@freeswitch.org>
	<51FABAE9.8070709@millican.us> <51FAC2C6.4030107@puzzled.xs4all.nl>
	<CAOLqyqwJ1S3hz6cxWVOL=3Eu_1EAxz7z=4W5KDx5Mcfq2_tguw@mail.gmail.com>
Message-ID: <000001403c02b723-f5ba4387-6ab1-4eb0-abd6-5a7f1c945371-000000@email.amazonses.com>

Writing to disk isn't a uniquely mongodb issue - even many SQL dbs have
configurable options for write durability.
But for failover replication, the writing to disk is less of the issue than
the latency of the replication...
-Avi

On Fri, Aug 2, 2013 at 1:13 AM, Juan Jose Comellas <juanjo at comellas.org>wrote:

> You can modify this behavior in MongoDB by changing the "write concern"
> [1] to ensure that the MongoDB client only returns to the caller when at
> least one of the replicas has acknowledged the data. The problem is that if
> you do that the performance drops dramatically. In my opinion, MongoDB is
> only suitable for storing non-critical data. I would never use it for a
> failover solution with FreeSWITCH.
>
> BTW, moving from a relational database to a NoSQL one is not trivial at
> all, especially if your application requires any kind of transactionality
> from the DB.
>
> [1] http://docs.mongodb.org/manual/core/write-concern/
>
>
>
> On Thu, Aug 1, 2013 at 5:19 PM, Patrick Lists <
> freeswitch-list at puzzled.xs4all.nl> wrote:
>
>> On 08/01/2013 09:45 PM, john at millican.us wrote:
>> [snip]
>> > I have used many
>> > SQL db and a few NoSQL and the NoSQL are much faster, easier to scale,
>> > and just plain more fun. My opinion and of course YMMV
>>
>> Doesn't MongoDB acknowledge a write when it still has the data in mem as
>> opposed to written on disk? So when the power fails and both FreeSWITCH
>> server 1 and MongoDB server 1 are down there was data in memory that has
>> not been replicated to MongoDB server 2. How is that going to result in
>> proper failover? How can you fire up FreeSWITCH server 2 with all call
>> data when MongoDB server 2 does not have all the call data? I'm no NoSQL
>> expert so would love to hear how a NoSQL based solution could do the job
>> reliably.
>>
>> Regards,
>> Patrick
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 Aug  2 03:14:23 2013
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 1 Aug 2013 16:14:23 -0700
Subject: [Freeswitch-users] FreeSWITCH Utilization in Government and
	Municipalities
In-Reply-To: <CABH8QX2j+YCW0KhDn+8rtZ5Eru8pYJM=oothid_bO95jaM1tZQ@mail.gmail.com>
References: <0AB0A933E0F9D24EAD7873D7B2241546011ECE42@EOCEXVS01.brevardco.int>
	<51FA7DBF.7040006@freeswitch.org>
	<CABH8QX2j+YCW0KhDn+8rtZ5Eru8pYJM=oothid_bO95jaM1tZQ@mail.gmail.com>
Message-ID: <CAKzWOxV2kYQLvk0Oe2+_Cg-rs=E30hLsoTUzPuyW3Ath-MsHWA@mail.gmail.com>

On Thu, Aug 1, 2013 at 3:32 PM, Jo?o Mesquita <jmesquita at freeswitch.org>wrote:

> Jeff, I am going to ClueCon and I have a big installation of fs on the gov
> I south america. I know, not the US but still might be able to help. I have
> 1500 extensions across 2 server sets. On the airport now flying to the us.
> I will also give a 15 min talk on the subject but not sure when. Lol.
>
Lightning talk scheduled for 1:15PM on Wednesday, right after Ken Rice.
-MC

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From juanjo at comellas.org  Fri Aug  2 04:44:22 2013
From: juanjo at comellas.org (Juan Jose Comellas)
Date: Thu, 1 Aug 2013 21:44:22 -0300
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <000001403c02b723-f5ba4387-6ab1-4eb0-abd6-5a7f1c945371-000000@email.amazonses.com>
References: <CE200751.7876B%krice@freeswitch.org>
	<51FABAE9.8070709@millican.us> <51FAC2C6.4030107@puzzled.xs4all.nl>
	<CAOLqyqwJ1S3hz6cxWVOL=3Eu_1EAxz7z=4W5KDx5Mcfq2_tguw@mail.gmail.com>
	<000001403c02b723-f5ba4387-6ab1-4eb0-abd6-5a7f1c945371-000000@email.amazonses.com>
Message-ID: <CAOLqyqwWBm7XjNtAAXbL=1ZfU2qqZUj=j5h-29TMUHy51TxWVw@mail.gmail.com>

That's true, but what's unique about MongoDB, AFAIK, is that by default the
official client would inform the caller that the write had been successful
before the data had even reached the server. This has been changed
recently, though, but the default durability settings are still too weak.


On Thu, Aug 1, 2013 at 7:32 PM, Avi Marcus <avi at avimarcus.net> wrote:

> Writing to disk isn't a uniquely mongodb issue - even many SQL dbs have
> configurable options for write durability.
> But for failover replication, the writing to disk is less of the issue
> than the latency of the replication...
> -Avi
>
>
> On Fri, Aug 2, 2013 at 1:13 AM, Juan Jose Comellas <juanjo at comellas.org>wrote:
>
>> You can modify this behavior in MongoDB by changing the "write concern"
>> [1] to ensure that the MongoDB client only returns to the caller when at
>> least one of the replicas has acknowledged the data. The problem is that if
>> you do that the performance drops dramatically. In my opinion, MongoDB is
>> only suitable for storing non-critical data. I would never use it for a
>> failover solution with FreeSWITCH.
>>
>> BTW, moving from a relational database to a NoSQL one is not trivial at
>> all, especially if your application requires any kind of transactionality
>> from the DB.
>>
>> [1] http://docs.mongodb.org/manual/core/write-concern/
>>
>>
>>
>> On Thu, Aug 1, 2013 at 5:19 PM, Patrick Lists <
>> freeswitch-list at puzzled.xs4all.nl> wrote:
>>
>>> On 08/01/2013 09:45 PM, john at millican.us wrote:
>>> [snip]
>>> > I have used many
>>> > SQL db and a few NoSQL and the NoSQL are much faster, easier to scale,
>>> > and just plain more fun. My opinion and of course YMMV
>>>
>>> Doesn't MongoDB acknowledge a write when it still has the data in mem as
>>> opposed to written on disk? So when the power fails and both FreeSWITCH
>>> server 1 and MongoDB server 1 are down there was data in memory that has
>>> not been replicated to MongoDB server 2. How is that going to result in
>>> proper failover? How can you fire up FreeSWITCH server 2 with all call
>>> data when MongoDB server 2 does not have all the call data? I'm no NoSQL
>>> expert so would love to hear how a NoSQL based solution could do the job
>>> reliably.
>>>
>>> Regards,
>>> Patrick
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jackal at cybershroud.net  Fri Aug  2 05:29:54 2013
From: jackal at cybershroud.net (Carlos Flor)
Date: Thu, 1 Aug 2013 21:29:54 -0400
Subject: [Freeswitch-users] FreeSWITCH -- 30 Second call drop
In-Reply-To: <20130801211304.35C9957E002@mail.mydcs.ca>
References: <20130801204721.GC20191@0rdior.com>
	<20130801211304.35C9957E002@mail.mydcs.ca>
Message-ID: <CAAYhoX+f9VcpnOde0-bAbcACpz2t0otnsP1zhsiEg+ZMjjxFGg@mail.gmail.com>

If that doesn't work, check your mtu and/or make sure you aren't blocking
icmp on your firewall if you want to do path-mtu discovery.  I had a 31
second call drop problem and it was because my icmp unreachable (needs
fragmentation) packets weren't getting all the way back to me (in my case
it was a bad route) and so I my device wasn't adjusting my mtu and the call
would drop exactly at 31 seconds in every single time.


On Thu, Aug 1, 2013 at 5:13 PM, Paul <pasha at prosperity4ever.com> wrote:

>
>
> On Thu, 1 Aug, 2013 at 1:47 PM, E.J.C. Lindner <office at fordior.net> wrote:
>
> Hi Paul, I saw you emails regarding 30 seconds call drop. I have exactly
> the same issue when I want to connect my client over OpenVPN to my FS
> server, which is also the vpn server at the same time (server is on a
> public IP). All my calls are dropped after 32 seconds... very annoying and
> I can't find the solution. So I'm constantly switching config for standard
> vs vpn for my client and FS (the standard config is just connecting to the
> public ip which is unencrypted). Did you find a solution in the mean time?
> Please inform me, cause it's really killing me. :=| I hope you can inform
> me on a short term; really appreciated!
> --
> Yours sincerely, E.J.C. Lindner
>
>
> Yes I found a solution to my problem.
>
> My case differs a little bit from yours in that my openvpn server is on
> the router, not on the PBX, in your case it's probably even a little
> simpler.
>
> So here is what I can advise you,
>
> 1. Ensure NAT is working properly over openVPN, so to test that all you
> need to do is have 2 phones on the same PBX able to call each other
> internally (extension 100 to extension 101 for instance).
>
> If that works then your VPN NAT is probably good.
>
> 2. Ensure that your PBX is reporting the correct external address to your
> vendor upstream (this is where it was messing me up).
>
> For example, for me my PBX was 10.0.0.40 (itnernal address) which the
> router was NATing to 24.38.231.4 (just a sample external address) and the
> same on the way back, when a request would come in for 24.38.231 .4 it
> would translate it to 10.0.0.40.
>
> Because my PBX was not aware of 24.38.231.4 as it's external address, it
> was only listening on 10.0.0.40, it wasn't responding to the SIP ACK
> packets correctly, and the call was being perceived as finished and
> therefore hung up.
>
> To fix it in FreeSwitch I had to edit external.xml and insert the proper
> IP address for sip_ext_ip and rtp_ext_ip (this is from memory)
>
> I hope this helps you out cuz it cost me hours of frustration :)
>
> Paul
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Fri Aug  2 05:34:21 2013
From: eidevm5 at gmail.com (Peter)
Date: Fri, 2 Aug 2013 11:34:21 +1000
Subject: [Freeswitch-users] No audio on either of an established call
In-Reply-To: <20130801203132.0612157E002@mail.mydcs.ca>
References: <CABkS=xpqqwJBd=cHGzz_aj3r8kvuQssP4HcrYCyCj9oP5EWLTg@mail.gmail.com>
	<CABkS=xqFknFok1NWgoB+tBhrDB+GzrQNA-wA+CjnftpmSmuFHA@mail.gmail.com>
	<20130801203132.0612157E002@mail.mydcs.ca>
Message-ID: <CABkS=xrgKqPkKjP2NjWMAefxOzdRTe3=JK18Xj6b7EQZJ6_b3Q@mail.gmail.com>

Correct.  Both Kamailio servers are on different networks.   I didn't show
the router in the mix for simplicity.


On Fri, Aug 2, 2013 at 6:31 AM, Paul <pasha at prosperity4ever.com> wrote:

> Where on your diagram is the Router? I assume that your Kamailio 1 and
> Kamailio 2 are on two different networks?
>
> Paul
>
>
> On Wed, 31 Jul, 2013 at 10:12 PM, Peter <eidevm5 at gmail.com> wrote:
>
> After checking everything for the umpteenth time, I discovered I was
> missing a route back to 10.1.254.0/24
>
> Once I put that in, bingo, audio!
>
>
> On Thu, Aug 1, 2013 at 12:33 PM, Peter <eidevm5 at gmail.com> wrote:
>
>> I currently have 2 SIP clients (Linphone) successfully calling each
>> other, but there is no audio on either end.
>>
>> The set up is as follows:
>>
>>
>>
>> Linphone1 (1000)  --> Kamailio 1   <-------> Freeswitch   <------>
>> Kamalio 2  <---- Linphone2 (2000)
>>
>>
>> Using Freeswitch 1.2.12 on CentOS (installed via RPM)
>>
>> Freeswitch has two interfaces:
>>
>> external - 10.1.1.206
>> internal -  10.10.10.206
>>
>> Each of the Linphone clients are registered their respective Kamailio
>> instance and Kamailio is configured to route via the appropriate interface
>> on Freeswitch.
>>
>> The SIP negotiation is working as I can call either Linphone client.
>>
>> I've done a tcpdump on each side of  Freeswitch and can see the RTP
>> traffic between the Linphone and the appropriate interface on Freeswitch.
>>
>> I've tried different codec combinations (mostly G711 and iLBC), and
>> different SIP clients but still get no audio.
>>
>> Any pointers on how to track down the issue?
>>
>> Thanks
>>
>> Peter
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nneul at mst.edu  Fri Aug  2 07:12:07 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Thu, 01 Aug 2013 22:12:07 -0500
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <51FABAE9.8070709@millican.us>
References: <CE200751.7876B%krice@freeswitch.org>
	<51FABAE9.8070709@millican.us>
Message-ID: <51FB2387.10907@mst.edu>

I've had good luck with a small deployment on percona as well, though there are a few minor issues (maybe stuff that can 
be corrected):

1. When FS tries to create indexes on various tables that already exist, it winds up causing a bunch of 
complaints/failures in the percona logs and creating various GAL_ dump files.

2. The external management of percona can be a mess. I tried running with Pacemaker/Corosync and really didn't have good 
luck with it failing over the floating IP properly, so I switched things back over to Keepalived, which I'm using for 
the freeswitch instances.

-- Nathan

On 08/01/2013 02:45 PM, john at millican.us wrote:
> Hello,
> If you are looking for a master-master replication I can say that Percona Xtradb cluster works very well.  There are a
> few extra settings that need to be in my.cnf pertaining to wsrep but not too bad to set up, certainly not painful.  I am
> currently running a 3 master cluster for an eCommerce site.   I can not speak to cross data center replication though as
> all three are racked above one another.  MMySQL is pretty quick but can it not compare to MongoDB or CouchBase.  I have
> used many SQL db and a few NoSQL and the NoSQL are much faster, easier to scale, and just plain more fun. My opinion and
> of course YMMV
> JohnM
>
> On 8/1/2013 1:38 PM, Ken Rice wrote:
>> Re: [Freeswitch-users] FreeSWITCH Core - Cassandra Database Postgresql offers streaming replication in PG9 versions...
>> Its a master/(warm)standby system but it works quite well (I use the native pgsql replication for high volume
>> calling). PostgreSQL is BSD licensed so this has also lead to several addons that use various methods for
>> replication... Some are opensourced others are not...
>>
>> On 8/1/13 12:10 PM, "Lloyd Aloysius" <lloyd.aloysius at sunteltech.ca> wrote:
>>
>>     Ken,
>>
>>     Thank you for the information. Can you please tell me how can we scale with postgres with multiple Data centers. I
>>     think postgres - replication is not possible?
>>
>>
>>     Thanks
>>     Lloyd
>>     *
>>       <http://www.sunteltech.ca/blog/>
>>     *
>>
>>     On Thu, Aug 1, 2013 at 12:52 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>>         the core of freeswtich requires a sql database. changing it to use a nosql database is not a trivial task.
>>
>>         Postgres is probably the most recommended database to use and is used in many high volume production
>>         environments. (its that the developers of freeswitch primarily use)
>>         Ken
>>         Sent from my iPad
>>
>>         On Aug 1, 2013, at 11:40, Ashwin Jain <ashwinrkjain at gmail.com> wrote:
>>
>>             Hi,
>>
>>             I am also looking for similar kind of solution.
>>
>>             I have a web application for which I am using backend as Cassandra.
>>             My web application also controls Freeswitch (as in it contains the UI for setting up bridging for calls).
>>             So, right now, for that I wrote a sync service which fetches data from Cassandra and puts it in MySQL.
>>
>>
>>
>>             On Thu, Aug 1, 2013 at 9:53 PM, Lloyd Aloysius <lloyd.aloysius at sunteltech.ca> wrote:
>>
>>
>>                 if you use MySQL  replication is painful. It is not faster. Binary logs are killing the cpu and
>>                 memory. Lots of interest in this direction. I saw OpenSIP using Casendra Driver for distribution.
>>
>>                 LLoyd
>>                 *  <http://www.sunteltech.ca/blog/>
>>                 *
>>
>>                 On Thu, Aug 1, 2013 at 12:04 PM, Jeff Leung <jleung at v10networks.ca> wrote:
>>
>>                     I?m not sure of Cassandra database will work as a database for the FreeSWICH core since the syntax
>>                     looks SQL but isn?t.
>>
>>                     If It provides ODBC connectivity, you may as well as give it a shot.
>>
>>
>>                     *From:* freeswitch-users-bounces at lists.freeswitch.org
>>                     [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Lloyd Aloysius
>>                     *Sent:* Thursday, August 1, 2013 8:14 AM
>>                     *To:* FreeSWITCH Users Help
>>                     *Subject:* [Freeswitch-users] FreeSWITCH Core - Cassandra Database
>>
>>
>>
>>                     Hi All
>>
>>
>>
>>                     Does any one have experience with FreeSWITCH Core and Cassandra Database. Sounds like Cassandra
>>                     (http://cassandra.apache.org/) have a better replication system. This will help to distribute
>>                     freeswitch across multiple data centers and provide a better fail over system.
>>
>>
>>
>>                     Any feedback ? Thank you in advance.
>>
>>
>>
>>                     Thanks
>>
>>                     Lloyd
>>
>>                     _________________________________________________________________________
>>                     Professional FreeSWITCH Consulting Services:
>>                     consulting at freeswitch.org
>>                     http://www.freeswitchsolutions.com
>>
>>                     
>>                     
>>
>>                     Official FreeSWITCH Sites
>>                     http://www.freeswitch.org
>>                     http://wiki.freeswitch.org
>>                     http://www.cluecon.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
>>
>>
>>
>>                 _________________________________________________________________________
>>                 Professional FreeSWITCH Consulting Services:
>>                 consulting at freeswitch.org
>>                 http://www.freeswitchsolutions.com
>>
>>                 
>>                 
>>
>>                 Official FreeSWITCH Sites
>>                 http://www.freeswitch.org
>>                 http://wiki.freeswitch.org
>>                 http://www.cluecon.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
>>
>>
>>
>>
>> --
>> Ken
>> _http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> _irc.freenode.net #freeswitch
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From bdfoster at davri.com  Fri Aug  2 09:10:13 2013
From: bdfoster at davri.com (Brian Foster)
Date: Fri, 2 Aug 2013 01:10:13 -0400
Subject: [Freeswitch-users] Mod_directory: "connecting call" sound?
Message-ID: <CAHgk4BXBAg8EHqizJz8QoCvsxjS_WGKY=tydoMfjpmB3RKfeLg@mail.gmail.com>

I can't find it anywhere in the wiki or in the config files, but is there a
way to specify a "connecting call" sound?

Example: A caller enters the directory application, he/she selects an
entry. When they select the entry, it goes straight to ringing. Before that
call is connected, can we specify that it plays a sound such as "One moment
while I connect your call"?

I realize this might be a feature request, in which case I'll probably take
it upon myself to rummage through the source code and come up with a patch
to go with a JIRA ticket.

Thank you,

Brian Foster
Project Manager/Owner's Representative
Davri Investments, Incorporated
P: +1-317-787-2686
M: +1-317-600-9753
Indianapolis, Indiana
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From kris at kriskinc.com  Fri Aug  2 09:54:52 2013
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Fri, 2 Aug 2013 01:54:52 -0400
Subject: [Freeswitch-users] FS Timezones
Message-ID: <CAKDfjgc5CRFndznhPCrFC67x11LXM-sw0UKL1hWfukAfqxtPXg@mail.gmail.com>

All,

  Looking through timezones.conf.xml some of the lines jump out at me:

<zone name="America/New_York" value="EST5EDT,M3.2.0,M11.1.0" />

  According to various bits of documentation on this format it appears
that the time of day (not day) specifying the change to daylight
savings is missing:

http://www-01.ibm.com/support/docview.wss?uid=isg3T1000252

  Does this mean the line should look more like this:

  <zone name="America/New_York"
value="EST5EDT,M3.2.0/2:00:00,M11.1.0/2:00:00" />

  ...or am I missing something?

Thanks!

-- 
Kristian Kielhofner


From mishehu at freeswitch.org  Fri Aug  2 10:20:53 2013
From: mishehu at freeswitch.org (Yossi Neiman)
Date: Fri, 02 Aug 2013 01:20:53 -0500
Subject: [Freeswitch-users] how to compress maked v1.2 into tarball and
 then make install
In-Reply-To: <CAHgk4BVex-PSBd=e4==J+=HBrw=2fuaL80Hy4TEDVP3u0i+jaA@mail.gmail.com>
References: <CAJNDQt4HwKZoXP=mjWB8EP6wi13a3HyQoEnKYhpUqyVot-SWGQ@mail.gmail.com>
	<BF1FDBD8-75BF-4039-BC5D-CC572025F8DD@jerris.com>
	<CAHgk4BVex-PSBd=e4==J+=HBrw=2fuaL80Hy4TEDVP3u0i+jaA@mail.gmail.com>
Message-ID: <51FB4FC5.40909@freeswitch.org>

Slackware user here - unless Bruce is trying to include the source files 
with the resulting binaries, the proper method is simply:

make install DESTDIR=/path/to/temporary/dir

Then /path/to/temporary/dir can be tar'ed up.

-Yossi

On 08/01/2013 10:42 AM, Brian Foster wrote:
>
> Isn't there tarballs on files.freeswitch.org 
> <http://files.freeswitch.org>?
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com <mailto:bdfoster at davri.com>
> Indianapolis, Indiana
>
> Sent from a mobile device.
>
> On Aug 1, 2013 10:48 AM, "Michael Jerris" <mike at jerris.com 
> <mailto:mike at jerris.com>> wrote:
>
>     What OS? This is what packages are.
>
>     Mike
>
>     On Jul 31, 2013, at 10:39 PM, Bruce Lefko <blefko5361 at gmail.com
>     <mailto:blefko5361 at gmail.com>> wrote:
>
>     > I would like to bootstrap, configure, and make v1.2, and then
>     make the compiled code into a tarball.
>     >
>     > I can then use the tarball to install freeswitch on other
>     machines with the same OS, but much quicker.  This works with
>     master, but on v1.2, it seems that "make install" after extracting
>     the tarball is recompiling some libs.
>     >
>     > How can I avoid this?
>     >
>     > Thanks!
>     >
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 zhulizhong at live.com  Fri Aug  2 11:01:40 2013
From: zhulizhong at live.com (James zhu)
Date: Fri, 2 Aug 2013 07:01:40 +0000
Subject: [Freeswitch-users] Can not make out going call from sangoma A200
Message-ID: <BLU171-W3672C8625546F122FFC013B5510@phx.gbl>

hello:I installed freeswitch and sangoma A200 with 2 FXOs. i set the tone group to CNand opermode=china. the A200 can make accept incoming calls from FreeTDM/2:1 , but I use same port to make out going port, the shows the port is CONGESTION.--------------------------system log-----------------------------------------------2013-08-02 14:36:42.003531 [DEBUG] mod_freetdm.c:640 [2:1] FreeTDM/2:1/13811737925 CHANNEL HANGUP EXIT2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:48 FreeTDM/2:1/13811737925 Standard HANGUP, cause: NORMAL_CIRCUIT_CONGESTION2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:679 (FreeTDM/2:1/13811737925) State HANGUP going to sleep2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:692 (FreeTDM/2:1/13811737925) Callstate Change DOWN -> HANGUP2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:447 (FreeTDM/2:1/13811737925) State Change CS_HANGUP -> CS_REPORTING2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925) Running State Change CS_REPORTING2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State REPORTING2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:92 FreeTDM/2:1/13811737925 Standard REPORTING, cause: NORMAL_CIRCUIT_CONGESTION2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State REPORTING going to sleep2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:441 (FreeTDM/2:1/13811737925) State Change CS_REPORTING -> CS_DESTROY2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1549 Session 37 (FreeTDM/2:1/13811737925) Locked, Waiting on external entities2013-08-02 14:36:42.023531 [DEBUG] switch_ivr_originate.c:3632 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]2013-08-02 14:36:42.023531 [INFO] mod_dptools.c:3192 Originate Failed.  Cause: NORMAL_CIRCUIT_CONGESTION2013-08-02 14:36:42.023531 [NOTICE] switch_channel.c:4593 Hangup sofia/internal/1008 at 192.168.0.173 [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]2013-08-02 14:36:42.023531 [DEBUG] switch_channel.c:3135 Send signal sofia/internal/1008 at 192.168.0.173 [KILL]2013-08-02 14:36:42.023531 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal---------------------------------------------the FS is download from git and wanpipe is latest version.  please give a help for that issue.
Best regards,

James.zhu

website: www.hiastar.com 		 	   		  
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From rnbrady at gmail.com  Fri Aug  2 15:03:34 2013
From: rnbrady at gmail.com (Richard Brady)
Date: Fri, 2 Aug 2013 12:03:34 +0100
Subject: [Freeswitch-users] originate loopback
In-Reply-To: <CAKzWOxVH8F08OJEEF=kTi0TmGUBSAP=AFc_4uRO6HSqU7tU3Og@mail.gmail.com>
References: <CAN5esQ+kM-Du3hQ1K2efH7cnnNHaKBT2hDjUnA8x=Uohyshb5w@mail.gmail.com>
	<CAKzWOxVH8F08OJEEF=kTi0TmGUBSAP=AFc_4uRO6HSqU7tU3Og@mail.gmail.com>
Message-ID: <CAN5esQJxH=oCv9MOD3nLHHdxF-uLZW4OEh5TZ1KDbifTsJBYKA@mail.gmail.com>

Thanks Mike

Just ran this with debug and looks from the logs uas if the Lua is on the A
leg:

EXECUTE loopback/set:job=1234,answer,park-b set(job=1234)
EXECUTE loopback/set:job=1234,answer,park-b answer()
EXECUTE loopback/set:job=1234,answer,park-b park()
EXECUTE loopback/set:job=1234,answer,park-a lua(loop.lua)

I'll update the wiki.

Richard

On 1 August 2013 19:33, Michael Collins <msc at freeswitch.org> wrote:

>
>
>
> On Thu, Aug 1, 2013 at 9:29 AM, Richard Brady <rnbrady at gmail.com> wrote:
>
>> Guys
>>
>> Please help with this mind bender from the wiki at
>> http://wiki.freeswitch.org/wiki/Loopback_endpoint:
>>
>> Example 4: loopback with A-leg inline dialplan
>>
>> originate loopback/set:job=1234\,answer\,park/default/inline lua:loop.lua inline
>>
>> Executes *set* app, then *answer* app, then *park* app on the A leg then
>> bridges to the *lua* app on the B leg
>>
>> Is the para at the end correct? Isn't it the A-leg that executes the lua
>> and the B-leg that executes set,answer,park?
>>
>
> I'm pretty sure that the description is correct. It's equivalent to this:
>
> originate loopback/set:job=1234\,answer\,park/default/inline &lua(loop.lua)
>
>
> The Lua stuff is on the B leg and the set, answer, park stuff is on the
> loopback/A leg.
> -MC
>
>
>> Richard
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 freeswitch-list at puzzled.xs4all.nl  Fri Aug  2 16:12:20 2013
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 02 Aug 2013 14:12:20 +0200
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <51FB2387.10907@mst.edu>
References: <CE200751.7876B%krice@freeswitch.org>
	<51FABAE9.8070709@millican.us> <51FB2387.10907@mst.edu>
Message-ID: <51FBA224.7080207@puzzled.xs4all.nl>

On 08/02/2013 05:12 AM, Nathan Neulinger wrote:
> I've had good luck with a small deployment on percona as well, though there are a few minor issues (maybe stuff that can
> be corrected):
>
> 1. When FS tries to create indexes on various tables that already exist, it winds up causing a bunch of
> complaints/failures in the percona logs and creating various GAL_ dump files.
>
> 2. The external management of percona can be a mess. I tried running with Pacemaker/Corosync and really didn't have good
> luck with it failing over the floating IP properly, so I switched things back over to Keepalived, which I'm using for
> the freeswitch instances.

For the floating IP to follow to the new box, on a *2* node setup, you 
need to disable quorum and add a constraint in your pacemaker/corosync 
config so the resource (PerconaDB) follows the floating IP. Something like:

$ pcs property set no-quorum-policy=ignore
$ pcs constraint colocation add PerconaDB ClusterIP INFINITY
$ pcs constraint order ClusterIP then PerconaDB

And this example helps to prevent resources to move around when the 
failed node comes back online. Make sure the 'resource-stickiness' value 
is higher than the 'prefers' one:

$ pcs constraint location PerconaDB prefers node-1=50
$ pcs resource rsc defaults resource-stickiness=100

I'm curious, did you use a shared filesystem (something like DRBD with 
GFS2/OCFS2) and regular Percona XtraDB on both nodes or did you use 
Percona XtraDB Cluster?

Regards,
Patrick


From stuart.mills3 at btopenworld.com  Fri Aug  2 18:01:23 2013
From: stuart.mills3 at btopenworld.com (Stuart Mills)
Date: Fri, 2 Aug 2013 15:01:23 +0100
Subject: [Freeswitch-users] Call at once and XML-CDR
Message-ID: <C3ABD749BB4545119E17C1257F270BA9@PBPC>

Hi,

Got a question regarding call at once combined with xml-cdr logging.

The problem I seem to get is when I try sending calls to 2 destinations and 1 call gets answered, so that wins the race, the xml-cdr logging seems to be a little confused. In the app_log I?m getting logging up to the point of the dial string, then nothing after that for the a-leg. What seems to happen is the rest of the app_log gets written to the cdr file of the bleg that answered the call first.

Is this by design or a bug in the xml-cdr logging?

Thanks,

Stuart
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From lloyd.aloysius at sunteltech.ca  Fri Aug  2 18:01:56 2013
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Fri, 2 Aug 2013 10:01:56 -0400
Subject: [Freeswitch-users] Mod_directory: "connecting call" sound?
In-Reply-To: <CAHgk4BXBAg8EHqizJz8QoCvsxjS_WGKY=tydoMfjpmB3RKfeLg@mail.gmail.com>
References: <CAHgk4BXBAg8EHqizJz8QoCvsxjS_WGKY=tydoMfjpmB3RKfeLg@mail.gmail.com>
Message-ID: <CAGvuWYbb1kQKd+4wjtdk1dddJVYDoMwyVV1FHo7AMvJ3CE2bzQ@mail.gmail.com>

before call directory application set up a
channel variable call_from="directory" , set/export

in user dial plan before bridge check the channel variable call_from and if
match "directory", play the  "One moment while I connect your call"

Lloyd

*
*
*  <http://www.sunteltech.ca/blog/>*


On Fri, Aug 2, 2013 at 1:10 AM, Brian Foster <bdfoster at davri.com> wrote:

> I can't find it anywhere in the wiki or in the config files, but is there
> a way to specify a "connecting call" sound?
>
> Example: A caller enters the directory application, he/she selects an
> entry. When they select the entry, it goes straight to ringing. Before that
> call is connected, can we specify that it plays a sound such as "One moment
> while I connect your call"?
>
> I realize this might be a feature request, in which case I'll probably
> take it upon myself to rummage through the source code and come up with a
> patch to go with a JIRA ticket.
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Representative
> Davri Investments, Incorporated
> P: +1-317-787-2686
> M: +1-317-600-9753
> Indianapolis, Indiana
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lloyd.aloysius at sunteltech.ca  Fri Aug  2 18:10:31 2013
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Fri, 2 Aug 2013 10:10:31 -0400
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <51FBA224.7080207@puzzled.xs4all.nl>
References: <CE200751.7876B%krice@freeswitch.org>
	<51FABAE9.8070709@millican.us>
	<51FB2387.10907@mst.edu> <51FBA224.7080207@puzzled.xs4all.nl>
Message-ID: <CAGvuWYa6ptobw=9ey0uN5AatC-UBEidUxf+QbxDeBRWGFwWv7g@mail.gmail.com>

Core Dev'e preferred database is PostgreSQL. I will choose PostgreSQL.

There are so many issues with depends on one data centers. The
more critical now a days is weather cause lots of issues.

I want to distribute freeswitch, minimum two data centers and two different
regions.

Does anyone have a PostgreSQL master-master replication working reliably?
Anything I need to consider using PostgreSQL master-master.

Thank you in advanced.

Lloyd

*
*
*  <http://www.sunteltech.ca/blog/>*


On Fri, Aug 2, 2013 at 8:12 AM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 08/02/2013 05:12 AM, Nathan Neulinger wrote:
> > I've had good luck with a small deployment on percona as well, though
> there are a few minor issues (maybe stuff that can
> > be corrected):
> >
> > 1. When FS tries to create indexes on various tables that already exist,
> it winds up causing a bunch of
> > complaints/failures in the percona logs and creating various GAL_ dump
> files.
> >
> > 2. The external management of percona can be a mess. I tried running
> with Pacemaker/Corosync and really didn't have good
> > luck with it failing over the floating IP properly, so I switched things
> back over to Keepalived, which I'm using for
> > the freeswitch instances.
>
> For the floating IP to follow to the new box, on a *2* node setup, you
> need to disable quorum and add a constraint in your pacemaker/corosync
> config so the resource (PerconaDB) follows the floating IP. Something like:
>
> $ pcs property set no-quorum-policy=ignore
> $ pcs constraint colocation add PerconaDB ClusterIP INFINITY
> $ pcs constraint order ClusterIP then PerconaDB
>
> And this example helps to prevent resources to move around when the
> failed node comes back online. Make sure the 'resource-stickiness' value
> is higher than the 'prefers' one:
>
> $ pcs constraint location PerconaDB prefers node-1=50
> $ pcs resource rsc defaults resource-stickiness=100
>
> I'm curious, did you use a shared filesystem (something like DRBD with
> GFS2/OCFS2) and regular Percona XtraDB on both nodes or did you use
> Percona XtraDB Cluster?
>
> Regards,
> Patrick
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 avi at avimarcus.net  Fri Aug  2 18:24:45 2013
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 2 Aug 2013 14:24:45 +0000
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
Message-ID: <000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>

Are you sure these aren't b leg things?
Do you have an example of your dial plan and the logs?

-Avi
On Aug 2, 2013 5:09 PM, "Stuart Mills" <stuart.mills3 at btopenworld.com>
wrote:

>   Hi,
>
> Got a question regarding call at once combined with xml-cdr logging.
>
> The problem I seem to get is when I try sending calls to 2 destinations
> and 1 call gets answered, so that wins the race, the xml-cdr logging seems
> to be a little confused. In the app_log I?m getting logging up to the point
> of the dial string, then nothing after that for the a-leg. What seems to
> happen is the rest of the app_log gets written to the cdr file of the bleg
> that answered the call first.
>
> Is this by design or a bug in the xml-cdr logging?
>
> Thanks,
>
> Stuart
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 freeswitch-list at puzzled.xs4all.nl  Fri Aug  2 18:31:51 2013
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 02 Aug 2013 16:31:51 +0200
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <CAGvuWYa6ptobw=9ey0uN5AatC-UBEidUxf+QbxDeBRWGFwWv7g@mail.gmail.com>
References: <CE200751.7876B%krice@freeswitch.org>
	<51FABAE9.8070709@millican.us> <51FB2387.10907@mst.edu>
	<51FBA224.7080207@puzzled.xs4all.nl>
	<CAGvuWYa6ptobw=9ey0uN5AatC-UBEidUxf+QbxDeBRWGFwWv7g@mail.gmail.com>
Message-ID: <51FBC2D7.9040702@puzzled.xs4all.nl>

On 08/02/2013 04:10 PM, Lloyd Aloysius wrote:
> Core Dev'e preferred database is PostgreSQL. I will choose PostgreSQL.
>
> There are so many issues with depends on one data centers. The
> more critical now a days is weather cause lots of issues.
>
> I want to distribute freeswitch, minimum two data centers and two
> different regions.
>
> Does anyone have a PostgreSQL master-master replication working
> reliably? Anything I need to consider using PostgreSQL master-master.

Make sure that any autoincrement value is unique across all nodes. So it 
should not be possible that both node1 and node2 can generate the same 
autoincrement value. Triggers may casue trouble and row based locking is 
a good idea. How much of that applies to PostgreSQL you should be able 
to find by RTFM and searching on Google.

Regards,
Patrick


From vallimamod.abdullah at imtelecom.fr  Fri Aug  2 18:38:55 2013
From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH)
Date: Fri, 2 Aug 2013 16:38:55 +0200
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <CAGvuWYa6ptobw=9ey0uN5AatC-UBEidUxf+QbxDeBRWGFwWv7g@mail.gmail.com>
References: <CE200751.7876B%krice@freeswitch.org>
	<51FABAE9.8070709@millican.us> <51FB2387.10907@mst.edu>
	<51FBA224.7080207@puzzled.xs4all.nl>
	<CAGvuWYa6ptobw=9ey0uN5AatC-UBEidUxf+QbxDeBRWGFwWv7g@mail.gmail.com>
Message-ID: <64F41102-0629-4F70-A632-6C257E1F79D2@imtelecom.fr>

Hi,

On Aug 2, 2013, at 4:10 PM, Lloyd Aloysius <lloyd.aloysius at sunteltech.ca> wrote:

> 
> Does anyone have a PostgreSQL master-master replication working reliably? Anything I need to consider using PostgreSQL master-master.
> 

You may be interested in Postgres-xc (http://postgres-xc.sourceforge.net) it looks a very interesting multi-master cluster. I haven't tried it yet but I plan to do so soon.

Best Regards,
Vallimamod.
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From stuart.mills3 at btopenworld.com  Fri Aug  2 18:57:13 2013
From: stuart.mills3 at btopenworld.com (Stuart Mills)
Date: Fri, 2 Aug 2013 15:57:13 +0100
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>
Message-ID: <1138329210A14BE4AEE53D7419427A15@PBPC>

Hi Avi,

Thanks for your reply.

Yeah I'm sure, I'm using ESL and if I dial a single number it performs as expected, it?s literally only when I'm making 2 or more calls at once it happens.

Cheers,

Stuart

From: Avi Marcus 
Sent: Friday, August 02, 2013 3:24 PM
To: FreeSWITCH Users Help 
Subject: Re: [Freeswitch-users] Call at once and XML-CDR

Are you sure these aren't b leg things?
Do you have an example of your dial plan and the logs?

-Avi

On Aug 2, 2013 5:09 PM, "Stuart Mills" <stuart.mills3 at btopenworld.com> wrote:

  Hi,

  Got a question regarding call at once combined with xml-cdr logging.

  The problem I seem to get is when I try sending calls to 2 destinations and 1 call gets answered, so that wins the race, the xml-cdr logging seems to be a little confused. In the app_log I?m getting logging up to the point of the dial string, then nothing after that for the a-leg. What seems to happen is the rest of the app_log gets written to the cdr file of the bleg that answered the call first.

  Is this by design or a bug in the xml-cdr logging?

  Thanks,

  Stuart

  _________________________________________________________________________
  Professional FreeSWITCH Consulting Services:
  consulting at freeswitch.org
  http://www.freeswitchsolutions.com

  
  

  Official FreeSWITCH Sites
  http://www.freeswitch.org
  http://wiki.freeswitch.org
  http://www.cluecon.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




--------------------------------------------------------------------------------
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 lloyd.aloysius at sunteltech.ca  Fri Aug  2 20:44:04 2013
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Fri, 2 Aug 2013 12:44:04 -0400
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <1138329210A14BE4AEE53D7419427A15@PBPC>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>
	<1138329210A14BE4AEE53D7419427A15@PBPC>
Message-ID: <CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>

With the default configuration switch only log a_leg.

You need to enable b_leg in xml_cdr.conf.xml . Then you will see the two
records in b_leg, with one get billsec > 0 for the the answered call.

Lloyd



On Fri, Aug 2, 2013 at 10:57 AM, Stuart Mills <stuart.mills3 at btopenworld.com
> wrote:

>   Hi Avi,
>
> Thanks for your reply.
>
> Yeah I'm sure, I'm using ESL and if I dial a single number it performs as
> expected, it?s literally only when I'm making 2 or more calls at once it
> happens.
>
> Cheers,
>
> Stuart
>
>  *From:* Avi Marcus <avi at avimarcus.net>
> *Sent:* Friday, August 02, 2013 3:24 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Call at once and XML-CDR
>
>
> Are you sure these aren't b leg things?
> Do you have an example of your dial plan and the logs?
>
> -Avi
> On Aug 2, 2013 5:09 PM, "Stuart Mills" <stuart.mills3 at btopenworld.com>
> wrote:
>
>>   Hi,
>>
>> Got a question regarding call at once combined with xml-cdr logging.
>>
>> The problem I seem to get is when I try sending calls to 2 destinations
>> and 1 call gets answered, so that wins the race, the xml-cdr logging seems
>> to be a little confused. In the app_log I?m getting logging up to the point
>> of the dial string, then nothing after that for the a-leg. What seems to
>> happen is the rest of the app_log gets written to the cdr file of the bleg
>> that answered the call first.
>>
>> Is this by design or a bug in the xml-cdr logging?
>>
>> Thanks,
>>
>> Stuart
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>  ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lloyd.aloysius at sunteltech.ca  Fri Aug  2 20:45:54 2013
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Fri, 2 Aug 2013 12:45:54 -0400
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>
	<1138329210A14BE4AEE53D7419427A15@PBPC>
	<CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
Message-ID: <CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>

Another note  look into the caller_profile and index. I think you have two
indexes for this situation.In your logic insert the records for the
multiple indexes.

Lloyd
*
*
*  <http://www.sunteltech.ca/blog/>*


On Fri, Aug 2, 2013 at 12:44 PM, Lloyd Aloysius <
lloyd.aloysius at sunteltech.ca> wrote:

> With the default configuration switch only log a_leg.
>
> You need to enable b_leg in xml_cdr.conf.xml . Then you will see the two
> records in b_leg, with one get billsec > 0 for the the answered call.
>
> Lloyd
>
>
>
> On Fri, Aug 2, 2013 at 10:57 AM, Stuart Mills <
> stuart.mills3 at btopenworld.com> wrote:
>
>>   Hi Avi,
>>
>> Thanks for your reply.
>>
>> Yeah I'm sure, I'm using ESL and if I dial a single number it performs as
>> expected, it?s literally only when I'm making 2 or more calls at once it
>> happens.
>>
>> Cheers,
>>
>> Stuart
>>
>>  *From:* Avi Marcus <avi at avimarcus.net>
>> *Sent:* Friday, August 02, 2013 3:24 PM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] Call at once and XML-CDR
>>
>>
>> Are you sure these aren't b leg things?
>> Do you have an example of your dial plan and the logs?
>>
>> -Avi
>> On Aug 2, 2013 5:09 PM, "Stuart Mills" <stuart.mills3 at btopenworld.com>
>> wrote:
>>
>>>   Hi,
>>>
>>> Got a question regarding call at once combined with xml-cdr logging.
>>>
>>> The problem I seem to get is when I try sending calls to 2 destinations
>>> and 1 call gets answered, so that wins the race, the xml-cdr logging seems
>>> to be a little confused. In the app_log I?m getting logging up to the point
>>> of the dial string, then nothing after that for the a-leg. What seems to
>>> happen is the rest of the app_log gets written to the cdr file of the bleg
>>> that answered the call first.
>>>
>>> Is this by design or a bug in the xml-cdr logging?
>>>
>>> Thanks,
>>>
>>> Stuart
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>  ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 nneul at mst.edu  Fri Aug  2 20:51:05 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Fri, 02 Aug 2013 11:51:05 -0500
Subject: [Freeswitch-users] FreeSWITCH Core - Cassandra Database
In-Reply-To: <51FBA224.7080207@puzzled.xs4all.nl>
References: <CE200751.7876B%krice@freeswitch.org>
	<51FABAE9.8070709@millican.us> <51FB2387.10907@mst.edu>
	<51FBA224.7080207@puzzled.xs4all.nl>
Message-ID: <51FBE379.603@mst.edu>


> For the floating IP to follow to the new box, on a *2* node setup, you
> need to disable quorum and add a constraint in your pacemaker/corosync
> config so the resource (PerconaDB) follows the floating IP. Something like:
>
> $ pcs property set no-quorum-policy=ignore
> $ pcs constraint colocation add PerconaDB ClusterIP INFINITY
> $ pcs constraint order ClusterIP then PerconaDB

I had something along those lines w/ a three-node cluster. I think this was close to the last config I used:


node $id="1945540483" freesw-db1.srv.mst.edu
node $id="1962317699" freesw-db2.srv.mst.edu
node $id="1979094915" freesw-db3.srv.mst.edu \
	attributes standby="off"
primitive p_ip_mysql ocf:heartbeat:IPaddr2 \
	params nic="eth0:0" iflabel="mysql" ip="131.151.246.119"
primitive p_mysql ocf:heartbeat:mysql \
	params config="/local/mysql/mysql.conf" \
		pid="/local/mysql/data/mysqld.pid" \
		socket="/var/lib/mysql/mysql.sock" \
		binary="/local/mysql/server/bin/mysqld" \
		datadir="/local/mysql/data" \
		log="/local/mysql/data/log" \
		additional_parameters="--basedir=/local/mysql/server --log-error=/local/mysql/data/log" \
	op monitor interval="30s" timeout="30s" \
	op start interval="0" timeout="120s" \
	op stop interval="0" timeout="120s"

clone cl_mysql p_mysql \
	meta interleave="true"

colocation c_ip_mysql inf: p_ip_mysql cl_mysql

property $id="cib-bootstrap-options" \
	dc-version="1.1.7-2.fc17-ee0730e13d124c3d58f00016c3376a1de5323cff" \
	cluster-infrastructure="corosync" \
	stonith-enabled="false"

rsc_defaults $id="rsc-options" \
	resource-stickiness="100"





> I'm curious, did you use a shared filesystem (something like DRBD with
> GFS2/OCFS2) and regular Percona XtraDB on both nodes or did you use
> Percona XtraDB Cluster?

Percona XtraDB Cluster.

The symptom/problem I ran into - aside from Corosync/PM being excessively complex for the situation - was that a DB 
instance failed, and despite the others being online and the cluster itself being up and working - it didn't fail the IP 
over. It may have been a glitch - didn't see it during testing - but just more trouble than it was worth for the limited 
use case.

-- Nathan

------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From andretodd at verizon.net  Fri Aug  2 21:20:13 2013
From: andretodd at verizon.net (Andre)
Date: Fri, 02 Aug 2013 13:20:13 -0400
Subject: [Freeswitch-users] Limit Database
Message-ID: <04f701ce8fa4$8d42d920$a7c88b60$@verizon.net>

HI, on Windows I'm trying to use Limit  with Microsoft SQL DB but it's still
using the sqlight database.

I have mod_Db and Mod_Hash loaded and under db.config.xml I have my odbc
setup like this 
<param name="odbc-dsn" value="sqldb64::" />  

I know the odbc works because I have the core database setup the same
<param name="odbc-dsn" value="sqldb64::" />  
   <param name="core-dbtype" value="MSSQL"/>  
 
Is there a step that I'm missing?
Thanks



From krice at freeswitch.org  Fri Aug  2 21:36:56 2013
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 02 Aug 2013 12:36:56 -0500
Subject: [Freeswitch-users] Limit Database
In-Reply-To: <04f701ce8fa4$8d42d920$a7c88b60$@verizon.net>
Message-ID: <CE215868.78885%krice@freeswitch.org>

Have you tried the odbd://DSN:: syntax?


On 8/2/13 12:20 PM, "Andre" <andretodd at verizon.net> wrote:

> HI, on Windows I'm trying to use Limit  with Microsoft SQL DB but it's still
> using the sqlight database.
> 
> I have mod_Db and Mod_Hash loaded and under db.config.xml I have my odbc
> setup like this 
> <param name="odbc-dsn" value="sqldb64::" />
> 
> I know the odbc works because I have the core database setup the same
> <param name="odbc-dsn" value="sqldb64::" />
>    <param name="core-dbtype" value="MSSQL"/>
>  
> Is there a step that I'm missing?
> Thanks
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From andretodd at verizon.net  Fri Aug  2 22:00:56 2013
From: andretodd at verizon.net (Andre)
Date: Fri, 02 Aug 2013 14:00:56 -0400
Subject: [Freeswitch-users] Limit Database
In-Reply-To: <CE215868.78885%krice@freeswitch.org>
References: <04f701ce8fa4$8d42d920$a7c88b60$@verizon.net>
	<CE215868.78885%krice@freeswitch.org>
Message-ID: <051701ce8faa$3d708680$b8519380$@verizon.net>

Just did. No change

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Friday, August 2, 2013 1:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Limit Database

Have you tried the odbd://DSN:: syntax?


On 8/2/13 12:20 PM, "Andre" <andretodd at verizon.net> wrote:

> HI, on Windows I'm trying to use Limit  with Microsoft SQL DB but it's 
> still using the sqlight database.
> 
> I have mod_Db and Mod_Hash loaded and under db.config.xml I have my 
> odbc setup like this <param name="odbc-dsn" value="sqldb64::" />
> 
> I know the odbc works because I have the core database setup the same 
> <param name="odbc-dsn" value="sqldb64::" />
>    <param name="core-dbtype" value="MSSQL"/>
>  
> Is there a step that I'm missing?
> Thanks
> 
> 
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
>  
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-use
> rs
> http://www.freeswitch.org

--
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 bdfoster at davri.com  Fri Aug  2 22:22:29 2013
From: bdfoster at davri.com (Brian Foster)
Date: Fri, 2 Aug 2013 14:22:29 -0400
Subject: [Freeswitch-users] Can not make out going call from sangoma A200
In-Reply-To: <BLU171-W3672C8625546F122FFC013B5510@phx.gbl>
References: <BLU171-W3672C8625546F122FFC013B5510@phx.gbl>
Message-ID: <CAHgk4BXX1nArMXv-T5CNzLbjh_wyn0fjzYNOUCaAPsUE14KNdA@mail.gmail.com>

That's the expected behavior if 2:1 is in use. So that means you use
another port to place your outbound call. Tips on how to do that, however,
I can't really produce. Hopefully someone with more experience with FreeTDM
can chime in.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 2, 2013 3:09 AM, "James zhu" <zhulizhong at live.com> wrote:

> hello:
> I installed freeswitch and sangoma A200 with 2 FXOs. i set the tone group
> to CN
> and opermode=china. the A200 can make accept incoming calls from
> FreeTDM/2:1 , but I use same port to make out going port, the shows the
> port is CONGESTION.
> --------------------------system
> log-----------------------------------------------
> 2013-08-02 14:36:42.003531 [DEBUG] mod_freetdm.c:640 [2:1]
> FreeTDM/2:1/13811737925 CHANNEL HANGUP EXIT
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:48
> FreeTDM/2:1/13811737925 Standard HANGUP, cause: NORMAL_CIRCUIT_CONGESTION
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:679
> (FreeTDM/2:1/13811737925) State HANGUP going to sleep
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:692
> (FreeTDM/2:1/13811737925) Callstate Change DOWN -> HANGUP
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:447
> (FreeTDM/2:1/13811737925) State Change CS_HANGUP -> CS_REPORTING
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal
> FreeTDM/2:1/13811737925 [BREAK]
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:416
> (FreeTDM/2:1/13811737925) Running State Change CS_REPORTING
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764
> (FreeTDM/2:1/13811737925) State REPORTING
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:92
> FreeTDM/2:1/13811737925 Standard REPORTING, cause: NORMAL_CIRCUIT_CONGESTION
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764
> (FreeTDM/2:1/13811737925) State REPORTING going to sleep
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:441
> (FreeTDM/2:1/13811737925) State Change CS_REPORTING -> CS_DESTROY
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal
> FreeTDM/2:1/13811737925 [BREAK]
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1549 Session 37
> (FreeTDM/2:1/13811737925) Locked, Waiting on external entities
> 2013-08-02 14:36:42.023531 [DEBUG] switch_ivr_originate.c:3632 Originate
> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
> 2013-08-02 14:36:42.023531 [INFO] mod_dptools.c:3192 Originate Failed.
>  Cause: NORMAL_CIRCUIT_CONGESTION
> 2013-08-02 14:36:42.023531 [NOTICE] switch_channel.c:4593 Hangup
> sofia/internal/1008 at 192.168.0.173 [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]
> 2013-08-02 14:36:42.023531 [DEBUG] switch_channel.c:3135 Send signal
> sofia/internal/1008 at 192.168.0.173 [KILL]
> 2013-08-02 14:36:42.023531 [DEBUG] switch_core_session.c:1341 Send signal
> sofia/internal
> ---------------------------------------------
> the FS is download from git and wanpipe is latest version.  please give a
> help for that issue.
>
> Best regards,
> James.zhu
> website: www.hiastar.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 karl at xtronics.com  Fri Aug  2 22:42:32 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Fri, 02 Aug 2013 13:42:32 -0500
Subject: [Freeswitch-users] best way to receive FAXs
In-Reply-To: <CE215868.78885%krice@freeswitch.org>
References: <CE215868.78885%krice@freeswitch.org>
Message-ID: <51FBFD98.9030909@xtronics.com>

Back in 1990's I started using a fax receiving programs (originally I found several solutions on 
Windoze that were terrible).  When I moved our server to Linux (2000) I discovered hylafax. With 
hylasfax and a particular modem all has worked flawlessly for years. In the process I learned way 
more about fax than I ever wanted to.

Now as I set up this freeswitch system, I see I can use t38modem as an end point.  So my question is 
has anyone been down this road with t38modem and hylafax?  Or should I just use a FAX service? Do 
these services work reliably with ALL the machines out there? (I'm doubtful)

FAX should soon disappear - but we still have customers in other countries that send in the 
occasional fax. The issue is that different fax machines from different times have buggy software 
that makes receiving faxes reliably a can of worms. What I have works, but I would like to get rid 
of the POTS phone-line.


--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

Life does not consist mainly, or even largely, of facts and
happenings. It consists mainly of the storm of thought that is
forever flowing through one's head.--Mark Twain
--------------------------------------------------------------------------------


From pasha at prosperity4ever.com  Fri Aug  2 23:52:43 2013
From: pasha at prosperity4ever.com (Paul)
Date: Fri, 02 Aug 2013 19:45:43 -0007
Subject: [Freeswitch-users] No audio on either of an established call
In-Reply-To: <CABkS=xrgKqPkKjP2NjWMAefxOzdRTe3=JK18Xj6b7EQZJ6_b3Q@mail.gmail.com>
References: <CABkS=xpqqwJBd=cHGzz_aj3r8kvuQssP4HcrYCyCj9oP5EWLTg@mail.gmail.com>
	<CABkS=xqFknFok1NWgoB+tBhrDB+GzrQNA-wA+CjnftpmSmuFHA@mail.gmail.com>
	<20130801203132.0612157E002@mail.mydcs.ca>
	<CABkS=xrgKqPkKjP2NjWMAefxOzdRTe3=JK18Xj6b7EQZJ6_b3Q@mail.gmail.com>
Message-ID: <20130802195244.B559157E001@mail.mydcs.ca>

I'm sorry, it looks like I missed the part where you mentioned that you 
have it all working now, that's why I was asking about where you router 
is because it very much sounded like routing/nat issues you were having 
:)

Paul

On Thu, 1 Aug, 2013 at 6:34 PM, Peter <eidevm5 at gmail.com> wrote:
> Correct.? Both Kamailio servers are on different networks.?? I 
> didn't show the router in the mix for simplicity.
> 
> 
> On Fri, Aug 2, 2013 at 6:31 AM, Paul <pasha at prosperity4ever.com> 
> wrote:
>> Where on your diagram is the Router? I assume that your Kamailio 1 
>> and Kamailio 2 are on two different networks?
>> 
>> Paul
>> 
>> 
>> On Wed, 31 Jul, 2013 at 10:12 PM, Peter <eidevm5 at gmail.com> wrote:
>>> After checking everything for the umpteenth time, I discovered I 
>>> was missing a route back to 10.1.254.0/24
>>> 
>>> Once I put that in, bingo, audio!
>>> 
>>> 
>>> On Thu, Aug 1, 2013 at 12:33 PM, Peter <eidevm5 at gmail.com> wrote:
>>>> I currently have 2 SIP clients (Linphone) successfully calling 
>>>> each other, but there is no audio on either end.
>>>> 
>>>> The set up is as follows:
>>>> 
>>>> 
>>>> 
>>>> Linphone1 (1000)? --> Kamailio 1?? <-------> Freeswitch?? 
>>>> <------>? Kamalio 2? <---- Linphone2 (2000)
>>>> 
>>>> 
>>>> Using Freeswitch 1.2.12 on CentOS (installed via RPM)
>>>> 
>>>> Freeswitch has two interfaces:
>>>> 
>>>> external - 10.1.1.206
>>>> internal -? 10.10.10.206
>>>> 
>>>> Each of the Linphone clients are registered their respective 
>>>> Kamailio instance and Kamailio is configured to route via the 
>>>> appropriate interface on Freeswitch.
>>>> 
>>>> The SIP negotiation is working as I can call either Linphone 
>>>> client.?? 
>>>> 
>>>> I've done a tcpdump on each side of? Freeswitch and can see the 
>>>> RTP traffic between the Linphone and the appropriate interface on 
>>>> Freeswitch.
>>>> 
>>>> I've tried different codec combinations (mostly G711 and iLBC), 
>>>> and different SIP clients but still get no audio.?? 
>>>> 
>>>> Any pointers on how to track down the issue?
>>>> 
>>>> Thanks
>>>> 
>>>> Peter
>>>> 
>>>> 
>>> 
>>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 pasha at prosperity4ever.com  Fri Aug  2 23:56:10 2013
From: pasha at prosperity4ever.com (Paul)
Date: Fri, 02 Aug 2013 19:49:10 -0007
Subject: [Freeswitch-users] FreeSWITCH -- 30 Second call drop
In-Reply-To: <CAAYhoX+f9VcpnOde0-bAbcACpz2t0otnsP1zhsiEg+ZMjjxFGg@mail.gmail.com>
References: <20130801204721.GC20191@0rdior.com>
	<20130801211304.35C9957E002@mail.mydcs.ca>
	<CAAYhoX+f9VcpnOde0-bAbcACpz2t0otnsP1zhsiEg+ZMjjxFGg@mail.gmail.com>
Message-ID: <20130802195612.5771257E001@mail.mydcs.ca>

That's a great piece of information to have as well. I do have my 
router not responding to ICMP requests, and I was noticing some errors 
in regards to ICMP unreachable packets while I was having issues, but 
for me setting the correct external ip for freeswitch and natting it 
correctly solved my audio issues. Not sure if the ICMP errors are still 
there or not, they might be, this is making me go back and look :)

Paul

On Thu, 1 Aug, 2013 at 6:29 PM, Carlos Flor <jackal at cybershroud.net> 
wrote:
> If that doesn't work, check your mtu and/or make sure you aren't 
> blocking icmp on your firewall if you want to do path-mtu discovery. 
> ?I had a 31 second call drop problem and it was because my icmp 
> unreachable (needs fragmentation) packets weren't getting all the way 
> back to me (in my case it was a bad route) and so I my device wasn't 
> adjusting my mtu and the call would drop exactly at 31 seconds in 
> every single time.
> 
> 
> On Thu, Aug 1, 2013 at 5:13 PM, Paul <pasha at prosperity4ever.com> 
> wrote:
>> 
>> 
>> 
>>> Hi Paul,
>>> 
>>> I saw you emails regarding 30 seconds call drop. I have exactly the
>>> same issue when I want to connect my client over OpenVPN to my FS
>>> server, which is also the vpn server at the same time (server is
>>> on a public IP). 
>>> 
>>> All my calls are dropped after 32 seconds... very annoying and I 
>>> can't
>>> find the solution. So I'm constantly switching config for standard 
>>> vs vpn 
>>> for my client and FS (the standard config is just connecting to the
>>> public ip which is unencrypted).
>>> 
>>> Did you find a solution in the mean time? Please inform me, cause 
>>> it's
>>> really killing me. :=|
>>> 
>>> I hope you can inform me on a short term; really appreciated!
>>> 
>>> -- 
>>> Yours sincerely, 
>>> E.J.C. Lindner
>>> 
>> Yes I found a solution to my problem.
>> 
>> My case differs a little bit from yours in that my openvpn server is 
>> on the router, not on the PBX, in your case it's probably even a 
>> little simpler.
>> 
>> So here is what I can advise you,
>> 
>> 1. Ensure NAT is working properly over openVPN, so to test that all 
>> you need to do is have 2 phones on the same PBX able to call each 
>> other internally (extension 100 to extension 101 for instance).
>> 
>> If that works then your VPN NAT is probably good.
>> 
>> 2. Ensure that your PBX is reporting the correct external address to 
>> your vendor upstream (this is where it was messing me up).
>> 
>> For example, for me my PBX was 10.0.0.40 (itnernal address) which 
>> the router was NATing to 24.38.231.4 (just a sample external 
>> address) and the same on the way back, when a request would come in 
>> for 24.38.231 .4 it would translate it to 10.0.0.40.
>> 
>> Because my PBX was not aware of 24.38.231.4 as it's external 
>> address, it was only listening on 10.0.0.40, it wasn't responding to 
>> the SIP ACK packets correctly, and the call was being perceived as 
>> finished and therefore hung up.
>> 
>> To fix it in FreeSwitch I had to edit external.xml and insert the 
>> proper IP address for sip_ext_ip and rtp_ext_ip (this is from memory)
>> 
>> I hope this helps you out cuz it cost me hours of frustration :)
>> 
>> Paul
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 bdfoster at davri.com  Sat Aug  3 00:02:54 2013
From: bdfoster at davri.com (Brian Foster)
Date: Fri, 2 Aug 2013 16:02:54 -0400
Subject: [Freeswitch-users] best way to receive FAXs
In-Reply-To: <51FBFD98.9030909@xtronics.com>
References: <CE215868.78885%krice@freeswitch.org>
	<51FBFD98.9030909@xtronics.com>
Message-ID: <CAHgk4BVRfHcSsy5LV5Xx8Auu+-EBLW+7xHUG4x=fgsgJBpws5A@mail.gmail.com>

I've personally had issues with hylafax. I dont know what it is or how to
fix it, but what I ended up doing is using mod_spandsp and dumping the
received faxes into a couchdb database. I have an example here (
http://github.com/bdfoster/fax-to-couchdb). It's rough, but it might give
you some ideas.

Take care.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 2, 2013 2:49 PM, "Karl Schmidt" <karl at xtronics.com> wrote:

> Back in 1990's I started using a fax receiving programs (originally I
> found several solutions on
> Windoze that were terrible).  When I moved our server to Linux (2000) I
> discovered hylafax. With
> hylasfax and a particular modem all has worked flawlessly for years. In
> the process I learned way
> more about fax than I ever wanted to.
>
> Now as I set up this freeswitch system, I see I can use t38modem as an end
> point.  So my question is
> has anyone been down this road with t38modem and hylafax?  Or should I
> just use a FAX service? Do
> these services work reliably with ALL the machines out there? (I'm
> doubtful)
>
> FAX should soon disappear - but we still have customers in other countries
> that send in the
> occasional fax. The issue is that different fax machines from different
> times have buggy software
> that makes receiving faxes reliably a can of worms. What I have works, but
> I would like to get rid
> of the POTS phone-line.
>
>
>
> --------------------------------------------------------------------------------
> Karl Schmidt                                  EMail Karl at xtronics.com
> Transtronics, Inc.                              WEB
> http://secure.transtronics.com
> 3209 West 9th Street                             Ph (785) 841-3089
> Lawrence, KS 66049                              FAX (785) 841-0434
>
> Life does not consist mainly, or even largely, of facts and
> happenings. It consists mainly of the storm of thought that is
> forever flowing through one's head.--Mark Twain
>
> --------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 karl at xtronics.com  Sat Aug  3 00:58:52 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Fri, 02 Aug 2013 15:58:52 -0500
Subject: [Freeswitch-users] [OL]Re:  best way to receive FAXs
In-Reply-To: <CAHgk4BVRfHcSsy5LV5Xx8Auu+-EBLW+7xHUG4x=fgsgJBpws5A@mail.gmail.com>
References: <CE215868.78885%krice@freeswitch.org>
	<51FBFD98.9030909@xtronics.com>
	<CAHgk4BVRfHcSsy5LV5Xx8Auu+-EBLW+7xHUG4x=fgsgJBpws5A@mail.gmail.com>
Message-ID: <51FC1D8C.4020204@xtronics.com>

On 08/02/2013 03:02 PM, Brian Foster wrote:
> I've personally had issues with hylafax. I dont know what it is or how to fix it, but what I ended
> up doing is using mod_spandsp and dumping the received faxes into a couchdb database. I have an
> example here (http://github.com/bdfoster/fax-to-couchdb). It's rough, but it might give you some ideas.

Thanks - I had to go through 6 modems before I found one that did fax reliably - it is the firmware 
code in the modem that matters - and all of the big names failed. Worked with an external Supra Fax 
modem 144LC

If you ever need to get it to work, I can send my Debian configs.

--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

Most economic fallacies derive from the tendency to assume that there is a fixed pie, that one party 
can gain only at the expense of another.
Milton Friedman
--------------------------------------------------------------------------------


From nneul at mst.edu  Sat Aug  3 01:36:14 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Fri, 02 Aug 2013 16:36:14 -0500
Subject: [Freeswitch-users] Best practices for memory pool usage in modules?
Message-ID: <51FC264E.2050500@mst.edu>

I'm looking into memory leak issues with mod_skinny, and it seems clear that it's repeatedly allocating memory when it 
doesn't really need to hang onto it permanently.

Since it has both a long standing connection to each device (listener) and has requests coming in regularly that 
currently are allocating memory and appear to be leaking it since the listener stays around.

Would having the code allocate a new temporary sub-pool for each request that comes in be appropriate, or would it be 
better to just use regular malloc for those allocate/use/destroy cases?

-- Nathan

------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From schoch+freeswitch.org at xwin32.com  Sat Aug  3 01:43:25 2013
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Fri, 2 Aug 2013 14:43:25 -0700
Subject: [Freeswitch-users] best way to receive FAXs
In-Reply-To: <51FBFD98.9030909@xtronics.com>
References: <CE215868.78885%krice@freeswitch.org>
	<51FBFD98.9030909@xtronics.com>
Message-ID: <CAOkuKdcWKfDwP+fGxfpZK3UT0x+pGJn17+5TVCu8EKQdtnBZjw@mail.gmail.com>

On Fri, Aug 2, 2013 at 11:42 AM, Karl Schmidt <karl at xtronics.com> wrote:

> What I have works, but I would like to get rid
> of the POTS phone-line.
>

When we switched to VoIP with FreeSWITCH here, we made a decision to keep
one POTS line as a backup. We got the cheapest service we could find (from
AT&T). We get value for our money because we use it as:

   1. Our incoming FAX line (using HylaFAX with a Best Data Smart One 56SK
   modem from the late 1990's that has worked perfectly for years);
   2. A backup DSL line, plugged into a dual-WAN router. (Our primary
   Internet is Comcast.)
   3. A POTS line for our alarm system, as highly recommended by our
   security company; and
   4. A backup phone line, in case a power failure is longer than the UPS
   run time. (This happened during the California electricity crisis of 2000
   and 2001.)

We can keep the cost low because other than the alarm system calling a
toll-free number, no outgoing calls are made from this line.

-- 
Steve
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From stuart.mills3 at btopenworld.com  Sat Aug  3 02:24:18 2013
From: stuart.mills3 at btopenworld.com (Stuart Mills)
Date: Fri, 2 Aug 2013 23:24:18 +0100
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC><000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com><1138329210A14BE4AEE53D7419427A15@PBPC><CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
	<CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
Message-ID: <3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>

Hi,

Thanks for your reply.

I have already changed the parameter log-b-leg to true in xml_cdr.conf and I am getting 2 records for the outbound calls and 1 for inbound. The issue is that the app_log section in the a_leg cdr seems to be split across the a_leg cdr and the b_leg cdr that ?won? the race to be answered first. The second b_leg has no duration as expected.

Am I making sense?

Regards,

Stuart 

From: Lloyd Aloysius 
Sent: Friday, August 02, 2013 5:45 PM
To: FreeSWITCH Users Help 
Subject: Re: [Freeswitch-users] Call at once and XML-CDR

Another note  look into the caller_profile and index. I think you have two indexes for this situation.In your logic insert the records for the multiple indexes. 


Lloyd







On Fri, Aug 2, 2013 at 12:44 PM, Lloyd Aloysius <lloyd.aloysius at sunteltech.ca> wrote:

  With the default configuration switch only log a_leg. 


  You need to enable b_leg in xml_cdr.conf.xml . Then you will see the two records in b_leg, with one get billsec > 0 for the the answered call. 


  Lloyd






  On Fri, Aug 2, 2013 at 10:57 AM, Stuart Mills <stuart.mills3 at btopenworld.com> wrote:

    Hi Avi,

    Thanks for your reply.

    Yeah I'm sure, I'm using ESL and if I dial a single number it performs as expected, it?s literally only when I'm making 2 or more calls at once it happens.

    Cheers,

    Stuart

    From: Avi Marcus 
    Sent: Friday, August 02, 2013 3:24 PM
    To: FreeSWITCH Users Help 
    Subject: Re: [Freeswitch-users] Call at once and XML-CDR

    Are you sure these aren't b leg things?
    Do you have an example of your dial plan and the logs?

    -Avi

    On Aug 2, 2013 5:09 PM, "Stuart Mills" <stuart.mills3 at btopenworld.com> wrote:

      Hi,

      Got a question regarding call at once combined with xml-cdr logging.

      The problem I seem to get is when I try sending calls to 2 destinations and 1 call gets answered, so that wins the race, the xml-cdr logging seems to be a little confused. In the app_log I?m getting logging up to the point of the dial string, then nothing after that for the a-leg. What seems to happen is the rest of the app_log gets written to the cdr file of the bleg that answered the call first.

      Is this by design or a bug in the xml-cdr logging?

      Thanks,

      Stuart

      _________________________________________________________________________
      Professional FreeSWITCH Consulting Services:
      consulting at freeswitch.org
      http://www.freeswitchsolutions.com

      
      

      Official FreeSWITCH Sites
      http://www.freeswitch.org
      http://wiki.freeswitch.org
      http://www.cluecon.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



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

    _________________________________________________________________________
    Professional FreeSWITCH Consulting Services:
    consulting at freeswitch.org
    http://www.freeswitchsolutions.com

    
    

    Official FreeSWITCH Sites
    http://www.freeswitch.org
    http://wiki.freeswitch.org
    http://www.cluecon.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


    _________________________________________________________________________
    Professional FreeSWITCH Consulting Services:
    consulting at freeswitch.org
    http://www.freeswitchsolutions.com

    
    

    Official FreeSWITCH Sites
    http://www.freeswitch.org
    http://wiki.freeswitch.org
    http://www.cluecon.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






--------------------------------------------------------------------------------
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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  Sat Aug  3 02:46:51 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 2 Aug 2013 17:46:51 -0500
Subject: [Freeswitch-users] Best practices for memory pool usage in
	modules?
In-Reply-To: <51FC264E.2050500@mst.edu>
References: <51FC264E.2050500@mst.edu>
Message-ID: <CAKbxfG-zeQRvP+yYzMs3QVm+HcmKB_2H=F+CHWhpQyZH3m=f_Q@mail.gmail.com>

If the object has a complex life cycle then pools are better.  If its short
lived frequent little one-off allocations may as well malloc
On Aug 2, 2013 4:40 PM, "Nathan Neulinger" <nneul at mst.edu> wrote:

> I'm looking into memory leak issues with mod_skinny, and it seems clear
> that it's repeatedly allocating memory when it
> doesn't really need to hang onto it permanently.
>
> Since it has both a long standing connection to each device (listener) and
> has requests coming in regularly that
> currently are allocating memory and appear to be leaking it since the
> listener stays around.
>
> Would having the code allocate a new temporary sub-pool for each request
> that comes in be appropriate, or would it be
> better to just use regular malloc for those allocate/use/destroy cases?
>
> -- Nathan
>
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nneul at mst.edu  Sat Aug  3 02:51:54 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Fri, 02 Aug 2013 17:51:54 -0500
Subject: [Freeswitch-users] Best practices for memory pool usage in
	modules?
In-Reply-To: <CAKbxfG-zeQRvP+yYzMs3QVm+HcmKB_2H=F+CHWhpQyZH3m=f_Q@mail.gmail.com>
References: <51FC264E.2050500@mst.edu>
	<CAKbxfG-zeQRvP+yYzMs3QVm+HcmKB_2H=F+CHWhpQyZH3m=f_Q@mail.gmail.com>
Message-ID: <51FC380A.4030303@mst.edu>

In almost all of these cases, it's "allocate space to package up a message/pdu then send it over the socket", so it 
sounds like a simple malloc makes more sense.

-- Nathan

On 08/02/2013 05:46 PM, Anthony Minessale wrote:
> If the object has a complex life cycle then pools are better.  If its short lived frequent little one-off allocations
> may as well malloc
>
> On Aug 2, 2013 4:40 PM, "Nathan Neulinger" <nneul at mst.edu <mailto:nneul at mst.edu>> wrote:
>
>     I'm looking into memory leak issues with mod_skinny, and it seems clear that it's repeatedly allocating memory when it
>     doesn't really need to hang onto it permanently.
>
>     Since it has both a long standing connection to each device (listener) and has requests coming in regularly that
>     currently are allocating memory and appear to be leaking it since the listener stays around.
>
>     Would having the code allocate a new temporary sub-pool for each request that comes in be appropriate, or would it be
>     better to just use regular malloc for those allocate/use/destroy cases?
>
>     -- Nathan
>
>     ------------------------------------------------------------
>     Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu>
>     Missouri S&T Information Technology (573) 612-1412 <tel:%28573%29%20612-1412>
>     System Administrator - Architect
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From stuart.mills3 at btopenworld.com  Sat Aug  3 03:17:38 2013
From: stuart.mills3 at btopenworld.com (Stuart Mills)
Date: Sat, 3 Aug 2013 00:17:38 +0100
Subject: [Freeswitch-users] att_xfer origination_cancel_key
Message-ID: <7EA4E9B1D66C4202A60862B65D251EAB@PBPC>

Hi,

I have noticed that att_xfer has a configurable cancel key to stop the transfer mid dial, is there an option to change keys for the other transfer features? 

0 is conference all 3 parties at the moment, but I'd like to designate a 4 or some other key, is this possible?

Many Thanks,

Stuart Mills
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From zhulizhong at live.com  Sat Aug  3 06:32:10 2013
From: zhulizhong at live.com (James zhu)
Date: Sat, 3 Aug 2013 02:32:10 +0000
Subject: [Freeswitch-users] Can not make out going call from sangoma A200
In-Reply-To: <CAHgk4BXX1nArMXv-T5CNzLbjh_wyn0fjzYNOUCaAPsUE14KNdA@mail.gmail.com>
References: <BLU171-W3672C8625546F122FFC013B5510@phx.gbl>,
	<CAHgk4BXX1nArMXv-T5CNzLbjh_wyn0fjzYNOUCaAPsUE14KNdA@mail.gmail.com>
Message-ID: <BLU171-W58FF5C141AECC333E5BD0FB5520@phx.gbl>

thanks, Brian.  actually the 2:1 is idle because the port can accept incoming call. afterI reload freetdm or restart FS, I still can not make outgoing calls. If the port physicallyfailed, how come I can make incoming call use the same port. confused. thanks again.
Best regards,

James.zhu

website: www.hiastar.com

Date: Fri, 2 Aug 2013 14:22:29 -0400
From: bdfoster at davri.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can not make out going call from sangoma A200

That's the expected behavior if 2:1 is in use. So that means you use another port to place your outbound call. Tips on how to do that, however, I can't really produce. Hopefully someone with more experience with FreeTDM can chime in.

Thank you,
Brian Foster

Project Manager/Owner's Rep.

Davri Investments, Inc.

O: 317-787-2686 x2102

M: 317-600-9753

E: bdfoster at davri.com

Indianapolis, Indiana
Sent from a mobile device.
On Aug 2, 2013 3:09 AM, "James zhu" <zhulizhong at live.com> wrote:




hello:I installed freeswitch and sangoma A200 with 2 FXOs. i set the tone group to CNand opermode=china. the A200 can make accept incoming calls from FreeTDM/2:1 , but I use same port to make out going port, the shows the port is CONGESTION.
--------------------------system log-----------------------------------------------2013-08-02 14:36:42.003531 [DEBUG] mod_freetdm.c:640 [2:1] FreeTDM/2:1/13811737925 CHANNEL HANGUP EXIT
2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:48 FreeTDM/2:1/13811737925 Standard HANGUP, cause: NORMAL_CIRCUIT_CONGESTION2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:679 (FreeTDM/2:1/13811737925) State HANGUP going to sleep
2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:692 (FreeTDM/2:1/13811737925) Callstate Change DOWN -> HANGUP2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:447 (FreeTDM/2:1/13811737925) State Change CS_HANGUP -> CS_REPORTING
2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925) Running State Change CS_REPORTING
2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State REPORTING2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:92 FreeTDM/2:1/13811737925 Standard REPORTING, cause: NORMAL_CIRCUIT_CONGESTION
2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State REPORTING going to sleep2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:441 (FreeTDM/2:1/13811737925) State Change CS_REPORTING -> CS_DESTROY
2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1549 Session 37 (FreeTDM/2:1/13811737925) Locked, Waiting on external entities
2013-08-02 14:36:42.023531 [DEBUG] switch_ivr_originate.c:3632 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]2013-08-02 14:36:42.023531 [INFO] mod_dptools.c:3192 Originate Failed.  Cause: NORMAL_CIRCUIT_CONGESTION
2013-08-02 14:36:42.023531 [NOTICE] switch_channel.c:4593 Hangup sofia/internal/1008 at 192.168.0.173 [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]2013-08-02 14:36:42.023531 [DEBUG] switch_channel.c:3135 Send signal sofia/internal/1008 at 192.168.0.173 [KILL]
2013-08-02 14:36:42.023531 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal---------------------------------------------the FS is download from git and wanpipe is latest version.  please give a help for that issue.

Best regards,

James.zhu

website: www.hiastar.com 		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.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




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 bdfoster at davri.com  Sat Aug  3 07:03:47 2013
From: bdfoster at davri.com (Brian Foster)
Date: Fri, 2 Aug 2013 23:03:47 -0400
Subject: [Freeswitch-users] Can not make out going call from sangoma A200
In-Reply-To: <BLU171-W58FF5C141AECC333E5BD0FB5520@phx.gbl>
References: <BLU171-W3672C8625546F122FFC013B5510@phx.gbl>
	<CAHgk4BXX1nArMXv-T5CNzLbjh_wyn0fjzYNOUCaAPsUE14KNdA@mail.gmail.com>
	<BLU171-W58FF5C141AECC333E5BD0FB5520@phx.gbl>
Message-ID: <CAHgk4BVdu6Dk3P=rrRBKquiydLUUGBUxSHauTv73tx=XPq=pjA@mail.gmail.com>

I apologize, I misread your email. Hopefully someone who knows FreeTDM will
come around. Have you contacted Sangoma? They are the maintainers of
FreeTDM as far as I know. You might try that route. They are very familiar
with freeswitch im sure :).

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 2, 2013 10:39 PM, "James zhu" <zhulizhong at live.com> wrote:

> thanks, Brian.  actually the 2:1 is idle because the port can accept
> incoming call. after
> I reload freetdm or restart FS, I still can not make outgoing calls. If
> the port physically
> failed, how come I can make incoming call use the same port. confused.
> thanks again.
>
> Best regards,
> James.zhu
> website: www.hiastar.com
>
> ------------------------------
> Date: Fri, 2 Aug 2013 14:22:29 -0400
> From: bdfoster at davri.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Can not make out going call from sangoma
> A200
>
> That's the expected behavior if 2:1 is in use. So that means you use
> another port to place your outbound call. Tips on how to do that, however,
> I can't really produce. Hopefully someone with more experience with FreeTDM
> can chime in.
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana
>
> Sent from a mobile device.
> On Aug 2, 2013 3:09 AM, "James zhu" <zhulizhong at live.com> wrote:
>
> hello:
> I installed freeswitch and sangoma A200 with 2 FXOs. i set the tone group
> to CN
> and opermode=china. the A200 can make accept incoming calls from
> FreeTDM/2:1 , but I use same port to make out going port, the shows the
> port is CONGESTION.
> --------------------------system
> log-----------------------------------------------
> 2013-08-02 14:36:42.003531 [DEBUG] mod_freetdm.c:640 [2:1]
> FreeTDM/2:1/13811737925 CHANNEL HANGUP EXIT
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:48
> FreeTDM/2:1/13811737925 Standard HANGUP, cause: NORMAL_CIRCUIT_CONGESTION
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:679
> (FreeTDM/2:1/13811737925) State HANGUP going to sleep
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:692
> (FreeTDM/2:1/13811737925) Callstate Change DOWN -> HANGUP
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:447
> (FreeTDM/2:1/13811737925) State Change CS_HANGUP -> CS_REPORTING
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal
> FreeTDM/2:1/13811737925 [BREAK]
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:416
> (FreeTDM/2:1/13811737925) Running State Change CS_REPORTING
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764
> (FreeTDM/2:1/13811737925) State REPORTING
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:92
> FreeTDM/2:1/13811737925 Standard REPORTING, cause: NORMAL_CIRCUIT_CONGESTION
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764
> (FreeTDM/2:1/13811737925) State REPORTING going to sleep
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:441
> (FreeTDM/2:1/13811737925) State Change CS_REPORTING -> CS_DESTROY
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal
> FreeTDM/2:1/13811737925 [BREAK]
> 2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1549 Session 37
> (FreeTDM/2:1/13811737925) Locked, Waiting on external entities
> 2013-08-02 14:36:42.023531 [DEBUG] switch_ivr_originate.c:3632 Originate
> Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
> 2013-08-02 14:36:42.023531 [INFO] mod_dptools.c:3192 Originate Failed.
>  Cause: NORMAL_CIRCUIT_CONGESTION
> 2013-08-02 14:36:42.023531 [NOTICE] switch_channel.c:4593 Hangup
> sofia/internal/1008 at 192.168.0.173 [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]
> 2013-08-02 14:36:42.023531 [DEBUG] switch_channel.c:3135 Send signal
> sofia/internal/1008 at 192.168.0.173 [KILL]
> 2013-08-02 14:36:42.023531 [DEBUG] switch_core_session.c:1341 Send signal
> sofia/internal
> ---------------------------------------------
> the FS is download from git and wanpipe is latest version.  please give a
> help for that issue.
>
> Best regards,
> James.zhu
> website: www.hiastar.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel
> Communication Server  Official FreeSWITCH Sites
> http://www.freeswitch.org http://wiki.freeswitch.org
> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jmesquita at freeswitch.org  Sat Aug  3 07:31:25 2013
From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Fri, 2 Aug 2013 22:31:25 -0500
Subject: [Freeswitch-users] att_xfer origination_cancel_key
In-Reply-To: <7EA4E9B1D66C4202A60862B65D251EAB@PBPC>
References: <7EA4E9B1D66C4202A60862B65D251EAB@PBPC>
Message-ID: <8B109374-6241-4C55-800A-6BCE4E71E618@freeswitch.org>

I am not looking at the code right now but if I recall correctly no you cannot. It is trivial to make single key configurable. Open a Jira and I can make a patch for it to configure using channel variables. If you want multiple keys like bda, forget it. A lot more complicated...

Sent from my iPhone

On Aug 2, 2013, at 6:17 PM, "Stuart Mills" <stuart.mills3 at btopenworld.com> wrote:

> Hi,
>  
> I have noticed that att_xfer has a configurable cancel key to stop the transfer mid dial, is there an option to change keys for the other transfer features?
>  
> 0 is conference all 3 parties at the moment, but I'd like to designate a 4 or some other key, is this possible?
>  
> Many Thanks,
>  
> Stuart Mills
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bhegades at gmail.com  Sat Aug  3 07:53:49 2013
From: bhegades at gmail.com (Mahendra Bhegade)
Date: Fri, 2 Aug 2013 20:53:49 -0700
Subject: [Freeswitch-users] Resources used on server for conferencing
Message-ID: <CAC7mRFnh6g=GFwUe4vKCgcYzk527X6XL0o3Nk2EJ67sQZ1b=nQ@mail.gmail.com>

Hi,

I am using flowroute to access incoming calls and then conference
additional phones using flowroute SIP gateway.

I would like to understand what resources are being used on the server that
is hosting the conference calls.

Now if flowroute is handling all the call what resources are being consumed
on the freeswitch server.

CPU, bandwidth, what else ?

If these are cellphone versus landlines does it make any difference ?

Please share some insight.

Mahendra
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From steveu at coppice.org  Sat Aug  3 08:42:52 2013
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 03 Aug 2013 12:42:52 +0800
Subject: [Freeswitch-users] [OL]Re:  best way to receive FAXs
In-Reply-To: <51FC1D8C.4020204@xtronics.com>
References: <CE215868.78885%krice@freeswitch.org>
	<51FBFD98.9030909@xtronics.com>
	<CAHgk4BVRfHcSsy5LV5Xx8Auu+-EBLW+7xHUG4x=fgsgJBpws5A@mail.gmail.com>
	<51FC1D8C.4020204@xtronics.com>
Message-ID: <51FC8A4C.9000407@coppice.org>

You seem to be making a mountain out of a mole hill. Unless you have 
some very specific need for HylaFAX, just install FreeSwitch and let it 
FAX for you. If you do need HylaFAX, then install HylaFAX and FreeSwitch 
and let the T.31 FAX modem facility in FreeSwitch be your modem. If you 
need to send more than 100k FAXes a day you might need to consider 
performance issues.

Regards,
Steve

On 08/03/2013 04:58 AM, Karl Schmidt wrote:
> On 08/02/2013 03:02 PM, Brian Foster wrote:
>> I've personally had issues with hylafax. I dont know what it is or how to fix it, but what I ended
>> up doing is using mod_spandsp and dumping the received faxes into a couchdb database. I have an
>> example here (http://github.com/bdfoster/fax-to-couchdb). It's rough, but it might give you some ideas.
> Thanks - I had to go through 6 modems before I found one that did fax reliably - it is the firmware
> code in the modem that matters - and all of the big names failed. Worked with an external Supra Fax
> modem 144LC
>
> If you ever need to get it to work, I can send my Debian configs.
>
> --------------------------------------------------------------------------------
> Karl Schmidt                                  EMail Karl at xtronics.com
> Transtronics, Inc.                              WEB http://secure.transtronics.com
> 3209 West 9th Street                             Ph (785) 841-3089
> Lawrence, KS 66049                              FAX (785) 841-0434
>
> Most economic fallacies derive from the tendency to assume that there is a fixed pie, that one party
> can gain only at the expense of another.
> Milton Friedman
> --------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 richard.mace at gmail.com  Sat Aug  3 10:08:52 2013
From: richard.mace at gmail.com (Richard Mace)
Date: Sat, 3 Aug 2013 07:08:52 +0100
Subject: [Freeswitch-users] Naming xml files
Message-ID: <CAK+KT3-Yum6wPXnYQTgMjiFubkU2-PMzh+j4=4r3FxoqwjkOyQ@mail.gmail.com>

Hi,
In the freeSWITCH 1.2 book, it talks about xml files, such as an incoming
route being called 01_ and then a name such as iptel.xml
Is the "01_" that important/a good idea/needed or is it just so that it is
displayed as the first entry in the directory when browsing the files?

Thanks in advance

Richard
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From vbvbrj at gmail.com  Sat Aug  3 10:18:35 2013
From: vbvbrj at gmail.com (Mimiko)
Date: Sat, 03 Aug 2013 09:18:35 +0300
Subject: [Freeswitch-users] Naming xml files
In-Reply-To: <CAK+KT3-Yum6wPXnYQTgMjiFubkU2-PMzh+j4=4r3FxoqwjkOyQ@mail.gmail.com>
References: <CAK+KT3-Yum6wPXnYQTgMjiFubkU2-PMzh+j4=4r3FxoqwjkOyQ@mail.gmail.com>
Message-ID: <51FCA0BB.3080407@gmail.com>

On 03.08.2013 09:08, Richard Mace wrote:
> Is the "01_" that important/a good idea/needed or is it just so that it
> is displayed as the first entry in the directory when browsing the files?
>

Naming of your files isn't important, but keep in mind that when 
including a list of *.xml files, Fs loads them in sorted ascending 
order. This is handy for example when loading dialplan rules from files, 
which must be in the correct order in resulting xml file, but keep each 
rule for example in different file for fast search and edit.

-- 
Mimiko desu.


From richard.mace at gmail.com  Sat Aug  3 10:29:40 2013
From: richard.mace at gmail.com (Richard Mace)
Date: Sat, 3 Aug 2013 07:29:40 +0100
Subject: [Freeswitch-users] Naming xml files
In-Reply-To: <51FCA0BB.3080407@gmail.com>
References: <CAK+KT3-Yum6wPXnYQTgMjiFubkU2-PMzh+j4=4r3FxoqwjkOyQ@mail.gmail.com>
	<51FCA0BB.3080407@gmail.com>
Message-ID: <CAK+KT38efEDLVsD8TYd2oMn9gOYK6sLa7opZx3r3QZu81ErTKw@mail.gmail.com>

On 3 August 2013 07:18, Mimiko <vbvbrj at gmail.com> wrote:

> On 03.08.2013 09:08, Richard Mace wrote:
> > Is the "01_" that important/a good idea/needed or is it just so that it
> > is displayed as the first entry in the directory when browsing the files?
> >
>
> Naming of your files isn't important, but keep in mind that when
> including a list of *.xml files, Fs loads them in sorted ascending
> order. This is handy for example when loading dialplan rules from files,
> which must be in the correct order in resulting xml file, but keep each
> rule for example in different file for fast search and edit.
>
> --
> Mimiko desu.
>

Thanks Mimiko

Richard
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From stuart.mills3 at btopenworld.com  Sat Aug  3 11:57:33 2013
From: stuart.mills3 at btopenworld.com (Stuart Mills)
Date: Sat, 3 Aug 2013 08:57:33 +0100
Subject: [Freeswitch-users] att_xfer origination_cancel_key
In-Reply-To: <8B109374-6241-4C55-800A-6BCE4E71E618@freeswitch.org>
References: <7EA4E9B1D66C4202A60862B65D251EAB@PBPC>
	<8B109374-6241-4C55-800A-6BCE4E71E618@freeswitch.org>
Message-ID: <B82AE16B-71B3-4860-8DC3-72D7EB1C2A8A@btopenworld.com>

That is great, thanks for your reply.

I haven't opened a Jira before, so will read up on the wiki and put a request in.

Regards

Stuart

Sent from my iPhone

On 3 Aug 2013, at 04:31, Jo?o Mesquita <jmesquita at freeswitch.org> wrote:

> I am not looking at the code right now but if I recall correctly no you cannot. It is trivial to make single key configurable. Open a Jira and I can make a patch for it to configure using channel variables. If you want multiple keys like bda, forget it. A lot more complicated...
> 
> Sent from my iPhone
> 
> On Aug 2, 2013, at 6:17 PM, "Stuart Mills" <stuart.mills3 at btopenworld.com> wrote:
> 
>> Hi,
>>  
>> I have noticed that att_xfer has a configurable cancel key to stop the transfer mid dial, is there an option to change keys for the other transfer features?
>>  
>> 0 is conference all 3 parties at the moment, but I'd like to designate a 4 or some other key, is this possible?
>>  
>> Many Thanks,
>>  
>> Stuart Mills
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 thomas.lee at octon.net  Sat Aug  3 14:41:07 2013
From: thomas.lee at octon.net (Thomas Lee)
Date: Sat, 3 Aug 2013 03:41:07 -0700 (PDT)
Subject: [Freeswitch-users] How to configure DTMF detection using 'info' and
	'rfc2833' both?
Message-ID: <1375526467565-7593558.post@n2.nabble.com>

How to configure DTMF detection using 'info' and 'rfc2833' both?Thank you.



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-configure-DTMF-detection-using-info-and-rfc2833-both-tp7593558.html
Sent from the freeswitch-users mailing list archive at Nabble.com.
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From steveayre at gmail.com  Sat Aug  3 14:42:00 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 3 Aug 2013 11:42:00 +0100
Subject: [Freeswitch-users] Restrict inbound codecs for specific user
Message-ID: <CAFiqYu=oCZRWze2Eqx8kUtQNQSNee0_nUL5a_WZzX1RjQZmb7g@mail.gmail.com>

Is there any variable that can be set in the user directory to
restrict/override the profile's inbound codec preferences for a specific
user account?

I'm aware I can do this with late negotiation and setting
absolute_codec_string, but is prefer to reject calls without a supported
dialplan before they hit the dialplan and go to a bridge.

-Steve
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From steveayre at gmail.com  Sat Aug  3 15:09:52 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 3 Aug 2013 12:09:52 +0100
Subject: [Freeswitch-users] Restrict inbound codecs for specific user
Message-ID: <CAFiqYunxhXqPJ7TSTrzwJcNPVU=_D4u9CYj2QdNtYeDypgPTOQ@mail.gmail.com>

Is there any variable that can be set in the user directory to
restrict/override the profile's inbound codec preferences for a specific
user account?

I'm aware I can do this with late negotiation and setting
absolute_codec_string, but is prefer to reject calls without a supported
dialplan before they hit the dialplan and go to a bridge.

-Steve
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From anthony.minessale at gmail.com  Sat Aug  3 17:30:12 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 3 Aug 2013 08:30:12 -0500
Subject: [Freeswitch-users] Restrict inbound codecs for specific user
In-Reply-To: <CAFiqYunxhXqPJ7TSTrzwJcNPVU=_D4u9CYj2QdNtYeDypgPTOQ@mail.gmail.com>
References: <CAFiqYunxhXqPJ7TSTrzwJcNPVU=_D4u9CYj2QdNtYeDypgPTOQ@mail.gmail.com>
Message-ID: <CAKbxfG-5NHPRsM_S6Aw47NDaQ5cajH3=d+uVX-dZLwix_01u6g@mail.gmail.com>

You might be able to set absolute_codec_string from the <variables> in time
for negotiation to take place but I can't recall.  If that doesn't work the
the answer would be no.



On Sat, Aug 3, 2013 at 6:09 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Is there any variable that can be set in the user directory to
> restrict/override the profile's inbound codec preferences for a specific
> user account?
>
> I'm aware I can do this with late negotiation and setting
> absolute_codec_string, but is prefer to reject calls without a supported
> dialplan before they hit the dialplan and go to a bridge.
>
> -Steve
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From bdfoster at davri.com  Sat Aug  3 19:42:58 2013
From: bdfoster at davri.com (Brian Foster)
Date: Sat, 3 Aug 2013 11:42:58 -0400
Subject: [Freeswitch-users] Mod_directory: "connecting call" sound?
In-Reply-To: <CAGvuWYbb1kQKd+4wjtdk1dddJVYDoMwyVV1FHo7AMvJ3CE2bzQ@mail.gmail.com>
References: <CAHgk4BXBAg8EHqizJz8QoCvsxjS_WGKY=tydoMfjpmB3RKfeLg@mail.gmail.com>
	<CAGvuWYbb1kQKd+4wjtdk1dddJVYDoMwyVV1FHo7AMvJ3CE2bzQ@mail.gmail.com>
Message-ID: <CAHgk4BWYwCxqD5Nv8CE8H4OR=yYHjDur-VWCVgEb-O9JTG=2QQ@mail.gmail.com>

Thanks Lloyd. I'll probably use that, maybe I can submit a patch so that
you can set a sound file to play after entry is confirmed.

Thank you,

Brian Foster
Project Manager/Owner's Representative
Davri Investments, Incorporated
P: +1-317-787-2686
M: +1-317-600-9753
Indianapolis, Indiana
before call directory application set up a
channel variable call_from="directory" , set/export

in user dial plan before bridge check the channel variable call_from and if
match "directory", play the  "One moment while I connect your call"

Lloyd

*
*
*  <http://www.sunteltech.ca/blog/>*


On Fri, Aug 2, 2013 at 1:10 AM, Brian Foster <bdfoster at davri.com> wrote:

> I can't find it anywhere in the wiki or in the config files, but is there
> a way to specify a "connecting call" sound?
>
> Example: A caller enters the directory application, he/she selects an
> entry. When they select the entry, it goes straight to ringing. Before that
> call is connected, can we specify that it plays a sound such as "One moment
> while I connect your call"?
>
> I realize this might be a feature request, in which case I'll probably
> take it upon myself to rummage through the source code and come up with a
> patch to go with a JIRA ticket.
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Representative
> Davri Investments, Incorporated
> P: +1-317-787-2686
> M: +1-317-600-9753
> Indianapolis, Indiana
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 imitev at c3i.bg  Sat Aug  3 18:07:07 2013
From: imitev at c3i.bg (Ivan Mitev)
Date: Sat, 03 Aug 2013 17:07:07 +0300
Subject: [Freeswitch-users] garbled audio with G726-32, other codecs are fine
Message-ID: <51FD0E8B.3060702@c3i.bg>

Hello

I'm migrating an office setup from asterisk to FS and in the process I 
was considering using G726-32 for some bandwidth starved remote 
endpoints. However I only get metallic/garbled audio with that codec 
even when simply playing moh to the endpoint, while other codecs work 
fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds 
marginally better but still garbled and really worse than G711.

The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest 
(centos6 64bit host). But please don't shoot ! :) - I know about virtual 
environment limitations but for these tests the host is only lightly 
loaded, there aren't any calls to the FS instance except my tests, and 
the fact that it works with other codecs makes me think that 
virtualization is not the issue here. I may be wrong though.

Is there any guide for debugging that kind of problem before reverting 
to a fresh install on bare-metal with the latest HEAD ? Until now I've 
tried:

- improving timers ; but the default soft timer (which I guess uses 
timerd) works best. The time interval between sent packets on a tcpdump 
trace looks identical to the output of "timer_test", so that doesn't 
seem to be a network/jitter problem. And there's no problem with other 
codecs, but maybe G726-XX is specific. For info the guest's clocksource 
is kvm_clock, while the host uses tsc.

- using different endpoints: the production ones are Linksys PAP2 
("fixed" for 20ms psize, and G726-32 SDP type indentification), but the 
same thing happens with linphone on a fedora 19 laptop.

A call with rtp media going through FS without transcoding - G726-32 to 
G726-32 - works perfectly (I can't hear the difference with G711). The 
problem is only when there's transcoding to G726 (from wav for moh, or 
from any other codec when bridging). I've looked at the wiki, posts, 
changelogs, jira, ..., but am a bit at a loss now.

Any pointers ?

Except that little problem, FS rocks, and I'm happy I can finally ditch 
asterisk. Kudos to the core devs and contributors.

Ivan



From khorsmann at gmail.com  Sun Aug  4 01:13:48 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Sat, 3 Aug 2013 23:13:48 +0200
Subject: [Freeswitch-users] How to configure DTMF detection using 'info'
 and 'rfc2833' both?
In-Reply-To: <1375526467565-7593558.post@n2.nabble.com>
References: <1375526467565-7593558.post@n2.nabble.com>
Message-ID: <CAFArqsbP6tRhJsEv-WT2cv_Rq7sWr_eJQQPaYW0OkF4eCB0fzg@mail.gmail.com>

Hi,

as the wiki described it should be this in your sip-profile

<param name="liberal-dtmf" value="true"/>



2013/8/3 Thomas Lee <thomas.lee at octon.net>

> How to configure DTMF detection using 'info' and 'rfc2833' both? Thank
> you.
> ------------------------------
> View this message in context: How to configure DTMF detection using
> 'info' and 'rfc2833' both?<http://freeswitch-users.2379917.n2.nabble.com/How-to-configure-DTMF-detection-using-info-and-rfc2833-both-tp7593558.html>
> Sent from the freeswitch-users mailing list archive<http://freeswitch-users.2379917.n2.nabble.com/>at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From lists at kavun.ch  Sun Aug  4 01:31:26 2013
From: lists at kavun.ch (Emrah)
Date: Sat, 3 Aug 2013 17:31:26 -0400
Subject: [Freeswitch-users] Possible Polycom bridging issues in all
	recent FreeSWITCH versions
In-Reply-To: <BAD9DC74-5691-4688-91A1-C2C9CD17D34C@kavun.ch>
References: <17D13B62-FDCC-462F-A376-65BC8DE474C5@kavun.ch>
	<CAAtWKeEvqNC_9D4L7PjKSkmgzrOryqQ3ER6Z7dqM4Fqi52JAbA@mail.gmail.com>
	<9CD85028-3064-4439-A846-705FED3F16D2@kavun.ch>
	<BAD9DC74-5691-4688-91A1-C2C9CD17D34C@kavun.ch>
Message-ID: <AAF4E78A-94C6-4CA2-A93F-7D03E5DE2C16@kavun.ch>

Hi guys,

I've been going in circles for quite some time now as you can see in the history of this message. And this is starting to weigh heavy.

The bug has made it into stable versions and I would love to see it fixed soon? What is needed to troubleshoot on my end?

The data is:
1. Caller cannot hear callee  in a bridge situation but video works fine. 
2. The issue happens when G721 at 32000h and H264 is used. I believe it also happens without H264 involved.
3. If the called party holds/unholds the line and a renegotiation of codecs happens the audio works 2-way to the detriment of 1-way video.
4. The issue does not occur when using apps instead of bridging calls. E.g.: conference.
5. Using an old mod_siren does not solve the problem.

Some attention to this would be much appreciated.

Thanks
On Jul 14, 2013, at 6:35 PM, Emrah <lists at kavun.ch> wrote:

> Anybody up to help on this?
> The problem only occurs if mod_siren is involved it seems like.
> A call in G722 in between 2 polycoms will be set up just fine.
> The same in G7221 at 32000h will cause the audio to be one-way only.
> This is with Polycom VVX1500 Media phones with video.
> I would be happy to provide logs if there is a way to filter them to specific users / extensions. My console is just too crowded otherwise.
> On Jul 1, 2013, at 1:18 PM, Emrah <lists at kavun.ch> wrote:
> 
>> Hi guys, 
>> 
>> Thanks for all the input.
>> 
>> Here are my clarifications along with more questions.
>> 
>> I am not using encryption for these extensions.
>> Is there a way to limit logs to specific users / extensions?
>> How can I go back in GIT releases to do some regression testing and how many releases are they per month / week / day? 
>> 
>> This is very recent I'm sure. The fascinating part is that it made it all the way to the stable branch.
>> 
>> Thanks
>> 
>> 
>> 
>> On Jun 30, 2013, at 12:35 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:
>> 
>>> Do you use audio encryption? If so, try disabling it and see whether it changes something in the bahaviour.
>>> 
>>>            __Yehavi:
>>> 
>>> 
>>> 2013/6/30 Emrah <lists at kavun.ch>
>>> Hi,
>>> 
>>> I have been experiencing the following issue since I upgraded my version of FS.
>>> In a scenario where a called is made from one Polycom VVX1500 to another, the video works both ways with only one way audio. A hold / unhold results in 2 way audio with 1 way video.
>>> I obviously suspected my configs, especially after  subsequently trying the latest git, the 1.4 beta and the stable 1.2.10.
>>> So far, the only way I found to solve this issue is by using an old copy of FreeSWITCH with my current configs.
>>> FreeSWITCH Version 1.3.13b+git~20130119T155523Z~6cb3be7d8b (git 6cb3be7 2013-01-19 15:55:23Z)
>>> 
>>> To clarify, the same config with any of the latest versions of FS does not work.
>>> 
>>> Questions:
>>> How can I troubleshoot this further and help resolve this issue?
>>> Any one has an idea of what has happened?
>>> 
>>> Thanks!
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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 bdfoster at davri.com  Sun Aug  4 01:37:14 2013
From: bdfoster at davri.com (Brian Foster)
Date: Sat, 3 Aug 2013 17:37:14 -0400
Subject: [Freeswitch-users] garbled audio with G726-32,
	other codecs are fine
In-Reply-To: <51FD0E8B.3060702@c3i.bg>
References: <51FD0E8B.3060702@c3i.bg>
Message-ID: <CAHgk4BVNA+NOg5ChyYxSddDxhMxgSmWVe1tQ7NiCPRKo6M5BCw@mail.gmail.com>

AAC bitpacking by any chance? I thought I had a similar issue, happened so
long ago I cant remember what I did.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev at c3i.bg> wrote:

> Hello
>
> I'm migrating an office setup from asterisk to FS and in the process I
> was considering using G726-32 for some bandwidth starved remote
> endpoints. However I only get metallic/garbled audio with that codec
> even when simply playing moh to the endpoint, while other codecs work
> fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds
> marginally better but still garbled and really worse than G711.
>
> The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest
> (centos6 64bit host). But please don't shoot ! :) - I know about virtual
> environment limitations but for these tests the host is only lightly
> loaded, there aren't any calls to the FS instance except my tests, and
> the fact that it works with other codecs makes me think that
> virtualization is not the issue here. I may be wrong though.
>
> Is there any guide for debugging that kind of problem before reverting
> to a fresh install on bare-metal with the latest HEAD ? Until now I've
> tried:
>
> - improving timers ; but the default soft timer (which I guess uses
> timerd) works best. The time interval between sent packets on a tcpdump
> trace looks identical to the output of "timer_test", so that doesn't
> seem to be a network/jitter problem. And there's no problem with other
> codecs, but maybe G726-XX is specific. For info the guest's clocksource
> is kvm_clock, while the host uses tsc.
>
> - using different endpoints: the production ones are Linksys PAP2
> ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the
> same thing happens with linphone on a fedora 19 laptop.
>
> A call with rtp media going through FS without transcoding - G726-32 to
> G726-32 - works perfectly (I can't hear the difference with G711). The
> problem is only when there's transcoding to G726 (from wav for moh, or
> from any other codec when bridging). I've looked at the wiki, posts,
> changelogs, jira, ..., but am a bit at a loss now.
>
> Any pointers ?
>
> Except that little problem, FS rocks, and I'm happy I can finally ditch
> asterisk. Kudos to the core devs and contributors.
>
> Ivan
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bhegades at gmail.com  Sun Aug  4 02:48:54 2013
From: bhegades at gmail.com (Mahendra Bhegade)
Date: Sat, 3 Aug 2013 15:48:54 -0700
Subject: [Freeswitch-users] Resource used on server for conferencing
Message-ID: <CAC7mRFmmX9zkQ6Aat8zooaGdofhg4B3gZxMzdfxcCHw=UWmEHA@mail.gmail.com>

Hi,

I am using flowroute to access incoming calls and then conference
additional phones using flowroute SIP gateway.

I would like to understand what resources are being used on the server that
is hosting the conference calls.

Now if flowroute is handling all the call what resources are being consumed
on the freeswitch server.

CPU, bandwidth, what else ?

If these are cellphone versus landlines does it make any difference ?

Please share some insight.
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From gabe at gundy.org  Sun Aug  4 03:12:57 2013
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 3 Aug 2013 17:12:57 -0600
Subject: [Freeswitch-users] Possible Polycom bridging issues in all
 recent FreeSWITCH versions
In-Reply-To: <AAF4E78A-94C6-4CA2-A93F-7D03E5DE2C16@kavun.ch>
References: <17D13B62-FDCC-462F-A376-65BC8DE474C5@kavun.ch>
	<CAAtWKeEvqNC_9D4L7PjKSkmgzrOryqQ3ER6Z7dqM4Fqi52JAbA@mail.gmail.com>
	<9CD85028-3064-4439-A846-705FED3F16D2@kavun.ch>
	<BAD9DC74-5691-4688-91A1-C2C9CD17D34C@kavun.ch>
	<AAF4E78A-94C6-4CA2-A93F-7D03E5DE2C16@kavun.ch>
Message-ID: <CAMwZ-tJ4sEfyeT=VGNMfcWm=sZK1jeB2mBT8JA8RAkvtVCqVpQ@mail.gmail.com>

On Sat, Aug 3, 2013 at 3:31 PM, Emrah <lists at kavun.ch> wrote:
> I've been going in circles for quite some time now as you can see in the history of this message. And this is starting to weigh heavy.
>
> The bug has made it into stable versions and I would love to see it fixed soon? What is needed to troubleshoot on my end?

This is what you need to know:
https://wiki.freeswitch.org/wiki/Reporting_Bugs


Best,
Gabe


From krice at freeswitch.org  Sun Aug  4 03:35:24 2013
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 03 Aug 2013 18:35:24 -0500
Subject: [Freeswitch-users] Resource used on server for conferencing
In-Reply-To: <CAC7mRFmmX9zkQ6Aat8zooaGdofhg4B3gZxMzdfxcCHw=UWmEHA@mail.gmail.com>
Message-ID: <CE22FDEC.788EE%krice@freeswitch.org>

you only have to send a query to the list once... Someone will eventually
respond if they know the answer

So now...

If you have calls coming into FS via FlowRoute, there is no difference
related to resources based on if its a landline or a cellphone, there could
be differences based on what codec is actually chosen (ie: G711 is less
costly in CPU performance then G729 or other compressed codecs)

Flowroute is not doing the conferencing, either your freeswitch box is, or a
local phone is doing the conference... If you are using mod_conference there
is cpu and ram usage for each call.. How much depends on various settings
etc...

Of course there is a bandwidth component for each call...

I would suggest getting the FreeSWITCH book, and start there, and probably
vising http://www.siptutorial.net/ and get a basic understanding of how VoIP
works in general


On 8/3/13 5:48 PM, "Mahendra Bhegade" <bhegades at gmail.com> wrote:

> Hi,
> 
> I am using flowroute to access incoming calls and then conference additional
> phones using flowroute SIP gateway.
> 
> I would like to understand what resources are being used on the server that is
> hosting the conference calls.
> 
> Now if flowroute is handling all the call what resources are being consumed on
> the freeswitch server.
> 
> CPU, bandwidth, what else ?
> 
> If these are cellphone versus landlines does it make any difference ?
> 
> Please share some insight.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From msc at freeswitch.org  Sun Aug  4 03:50:10 2013
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 3 Aug 2013 16:50:10 -0700
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>
	<1138329210A14BE4AEE53D7419427A15@PBPC>
	<CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
	<CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
	<3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
Message-ID: <CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>

It might be best of you posted those on pastebin with explanations and
console logs of the calls.
-MC


On Fri, Aug 2, 2013 at 3:24 PM, Stuart Mills
<stuart.mills3 at btopenworld.com>wrote:

>   Hi,
>
> Thanks for your reply.
>
> I have already changed the parameter log-b-leg to true in xml_cdr.conf and
> I am getting 2 records for the outbound calls and 1 for inbound. The issue
> is that the app_log section in the a_leg cdr seems to be split across the
> a_leg cdr and the b_leg cdr that ?won? the race to be answered first. The
> second b_leg has no duration as expected.
>
> Am I making sense?
>
> Regards,
>
> Stuart
>
>  *From:* Lloyd Aloysius <lloyd.aloysius at sunteltech.ca>
> *Sent:* Friday, August 02, 2013 5:45 PM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Call at once and XML-CDR
>
> Another note  look into the caller_profile and index. I think you have two
> indexes for this situation.In your logic insert the records for the
> multiple indexes.
>
> Lloyd
>  *
> *
> **
>
>
> On Fri, Aug 2, 2013 at 12:44 PM, Lloyd Aloysius <
> lloyd.aloysius at sunteltech.ca> wrote:
>
>> With the default configuration switch only log a_leg.
>>
>> You need to enable b_leg in xml_cdr.conf.xml . Then you will see the two
>> records in b_leg, with one get billsec > 0 for the the answered call.
>>
>> Lloyd
>>
>>
>>
>> On Fri, Aug 2, 2013 at 10:57 AM, Stuart Mills <
>> stuart.mills3 at btopenworld.com> wrote:
>>
>>>   Hi Avi,
>>>
>>> Thanks for your reply.
>>>
>>> Yeah I'm sure, I'm using ESL and if I dial a single number it performs
>>> as expected, it?s literally only when I'm making 2 or more calls at once it
>>> happens.
>>>
>>> Cheers,
>>>
>>> Stuart
>>>
>>>  *From:* Avi Marcus <avi at avimarcus.net>
>>> *Sent:* Friday, August 02, 2013 3:24 PM
>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> *Subject:* Re: [Freeswitch-users] Call at once and XML-CDR
>>>
>>>
>>> Are you sure these aren't b leg things?
>>> Do you have an example of your dial plan and the logs?
>>>
>>> -Avi
>>> On Aug 2, 2013 5:09 PM, "Stuart Mills" <stuart.mills3 at btopenworld.com>
>>> wrote:
>>>
>>>>   Hi,
>>>>
>>>> Got a question regarding call at once combined with xml-cdr logging.
>>>>
>>>> The problem I seem to get is when I try sending calls to 2 destinations
>>>> and 1 call gets answered, so that wins the race, the xml-cdr logging seems
>>>> to be a little confused. In the app_log I?m getting logging up to the point
>>>> of the dial string, then nothing after that for the a-leg. What seems to
>>>> happen is the rest of the app_log gets written to the cdr file of the bleg
>>>> that answered the call first.
>>>>
>>>> Is this by design or a bug in the xml-cdr logging?
>>>>
>>>> Thanks,
>>>>
>>>> Stuart
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>> ------------------------------
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
-------------- next part --------------
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From msc at freeswitch.org  Sun Aug  4 03:57:55 2013
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 3 Aug 2013 16:57:55 -0700
Subject: [Freeswitch-users] att_xfer origination_cancel_key
In-Reply-To: <B82AE16B-71B3-4860-8DC3-72D7EB1C2A8A@btopenworld.com>
References: <7EA4E9B1D66C4202A60862B65D251EAB@PBPC>
	<8B109374-6241-4C55-800A-6BCE4E71E618@freeswitch.org>
	<B82AE16B-71B3-4860-8DC3-72D7EB1C2A8A@btopenworld.com>
Message-ID: <CAKzWOxXk1Vrc21CNA7SU1HzdJaAVMhG8hzb2dw0JgdHPZeY-Ow@mail.gmail.com>

You may also research and see how you might add this yourself. The place to
look is in mod_dialplan.c at approx line 2230 inside static switch_status_t
xfer_on_dtmf(...) :

            if (dtmf->digit == '0') {
                switch_caller_extension_t *extension = NULL;
                const char *app = "three_way";
<snip>

In any case, JM is right that it is relatively easy to change that to check
for a channel variable. If you peruse the code you'll find lots of examples
of that sort of thing. If it's not your cup of tea then by all means open a
jira as a feature request.

-MC


On Sat, Aug 3, 2013 at 12:57 AM, Stuart Mills <stuart.mills3 at btopenworld.com
> wrote:

> That is great, thanks for your reply.
>
> I haven't opened a Jira before, so will read up on the wiki and put a
> request in.
>
> Regards
>
> Stuart
>
> Sent from my iPhone
>
> On 3 Aug 2013, at 04:31, Jo?o Mesquita <jmesquita at freeswitch.org> wrote:
>
> I am not looking at the code right now but if I recall correctly no you
> cannot. It is trivial to make single key configurable. Open a Jira and I
> can make a patch for it to configure using channel variables. If you want
> multiple keys like bda, forget it. A lot more complicated...
>
> Sent from my iPhone
>
> On Aug 2, 2013, at 6:17 PM, "Stuart Mills" <stuart.mills3 at btopenworld.com>
> wrote:
>
>  Hi,
>
> I have noticed that att_xfer has a configurable cancel key to stop the
> transfer mid dial, is there an option to change keys for the other transfer
> features?
>
> 0 is conference all 3 parties at the moment, but I'd like to designate a 4
> or some other key, is this possible?
>
> Many Thanks,
>
> Stuart Mills
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
-------------- next part --------------
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From stuart.mills3 at btopenworld.com  Sun Aug  4 04:16:41 2013
From: stuart.mills3 at btopenworld.com (Stuart Mills)
Date: Sun, 4 Aug 2013 01:16:41 +0100
Subject: [Freeswitch-users] att_xfer origination_cancel_key
In-Reply-To: <CAKzWOxXk1Vrc21CNA7SU1HzdJaAVMhG8hzb2dw0JgdHPZeY-Ow@mail.gmail.com>
References: <7EA4E9B1D66C4202A60862B65D251EAB@PBPC><8B109374-6241-4C55-800A-6BCE4E71E618@freeswitch.org><B82AE16B-71B3-4860-8DC3-72D7EB1C2A8A@btopenworld.com>
	<CAKzWOxXk1Vrc21CNA7SU1HzdJaAVMhG8hzb2dw0JgdHPZeY-Ow@mail.gmail.com>
Message-ID: <A09F84DC4E8547CE8C3F3707B63FCFD0@PBPC>

Thanks for the tip Michael, I might just take a look at that code when I get a spare minute - I?ll probably still open a jira though.

Stuart

From: Michael Collins 
Sent: Sunday, August 04, 2013 12:57 AM
To: FreeSWITCH Users Help 
Subject: Re: [Freeswitch-users] att_xfer origination_cancel_key

You may also research and see how you might add this yourself. The place to look is in mod_dialplan.c at approx line 2230 inside static switch_status_t xfer_on_dtmf(...) :

            if (dtmf->digit == '0') {
                switch_caller_extension_t *extension = NULL;
                const char *app = "three_way";

<snip>


In any case, JM is right that it is relatively easy to change that to check for a channel variable. If you peruse the code you'll find lots of examples of that sort of thing. If it's not your cup of tea then by all means open a jira as a feature request.


-MC




On Sat, Aug 3, 2013 at 12:57 AM, Stuart Mills <stuart.mills3 at btopenworld.com> wrote:

  That is great, thanks for your reply.

  I haven't opened a Jira before, so will read up on the wiki and put a request in.

  Regards

  Stuart

  Sent from my iPhone

  On 3 Aug 2013, at 04:31, Jo?o Mesquita <jmesquita at freeswitch.org> wrote:


    I am not looking at the code right now but if I recall correctly no you cannot. It is trivial to make single key configurable. Open a Jira and I can make a patch for it to configure using channel variables. If you want multiple keys like bda, forget it. A lot more complicated...

    Sent from my iPhone

    On Aug 2, 2013, at 6:17 PM, "Stuart Mills" <stuart.mills3 at btopenworld.com> wrote:


      Hi,

      I have noticed that att_xfer has a configurable cancel key to stop the transfer mid dial, is there an option to change keys for the other transfer features? 

      0 is conference all 3 parties at the moment, but I'd like to designate a 4 or some other key, is this possible?

      Many Thanks,

      Stuart Mills
      _________________________________________________________________________
      Professional FreeSWITCH Consulting Services:
      consulting at freeswitch.org
      http://www.freeswitchsolutions.com

      
      

      Official FreeSWITCH Sites
      http://www.freeswitch.org
      http://wiki.freeswitch.org
      http://www.cluecon.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

    _________________________________________________________________________
    Professional FreeSWITCH Consulting Services:
    consulting at freeswitch.org
    http://www.freeswitchsolutions.com

    
    

    Official FreeSWITCH Sites
    http://www.freeswitch.org
    http://wiki.freeswitch.org
    http://www.cluecon.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


  _________________________________________________________________________
  Professional FreeSWITCH Consulting Services:
  consulting at freeswitch.org
  http://www.freeswitchsolutions.com

  
  

  Official FreeSWITCH Sites
  http://www.freeswitch.org
  http://wiki.freeswitch.org
  http://www.cluecon.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





-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org




--------------------------------------------------------------------------------
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 stuart.mills3 at btopenworld.com  Sun Aug  4 04:17:40 2013
From: stuart.mills3 at btopenworld.com (Stuart Mills)
Date: Sun, 4 Aug 2013 01:17:40 +0100
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC><000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com><1138329210A14BE4AEE53D7419427A15@PBPC><CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com><CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com><3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
	<CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>
Message-ID: <B782F8192B3E46608CE6666897F86BEB@PBPC>

yeah I can do that, although I've never used pastebin, do I need to register to use that?

From: Michael Collins 
Sent: Sunday, August 04, 2013 12:50 AM
To: FreeSWITCH Users Help 
Subject: Re: [Freeswitch-users] Call at once and XML-CDR

It might be best of you posted those on pastebin with explanations and console logs of the calls.

-MC




On Fri, Aug 2, 2013 at 3:24 PM, Stuart Mills <stuart.mills3 at btopenworld.com> wrote:

  Hi,

  Thanks for your reply.

  I have already changed the parameter log-b-leg to true in xml_cdr.conf and I am getting 2 records for the outbound calls and 1 for inbound. The issue is that the app_log section in the a_leg cdr seems to be split across the a_leg cdr and the b_leg cdr that ?won? the race to be answered first. The second b_leg has no duration as expected.

  Am I making sense?

  Regards,

  Stuart 

  From: Lloyd Aloysius 
  Sent: Friday, August 02, 2013 5:45 PM
  To: FreeSWITCH Users Help 
  Subject: Re: [Freeswitch-users] Call at once and XML-CDR

  Another note  look into the caller_profile and index. I think you have two indexes for this situation.In your logic insert the records for the multiple indexes. 


  Lloyd







  On Fri, Aug 2, 2013 at 12:44 PM, Lloyd Aloysius <lloyd.aloysius at sunteltech.ca> wrote:

    With the default configuration switch only log a_leg. 


    You need to enable b_leg in xml_cdr.conf.xml . Then you will see the two records in b_leg, with one get billsec > 0 for the the answered call. 


    Lloyd






    On Fri, Aug 2, 2013 at 10:57 AM, Stuart Mills <stuart.mills3 at btopenworld.com> wrote:

      Hi Avi,

      Thanks for your reply.

      Yeah I'm sure, I'm using ESL and if I dial a single number it performs as expected, it?s literally only when I'm making 2 or more calls at once it happens.

      Cheers,

      Stuart

      From: Avi Marcus 
      Sent: Friday, August 02, 2013 3:24 PM
      To: FreeSWITCH Users Help 
      Subject: Re: [Freeswitch-users] Call at once and XML-CDR

      Are you sure these aren't b leg things?
      Do you have an example of your dial plan and the logs?

      -Avi

      On Aug 2, 2013 5:09 PM, "Stuart Mills" <stuart.mills3 at btopenworld.com> wrote:

        Hi,

        Got a question regarding call at once combined with xml-cdr logging.

        The problem I seem to get is when I try sending calls to 2 destinations and 1 call gets answered, so that wins the race, the xml-cdr logging seems to be a little confused. In the app_log I?m getting logging up to the point of the dial string, then nothing after that for the a-leg. What seems to happen is the rest of the app_log gets written to the cdr file of the bleg that answered the call first.

        Is this by design or a bug in the xml-cdr logging?

        Thanks,

        Stuart

        _________________________________________________________________________
        Professional FreeSWITCH Consulting Services:
        consulting at freeswitch.org
        http://www.freeswitchsolutions.com

        
        

        Official FreeSWITCH Sites
        http://www.freeswitch.org
        http://wiki.freeswitch.org
        http://www.cluecon.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



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

      _________________________________________________________________________
      Professional FreeSWITCH Consulting Services:
      consulting at freeswitch.org
      http://www.freeswitchsolutions.com

      
      

      Official FreeSWITCH Sites
      http://www.freeswitch.org
      http://wiki.freeswitch.org
      http://www.cluecon.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


      _________________________________________________________________________
      Professional FreeSWITCH Consulting Services:
      consulting at freeswitch.org
      http://www.freeswitchsolutions.com

      
      

      Official FreeSWITCH Sites
      http://www.freeswitch.org
      http://wiki.freeswitch.org
      http://www.cluecon.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





------------------------------------------------------------------------------
  _________________________________________________________________________
  Professional FreeSWITCH Consulting Services:
  consulting at freeswitch.org
  http://www.freeswitchsolutions.com

  
  

  Official FreeSWITCH Sites
  http://www.freeswitch.org
  http://wiki.freeswitch.org
  http://www.cluecon.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


  _________________________________________________________________________
  Professional FreeSWITCH Consulting Services:
  consulting at freeswitch.org
  http://www.freeswitchsolutions.com

  
  

  Official FreeSWITCH Sites
  http://www.freeswitch.org
  http://wiki.freeswitch.org
  http://www.cluecon.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





-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org




--------------------------------------------------------------------------------
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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  Sun Aug  4 05:16:01 2013
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 3 Aug 2013 18:16:01 -0700
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <B782F8192B3E46608CE6666897F86BEB@PBPC>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>
	<1138329210A14BE4AEE53D7419427A15@PBPC>
	<CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
	<CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
	<3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
	<CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>
	<B782F8192B3E46608CE6666897F86BEB@PBPC>
Message-ID: <CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com>

No, but there is a test you need to pass to get in. Kinda like "Speak
friend and enter" if you get my meaning.

-MC
On Aug 3, 2013 5:21 PM, "Stuart Mills" <stuart.mills3 at btopenworld.com>
wrote:

>   yeah I can do that, although I've never used pastebin, do I need to
> register to use that?
>
>  *From:* Michael Collins <msc at freeswitch.org>
> *Sent:* Sunday, August 04, 2013 12:50 AM
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Subject:* Re: [Freeswitch-users] Call at once and XML-CDR
>
>  It might be best of you posted those on pastebin with explanations and
> console logs of the calls.
> -MC
>
>
> On Fri, Aug 2, 2013 at 3:24 PM, Stuart Mills <
> stuart.mills3 at btopenworld.com> wrote:
>
>>   Hi,
>>
>> Thanks for your reply.
>>
>> I have already changed the parameter log-b-leg to true in xml_cdr.conf
>> and I am getting 2 records for the outbound calls and 1 for inbound. The
>> issue is that the app_log section in the a_leg cdr seems to be split across
>> the a_leg cdr and the b_leg cdr that ?won? the race to be answered first.
>> The second b_leg has no duration as expected.
>>
>> Am I making sense?
>>
>> Regards,
>>
>> Stuart
>>
>>  *From:* Lloyd Aloysius <lloyd.aloysius at sunteltech.ca>
>> *Sent:* Friday, August 02, 2013 5:45 PM
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Subject:* Re: [Freeswitch-users] Call at once and XML-CDR
>>
>> Another note  look into the caller_profile and index. I think you have
>> two indexes for this situation.In your logic insert the records for the
>> multiple indexes.
>>
>> Lloyd
>>  *
>> *
>> **
>>
>>
>> On Fri, Aug 2, 2013 at 12:44 PM, Lloyd Aloysius <
>> lloyd.aloysius at sunteltech.ca> wrote:
>>
>>> With the default configuration switch only log a_leg.
>>>
>>> You need to enable b_leg in xml_cdr.conf.xml . Then you will see the two
>>> records in b_leg, with one get billsec > 0 for the the answered call.
>>>
>>> Lloyd
>>>
>>>
>>>
>>> On Fri, Aug 2, 2013 at 10:57 AM, Stuart Mills <
>>> stuart.mills3 at btopenworld.com> wrote:
>>>
>>>>   Hi Avi,
>>>>
>>>> Thanks for your reply.
>>>>
>>>> Yeah I'm sure, I'm using ESL and if I dial a single number it performs
>>>> as expected, it?s literally only when I'm making 2 or more calls at once it
>>>> happens.
>>>>
>>>> Cheers,
>>>>
>>>> Stuart
>>>>
>>>>  *From:* Avi Marcus <avi at avimarcus.net>
>>>> *Sent:* Friday, August 02, 2013 3:24 PM
>>>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>> *Subject:* Re: [Freeswitch-users] Call at once and XML-CDR
>>>>
>>>>
>>>> Are you sure these aren't b leg things?
>>>> Do you have an example of your dial plan and the logs?
>>>>
>>>> -Avi
>>>> On Aug 2, 2013 5:09 PM, "Stuart Mills" <stuart.mills3 at btopenworld.com>
>>>> wrote:
>>>>
>>>>>   Hi,
>>>>>
>>>>> Got a question regarding call at once combined with xml-cdr logging.
>>>>>
>>>>> The problem I seem to get is when I try sending calls to 2
>>>>> destinations and 1 call gets answered, so that wins the race, the xml-cdr
>>>>> logging seems to be a little confused. In the app_log I?m getting logging
>>>>> up to the point of the dial string, then nothing after that for the a-leg.
>>>>> What seems to happen is the rest of the app_log gets written to the cdr
>>>>> file of the bleg that answered the call first.
>>>>>
>>>>> Is this by design or a bug in the xml-cdr logging?
>>>>>
>>>>> Thanks,
>>>>>
>>>>> Stuart
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>> ------------------------------
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>  ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 intralanman at freeswitch.org  Sun Aug  4 05:23:36 2013
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Sat, 3 Aug 2013 21:23:36 -0400
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>
	<1138329210A14BE4AEE53D7419427A15@PBPC>
	<CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
	<CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
	<3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
	<CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>
	<B782F8192B3E46608CE6666897F86BEB@PBPC>
	<CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com>
Message-ID: <CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com>

On Aug 3, 2013 9:19 PM, "Michael Collins" <msc at freeswitch.org> wrote:
>
>  "Speak friend and enter"

+1

Love the LOTR reference :-)

-Ray
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From thomas.lee at octon.net  Sun Aug  4 05:25:46 2013
From: thomas.lee at octon.net (Thomas Lee)
Date: Sat, 3 Aug 2013 18:25:46 -0700 (PDT)
Subject: [Freeswitch-users] How to configure DTMF detection using 'info'
 and 'rfc2833' both?
In-Reply-To: <CAFArqsbP6tRhJsEv-WT2cv_Rq7sWr_eJQQPaYW0OkF4eCB0fzg@mail.gmail.com>
References: <1375526467565-7593558.post@n2.nabble.com>
	<CAFArqsbP6tRhJsEv-WT2cv_Rq7sWr_eJQQPaYW0OkF4eCB0fzg@mail.gmail.com>
Message-ID: <1375579546312-7593575.post@n2.nabble.com>

I'll try it out.....

Thank you.




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-configure-DTMF-detection-using-info-and-rfc2833-both-tp7593558p7593575.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From bdfoster at davri.com  Sun Aug  4 05:32:59 2013
From: bdfoster at davri.com (Brian Foster)
Date: Sat, 3 Aug 2013 21:32:59 -0400
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>
	<1138329210A14BE4AEE53D7419427A15@PBPC>
	<CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
	<CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
	<3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
	<CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>
	<B782F8192B3E46608CE6666897F86BEB@PBPC>
	<CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com>
	<CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com>
Message-ID: <CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com>

It's an "if you fail this simple test, part of your problem might be
attention to detail" sort of test.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 3, 2013 9:30 PM, "Raymond Chandler" <intralanman at freeswitch.org>
wrote:

>
> On Aug 3, 2013 9:19 PM, "Michael Collins" <msc at freeswitch.org> wrote:
> >
> >  "Speak friend and enter"
>
> +1
>
> Love the LOTR reference :-)
>
> -Ray
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jleung at v10networks.ca  Sun Aug  4 06:22:22 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Sat, 3 Aug 2013 19:22:22 -0700
Subject: [Freeswitch-users] garbled audio with G726-32,
	other codecs are fine
In-Reply-To: <CAHgk4BVNA+NOg5ChyYxSddDxhMxgSmWVe1tQ7NiCPRKo6M5BCw@mail.gmail.com>
References: <51FD0E8B.3060702@c3i.bg>
	<CAHgk4BVNA+NOg5ChyYxSddDxhMxgSmWVe1tQ7NiCPRKo6M5BCw@mail.gmail.com>
Message-ID: <003901ce90b9$749f0090$5ddd01b0$@v10networks.ca>

You can turn off G726 AAC bit-packing in spandsp.conf.xml.

 

By the way, there are other codecs out there you can try. SPEEX comes to
mind if all your endpoints don't deal with the PSTN.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
Foster
Sent: Saturday, August 3, 2013 2:37 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] garbled audio with G726-32, other codecs are
fine

 

AAC bitpacking by any chance? I thought I had a similar issue, happened so
long ago I cant remember what I did.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.

On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev at c3i.bg> wrote:

Hello

I'm migrating an office setup from asterisk to FS and in the process I
was considering using G726-32 for some bandwidth starved remote
endpoints. However I only get metallic/garbled audio with that codec
even when simply playing moh to the endpoint, while other codecs work
fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds
marginally better but still garbled and really worse than G711.

The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest
(centos6 64bit host). But please don't shoot ! :) - I know about virtual
environment limitations but for these tests the host is only lightly
loaded, there aren't any calls to the FS instance except my tests, and
the fact that it works with other codecs makes me think that
virtualization is not the issue here. I may be wrong though.

Is there any guide for debugging that kind of problem before reverting
to a fresh install on bare-metal with the latest HEAD ? Until now I've
tried:

- improving timers ; but the default soft timer (which I guess uses
timerd) works best. The time interval between sent packets on a tcpdump
trace looks identical to the output of "timer_test", so that doesn't
seem to be a network/jitter problem. And there's no problem with other
codecs, but maybe G726-XX is specific. For info the guest's clocksource
is kvm_clock, while the host uses tsc.

- using different endpoints: the production ones are Linksys PAP2
("fixed" for 20ms psize, and G726-32 SDP type indentification), but the
same thing happens with linphone on a fedora 19 laptop.

A call with rtp media going through FS without transcoding - G726-32 to
G726-32 - works perfectly (I can't hear the difference with G711). The
problem is only when there's transcoding to G726 (from wav for moh, or
from any other codec when bridging). I've looked at the wiki, posts,
changelogs, jira, ..., but am a bit at a loss now.

Any pointers ?

Except that little problem, FS rocks, and I'm happy I can finally ditch
asterisk. Kudos to the core devs and contributors.

Ivan


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 lists at kavun.ch  Sun Aug  4 06:34:04 2013
From: lists at kavun.ch (Emrah)
Date: Sat, 3 Aug 2013 22:34:04 -0400
Subject: [Freeswitch-users] Possible Polycom bridging issues in all
	recent FreeSWITCH versions
In-Reply-To: <CAMwZ-tJ4sEfyeT=VGNMfcWm=sZK1jeB2mBT8JA8RAkvtVCqVpQ@mail.gmail.com>
References: <17D13B62-FDCC-462F-A376-65BC8DE474C5@kavun.ch>
	<CAAtWKeEvqNC_9D4L7PjKSkmgzrOryqQ3ER6Z7dqM4Fqi52JAbA@mail.gmail.com>
	<9CD85028-3064-4439-A846-705FED3F16D2@kavun.ch>
	<BAD9DC74-5691-4688-91A1-C2C9CD17D34C@kavun.ch>
	<AAF4E78A-94C6-4CA2-A93F-7D03E5DE2C16@kavun.ch>
	<CAMwZ-tJ4sEfyeT=VGNMfcWm=sZK1jeB2mBT8JA8RAkvtVCqVpQ@mail.gmail.com>
Message-ID: <BEF9D0E4-EC99-4724-99D4-72A6DB09551C@kavun.ch>

I haven't checked the link yet, but I like the response already. :) I actually had a pretty good treatment if my memory serves well. 
Thanks and I'll make sure I get it through the process.

Cheers
On Aug 3, 2013, at 7:12 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sat, Aug 3, 2013 at 3:31 PM, Emrah <lists at kavun.ch> wrote:
>> I've been going in circles for quite some time now as you can see in the history of this message. And this is starting to weigh heavy.
>> 
>> The bug has made it into stable versions and I would love to see it fixed soon? What is needed to troubleshoot on my end?
> 
> This is what you need to know:
> https://wiki.freeswitch.org/wiki/Reporting_Bugs
> 
> 
> Best,
> Gabe
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 max at nysolutions.com  Sun Aug  4 06:43:27 2013
From: max at nysolutions.com (Moishe Grunstein)
Date: Sun, 4 Aug 2013 02:43:27 +0000
Subject: [Freeswitch-users] Possible Polycom bridging issues in
	all	recent FreeSWITCH versions
In-Reply-To: <BEF9D0E4-EC99-4724-99D4-72A6DB09551C@kavun.ch>
References: <17D13B62-FDCC-462F-A376-65BC8DE474C5@kavun.ch>
	<CAAtWKeEvqNC_9D4L7PjKSkmgzrOryqQ3ER6Z7dqM4Fqi52JAbA@mail.gmail.com>
	<9CD85028-3064-4439-A846-705FED3F16D2@kavun.ch>
	<BAD9DC74-5691-4688-91A1-C2C9CD17D34C@kavun.ch>
	<AAF4E78A-94C6-4CA2-A93F-7D03E5DE2C16@kavun.ch>
	<CAMwZ-tJ4sEfyeT=VGNMfcWm=sZK1jeB2mBT8JA8RAkvtVCqVpQ@mail.gmail.com>
	<BEF9D0E4-EC99-4724-99D4-72A6DB09551C@kavun.ch>
Message-ID: <BA6A05D48E282C4C9989496521266BA1028B431CB1@ex2.connex.local>

I have seen similar issue, however I always worked around it, and never really analyzed the packets to see if the bug is in Freeswitch or with Polycom, I have not used Video so doubt this is Video related.


Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com 
Polycom Certified VAR
Microsoft Small Business Specialist, Cisco SMB Select Certified

Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Emrah
Sent: Saturday, August 03, 2013 10:34 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Possible Polycom bridging issues in all recent FreeSWITCH versions

I haven't checked the link yet, but I like the response already. :) I actually had a pretty good treatment if my memory serves well. 
Thanks and I'll make sure I get it through the process.

Cheers
On Aug 3, 2013, at 7:12 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Sat, Aug 3, 2013 at 3:31 PM, Emrah <lists at kavun.ch> wrote:
>> I've been going in circles for quite some time now as you can see in the history of this message. And this is starting to weigh heavy.
>> 
>> The bug has made it into stable versions and I would love to see it fixed soon... What is needed to troubleshoot on my end?
> 
> This is what you need to know:
> https://wiki.freeswitch.org/wiki/Reporting_Bugs
> 
> 
> Best,
> Gabe
> 
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
>  
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-use
> rs
> http://www.freeswitch.org


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 frankjr at mcpeekdodge.com  Sun Aug  4 07:51:11 2013
From: frankjr at mcpeekdodge.com (Frank Busalacchi Jr)
Date: Sun, 4 Aug 2013 03:51:11 +0000
Subject: [Freeswitch-users] Call Intercept
Message-ID: <C066384B0C1F1E4B9B870F050D7F1A9D44B103EC@ADAMA.mcpeekdodge.local>

Hi everyone!  First and foremost, thanks for a great piece of software, and all the time you guys still put into this program day in and day out.  Thank you!

I'm having difficulty implementing an "intercept" system, and I'm thinking that I have to be "conceptually" misunderstanding something.  Towards those ends, I'm hoping someone can get me back on track, I'm sick of banging my head on the wall to try to make this work.

So here is my scenario:

1. A call (Call A) comes into the server from my voip provider on a DID.

2. My XML dialplan transfers the call to an extension in the XML dialplan of 1111.

3. My XML dialplan recognizes 1111 as a local extension, and executes a LUA script to handle the actual bridging of Call A to the appropriate devices.  I use the LUA script because I do some logic to decide whether to ring just the user's SIP phone, the user's cell phone, or a combination of both simultaneously.  Bottom line is that the LUA script does something like this after building the originate string in dialString:

session2 = freeswitch.Session(dialString);

   if session2:answered() == true then
      freeswitch.bridge(session,session2);
      return true;
   else
      return false;
   end

---All of the above works as expected ---

My issue is that when the LUA script starts ringing the sip phone registered as 1111, I want to 'intercept' that call from a different extension.  So here is what I do:

1.  Extension 1112 hears that his co-worker's extension 1111 is ringing, and wants to answer it.  He dials **1111

2.  My XML dialplan recognizes the request to pickup/intercept the call ringing at 1111, and calls a LUA script to figure out the uuid of the ringing phone.

3.  The LUA script sets a channel variable with the UUID it choose as the appropriate candidate to intercept, and exits.

4.  The XML dialplan resumes control, and does a <action application="intercept" data="${intercept-uuid}"/> .

5.  The calls aren't intercepted/bridged.

A pastebin of the log is here: http://pastebin.freeswitch.org/21267

What am I missing conceptually here?


Thanks,
Frank

Frank Busalacchi Jr
President
Direct (714) 254-2612
[Description: C:\Users\frankjr\Downloads\McPeekEmailLogo.jpg]

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From itsme.kunnu at gmail.com  Sun Aug  4 11:03:08 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 4 Aug 2013 00:03:08 -0700
Subject: [Freeswitch-users] Freeswitch installation
Message-ID: <CAMvhjJvS+_PA_Ou2WKxPnscp31y+7U8Fx5K0ZC3rgqLi6PnwiQ@mail.gmail.com>

Sir i have downloaded the msi file for windows of freeswitch and also ran
the setup file to install freeswitch...but when i double click on the
freeswitchconsole.exe file from the windows explorer it automatically
closes...kindly help with this...
Ashish mishra
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From jleung at v10networks.ca  Sun Aug  4 11:06:12 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Sun, 4 Aug 2013 00:06:12 -0700
Subject: [Freeswitch-users] Freeswitch installation
In-Reply-To: <CAMvhjJvS+_PA_Ou2WKxPnscp31y+7U8Fx5K0ZC3rgqLi6PnwiQ@mail.gmail.com>
References: <CAMvhjJvS+_PA_Ou2WKxPnscp31y+7U8Fx5K0ZC3rgqLi6PnwiQ@mail.gmail.com>
Message-ID: <004d01ce90e1$1b78b560$526a2020$@v10networks.ca>

Install the VC2012 and the VC2010 Runtime before you install the FreeSWITCH
windows binaries

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ashish
Mishra
Sent: Sunday, August 4, 2013 12:03 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Freeswitch installation

 

Sir i have downloaded the msi file for windows of freeswitch and also ran
the setup file to install freeswitch...but when i double click on the
freeswitchconsole.exe file from the windows explorer it automatically
closes...kindly help with this...
Ashish mishra

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From itsme.kunnu at gmail.com  Sun Aug  4 11:12:36 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 4 Aug 2013 00:12:36 -0700
Subject: [Freeswitch-users] Freeswitch installation
In-Reply-To: <004d01ce90e1$1b78b560$526a2020$@v10networks.ca>
References: <CAMvhjJvS+_PA_Ou2WKxPnscp31y+7U8Fx5K0ZC3rgqLi6PnwiQ@mail.gmail.com>
	<004d01ce90e1$1b78b560$526a2020$@v10networks.ca>
Message-ID: <CAMvhjJuAkxgmo=VS8YNayZvCR8gK1F1BW_JZOXJ4wwSrR7Xx9A@mail.gmail.com>

Sir i have already installed VC2010 EXPRESS...
On Aug 4, 2013 12:40 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:

> Install the VC2012 and the VC2010 Runtime before you install the
> FreeSWITCH windows binaries****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish
> Mishra
> *Sent:* Sunday, August 4, 2013 12:03 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Freeswitch installation****
>
> ** **
>
> Sir i have downloaded the msi file for windows of freeswitch and also ran
> the setup file to install freeswitch...but when i double click on the
> freeswitchconsole.exe file from the windows explorer it automatically
> closes...kindly help with this...
> Ashish mishra****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Sun Aug  4 11:37:12 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 4 Aug 2013 00:37:12 -0700
Subject: [Freeswitch-users] Freeswitch installation
In-Reply-To: <CAMvhjJuAkxgmo=VS8YNayZvCR8gK1F1BW_JZOXJ4wwSrR7Xx9A@mail.gmail.com>
References: <CAMvhjJvS+_PA_Ou2WKxPnscp31y+7U8Fx5K0ZC3rgqLi6PnwiQ@mail.gmail.com>
	<004d01ce90e1$1b78b560$526a2020$@v10networks.ca>
	<CAMvhjJuAkxgmo=VS8YNayZvCR8gK1F1BW_JZOXJ4wwSrR7Xx9A@mail.gmail.com>
Message-ID: <CAMvhjJux-ZF8csGV9AJVPwpk-yBgn_s0WAbjenbBy3jQDTEE+g@mail.gmail.com>

Sir can you tell me how to launch freeswitch from freeswitch binaries that
i have installed...???

Ashish
On Aug 4, 2013 12:42 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:

> Sir i have already installed VC2010 EXPRESS...
> On Aug 4, 2013 12:40 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:
>
>> Install the VC2012 and the VC2010 Runtime before you install the
>> FreeSWITCH windows binaries****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish
>> Mishra
>> *Sent:* Sunday, August 4, 2013 12:03 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] Freeswitch installation****
>>
>> ** **
>>
>> Sir i have downloaded the msi file for windows of freeswitch and also ran
>> the setup file to install freeswitch...but when i double click on the
>> freeswitchconsole.exe file from the windows explorer it automatically
>> closes...kindly help with this...
>> Ashish mishra****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 jleung at v10networks.ca  Sun Aug  4 11:37:11 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Sun, 4 Aug 2013 00:37:11 -0700
Subject: [Freeswitch-users] Freeswitch installation
In-Reply-To: <CAMvhjJuAkxgmo=VS8YNayZvCR8gK1F1BW_JZOXJ4wwSrR7Xx9A@mail.gmail.com>
References: <CAMvhjJvS+_PA_Ou2WKxPnscp31y+7U8Fx5K0ZC3rgqLi6PnwiQ@mail.gmail.com>	<004d01ce90e1$1b78b560$526a2020$@v10networks.ca>
	<CAMvhjJuAkxgmo=VS8YNayZvCR8gK1F1BW_JZOXJ4wwSrR7Xx9A@mail.gmail.com>
Message-ID: <005801ce90e5$6f12b2d0$4d381870$@v10networks.ca>

Well, run it under a command prompt window and see what the console is
telling you. I.e. do not start it up from Windows Explorer.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ashish
Mishra
Sent: Sunday, August 4, 2013 12:13 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Freeswitch installation

 

Sir i have already installed VC2010 EXPRESS...

On Aug 4, 2013 12:40 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:

Install the VC2012 and the VC2010 Runtime before you install the FreeSWITCH
windows binaries

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ashish
Mishra
Sent: Sunday, August 4, 2013 12:03 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Freeswitch installation

 

Sir i have downloaded the msi file for windows of freeswitch and also ran
the setup file to install freeswitch...but when i double click on the
freeswitchconsole.exe file from the windows explorer it automatically
closes...kindly help with this...
Ashish mishra


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 itsme.kunnu at gmail.com  Sun Aug  4 13:43:47 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 4 Aug 2013 02:43:47 -0700
Subject: [Freeswitch-users] Freeswitch installation
In-Reply-To: <005801ce90e5$6f12b2d0$4d381870$@v10networks.ca>
References: <CAMvhjJvS+_PA_Ou2WKxPnscp31y+7U8Fx5K0ZC3rgqLi6PnwiQ@mail.gmail.com>
	<004d01ce90e1$1b78b560$526a2020$@v10networks.ca>
	<CAMvhjJuAkxgmo=VS8YNayZvCR8gK1F1BW_JZOXJ4wwSrR7Xx9A@mail.gmail.com>
	<005801ce90e5$6f12b2d0$4d381870$@v10networks.ca>
Message-ID: <CAMvhjJv9u5mcUDPxUch0mNsEZ_o1POuP1quM9uUPMM7K2hgtWA@mail.gmail.com>

Sir i tried running it from cmd prompt and it showed me
Cannot open pid file C:/Program Files/FreeSWITCH/run/freeswitch.pid
On Aug 4, 2013 1:13 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:

> Well, run it under a command prompt window and see what the console is
> telling you. I.e. do not start it up from Windows Explorer.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish
> Mishra
> *Sent:* Sunday, August 4, 2013 12:13 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Freeswitch installation****
>
> ** **
>
> Sir i have already installed VC2010 EXPRESS...****
>
> On Aug 4, 2013 12:40 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:****
>
> Install the VC2012 and the VC2010 Runtime before you install the
> FreeSWITCH windows binaries****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish
> Mishra
> *Sent:* Sunday, August 4, 2013 12:03 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Freeswitch installation****
>
>  ****
>
> Sir i have downloaded the msi file for windows of freeswitch and also ran
> the setup file to install freeswitch...but when i double click on the
> freeswitchconsole.exe file from the windows explorer it automatically
> closes...kindly help with this...
> Ashish mishra****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 cmrienzo at gmail.com  Sun Aug  4 14:49:52 2013
From: cmrienzo at gmail.com (cmrienzo at gmail.com)
Date: Sun, 4 Aug 2013 06:49:52 -0400
Subject: [Freeswitch-users] Call Intercept
In-Reply-To: <C066384B0C1F1E4B9B870F050D7F1A9D44B103EC@ADAMA.mcpeekdodge.local>
References: <C066384B0C1F1E4B9B870F050D7F1A9D44B103EC@ADAMA.mcpeekdodge.local>
Message-ID: <7D5E4145-4B7E-4856-80A7-D93A31258AD4@gmail.com>

That channel variable was fetched in the routing state before the lua application was executed.   You can either transfer to a new extension to execute the intercept or execute the intercept in the lua script. 

Chris


On Aug 3, 2013, at 11:51 PM, Frank Busalacchi Jr <frankjr at mcpeekdodge.com> wrote:

> Hi everyone!  First and foremost, thanks for a great piece of software, and all the time you guys still put into this program day in and day out.  Thank you!
>  
> I'm having difficulty implementing an "intercept" system, and I'm thinking that I have to be "conceptually" misunderstanding something.  Towards those ends, I'm hoping someone can get me back on track, I?m sick of banging my head on the wall to try to make this work.
>  
> So here is my scenario:
>  
> 1. A call (Call A) comes into the server from my voip provider on a DID.
>  
> 2. My XML dialplan transfers the call to an extension in the XML dialplan of 1111.
>  
> 3. My XML dialplan recognizes 1111 as a local extension, and executes a LUA script to handle the actual bridging of Call A to the appropriate devices.  I use the LUA script because I do some logic to decide whether to ring just the user's SIP phone, the user's cell phone, or a combination of both simultaneously.  Bottom line is that the LUA script does something like this after building the originate string in dialString:
>  
> session2 = freeswitch.Session(dialString);
>  
>    if session2:answered() == true then
>       freeswitch.bridge(session,session2);
>       return true;
>    else
>       return false;
>    end
>  
> ---All of the above works as expected ---
>  
> My issue is that when the LUA script starts ringing the sip phone registered as 1111, I want to 'intercept' that call from a different extension.  So here is what I do:
>  
> 1.  Extension 1112 hears that his co-worker's extension 1111 is ringing, and wants to answer it.  He dials **1111
>  
> 2.  My XML dialplan recognizes the request to pickup/intercept the call ringing at 1111, and calls a LUA script to figure out the uuid of the ringing phone.
>  
> 3.  The LUA script sets a channel variable with the UUID it choose as the appropriate candidate to intercept, and exits.
>  
> 4.  The XML dialplan resumes control, and does a <action application="intercept" data="${intercept-uuid}"/> .
>  
> 5.  The calls aren't intercepted/bridged.
>  
> A pastebin of the log is here: http://pastebin.freeswitch.org/21267
>  
> What am I missing conceptually here?
>  
>  
> Thanks,
> Frank
>  
> Frank Busalacchi Jr        
> President
> Direct (714) 254-2612
> <image001.jpg>
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Sun Aug  4 15:57:23 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 4 Aug 2013 04:57:23 -0700
Subject: [Freeswitch-users] Software phone
Message-ID: <CAMvhjJs4403+LxEKMj1VnQw6bNBdGFpEMWaTORu5fvixiPGGvg@mail.gmail.com>

Hello, in order to use a softphone do i need to install freeswitch on the
client machine also...or simply i can download the softphone and can
connect to a server machine on which freeswitch is already installed...???
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From max at nysolutions.com  Sun Aug  4 17:49:21 2013
From: max at nysolutions.com (Moishe Grunstein)
Date: Sun, 4 Aug 2013 13:49:21 +0000
Subject: [Freeswitch-users] Software phone
In-Reply-To: <CAMvhjJs4403+LxEKMj1VnQw6bNBdGFpEMWaTORu5fvixiPGGvg@mail.gmail.com>
References: <CAMvhjJs4403+LxEKMj1VnQw6bNBdGFpEMWaTORu5fvixiPGGvg@mail.gmail.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA1028B433B5F@ex2.connex.local>

Just install softphone on other machine and connect, if you are referring to the freeswitch softphone it is a separate binary.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ashish Mishra
Sent: Sunday, August 4, 2013 7:57 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Software phone


Hello, in order to use a softphone do i need to install freeswitch on the client machine also...or simply i can download the softphone and can connect to a server machine on which freeswitch is already installed...???
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From itsme.kunnu at gmail.com  Sun Aug  4 18:04:40 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 4 Aug 2013 07:04:40 -0700
Subject: [Freeswitch-users] Software phone
In-Reply-To: <BA6A05D48E282C4C9989496521266BA1028B433B5F@ex2.connex.local>
References: <CAMvhjJs4403+LxEKMj1VnQw6bNBdGFpEMWaTORu5fvixiPGGvg@mail.gmail.com>
	<BA6A05D48E282C4C9989496521266BA1028B433B5F@ex2.connex.local>
Message-ID: <CAMvhjJs5kP+=rhvgE9KpVKw+HbDQSYATgTp4cJkdU5a9psGGZw@mail.gmail.com>

Sir thank you for your reply...can u help me with how to enable freeswitch
to use gsm modem..
On Aug 4, 2013 7:22 PM, "Moishe Grunstein" <max at nysolutions.com> wrote:

> Just install softphone on other machine and connect, if you are referring
> to the freeswitch softphone it is a separate binary.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish
> Mishra
> *Sent:* Sunday, August 4, 2013 7:57 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Software phone****
>
> ** **
>
> Hello, in order to use a softphone do i need to install freeswitch on the
> client machine also...or simply i can download the softphone and can
> connect to a server machine on which freeswitch is already installed...???
> ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lloyd.aloysius at gmail.com  Sun Aug  4 18:07:24 2013
From: lloyd.aloysius at gmail.com (Lloyd Aloysius)
Date: Sun, 4 Aug 2013 10:07:24 -0400
Subject: [Freeswitch-users] Call Intercept
In-Reply-To: <7D5E4145-4B7E-4856-80A7-D93A31258AD4@gmail.com>
References: <C066384B0C1F1E4B9B870F050D7F1A9D44B103EC@ADAMA.mcpeekdodge.local>
	<7D5E4145-4B7E-4856-80A7-D93A31258AD4@gmail.com>
Message-ID: <CAGvuWYa4oZfTcVjn=RpKK4yyPDEhB03gbtd_kiSW=8as0PFhiQ@mail.gmail.com>

Post your lua script here

before the bridge(How to set the UUID for intercept) and intercept part

---

I use this way and it always works
session:execute("set","intercept_unbridged_only=true")

Lloyd


On Sun, Aug 4, 2013 at 6:49 AM, <cmrienzo at gmail.com> wrote:

> That channel variable was fetched in the routing state before the lua
> application was executed.   You can either transfer to a new extension to
> execute the intercept or execute the intercept in the lua script.
>
> Chris
>
>
> On Aug 3, 2013, at 11:51 PM, Frank Busalacchi Jr <frankjr at mcpeekdodge.com>
> wrote:
>
>  Hi everyone!  First and foremost, thanks for a great piece of software,
> and all the time you guys still put into this program day in and day out.
> Thank you!****
>
> ** **
>
> I'm having difficulty implementing an "intercept" system, and I'm thinking
> that I have to be "conceptually" misunderstanding something.  Towards those
> ends, I'm hoping someone can get me back on track, I?m sick of banging my
> head on the wall to try to make this work.****
>
> ** **
>
> So here is my scenario:****
>
> ** **
>
> 1. A call (Call A) comes into the server from my voip provider on a DID.**
> **
>
> ** **
>
> 2. My XML dialplan transfers the call to an extension in the XML dialplan
> of 1111.****
>
> ** **
>
> 3. My XML dialplan recognizes 1111 as a local extension, and executes a
> LUA script to handle the actual bridging of Call A to the appropriate
> devices.  I use the LUA script because I do some logic to decide whether to
> ring just the user's SIP phone, the user's cell phone, or a combination of
> both simultaneously.  Bottom line is that the LUA script does something
> like this after building the originate string in dialString:****
>
> ** **
>
> session2 = freeswitch.Session(dialString);****
>
> ** **
>
>    if session2:answered() == true then****
>
>       freeswitch.bridge(session,session2);****
>
>       return true;****
>
>    else****
>
>       return false;****
>
>    end****
>
> ** **
>
> ---All of the above works as expected ---****
>
> ** **
>
> My issue is that when the LUA script starts ringing the sip phone
> registered as 1111, I want to 'intercept' that call from a different
> extension.  So here is what I do:****
>
> ** **
>
> 1.  Extension 1112 hears that his co-worker's extension 1111 is ringing,
> and wants to answer it.  He dials **1111****
>
> ** **
>
> 2.  My XML dialplan recognizes the request to pickup/intercept the call
> ringing at 1111, and calls a LUA script to figure out the uuid of the
> ringing phone.****
>
> ** **
>
> 3.  The LUA script sets a channel variable with the UUID it choose as the
> appropriate candidate to intercept, and exits.****
>
> ** **
>
> 4.  The XML dialplan resumes control, and does a <action
> application="intercept" data="${intercept-uuid}"/> .****
>
> ** **
>
> 5.  The calls aren't intercepted/bridged.****
>
> ** **
>
> A pastebin of the log is here: http://pastebin.freeswitch.org/21267 ****
>
> ** **
>
> What am I missing conceptually here?****
>
> ** **
>
> ** **
>
> Thanks,****
>
> Frank****
>
> ** **
>
> *Frank Busalacchi Jr         ***
>
> *President*****
>
> *Direct (714) 254-2612*****
>
> <image001.jpg>****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 rafal.gwizdala at gmail.com  Sun Aug  4 18:14:07 2013
From: rafal.gwizdala at gmail.com (Rafal Gwizdala)
Date: Sun, 4 Aug 2013 16:14:07 +0200
Subject: [Freeswitch-users] Freeswitch installation
In-Reply-To: <CAMvhjJv9u5mcUDPxUch0mNsEZ_o1POuP1quM9uUPMM7K2hgtWA@mail.gmail.com>
References: <CAMvhjJvS+_PA_Ou2WKxPnscp31y+7U8Fx5K0ZC3rgqLi6PnwiQ@mail.gmail.com>
	<004d01ce90e1$1b78b560$526a2020$@v10networks.ca>
	<CAMvhjJuAkxgmo=VS8YNayZvCR8gK1F1BW_JZOXJ4wwSrR7Xx9A@mail.gmail.com>
	<005801ce90e5$6f12b2d0$4d381870$@v10networks.ca>
	<CAMvhjJv9u5mcUDPxUch0mNsEZ_o1POuP1quM9uUPMM7K2hgtWA@mail.gmail.com>
Message-ID: <CALtno0-EZ9kKjAUo2qOXgWcTH7ADK750FXK1go0L9i8wjg9UGg@mail.gmail.com>

Run the console in administrator mode (right click on command line/run as
administrator). Then start freeswitch.console.exe
or just configure FS as a windows service under local system account


On Sun, Aug 4, 2013 at 11:43 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Sir i tried running it from cmd prompt and it showed me
> Cannot open pid file C:/Program Files/FreeSWITCH/run/freeswitch.pid
> On Aug 4, 2013 1:13 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:
>
>> Well, run it under a command prompt window and see what the console is
>> telling you. I.e. do not start it up from Windows Explorer.****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish
>> Mishra
>> *Sent:* Sunday, August 4, 2013 12:13 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Freeswitch installation****
>>
>> ** **
>>
>> Sir i have already installed VC2010 EXPRESS...****
>>
>> On Aug 4, 2013 12:40 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:****
>>
>> Install the VC2012 and the VC2010 Runtime before you install the
>> FreeSWITCH windows binaries****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish
>> Mishra
>> *Sent:* Sunday, August 4, 2013 12:03 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] Freeswitch installation****
>>
>>  ****
>>
>> Sir i have downloaded the msi file for windows of freeswitch and also ran
>> the setup file to install freeswitch...but when i double click on the
>> freeswitchconsole.exe file from the windows explorer it automatically
>> closes...kindly help with this...
>> Ashish mishra****
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Sun Aug  4 18:22:26 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 4 Aug 2013 07:22:26 -0700
Subject: [Freeswitch-users] Freeswitch installation
In-Reply-To: <CALtno0-EZ9kKjAUo2qOXgWcTH7ADK750FXK1go0L9i8wjg9UGg@mail.gmail.com>
References: <CAMvhjJvS+_PA_Ou2WKxPnscp31y+7U8Fx5K0ZC3rgqLi6PnwiQ@mail.gmail.com>
	<004d01ce90e1$1b78b560$526a2020$@v10networks.ca>
	<CAMvhjJuAkxgmo=VS8YNayZvCR8gK1F1BW_JZOXJ4wwSrR7Xx9A@mail.gmail.com>
	<005801ce90e5$6f12b2d0$4d381870$@v10networks.ca>
	<CAMvhjJv9u5mcUDPxUch0mNsEZ_o1POuP1quM9uUPMM7K2hgtWA@mail.gmail.com>
	<CALtno0-EZ9kKjAUo2qOXgWcTH7ADK750FXK1go0L9i8wjg9UGg@mail.gmail.com>
Message-ID: <CAMvhjJsy_aEJsXz9t7NHvGBHKi_VaF4Vu=ginej8+-Z8=yyCSQ@mail.gmail.com>

Should i simply run the cmd in admin mode and type freeswitch.console.exe
and press enter or do i have to type something else...???
On Aug 4, 2013 7:48 PM, "Rafal Gwizdala" <rafal.gwizdala at gmail.com> wrote:

> Run the console in administrator mode (right click on command line/run as
> administrator). Then start freeswitch.console.exe
> or just configure FS as a windows service under local system account
>
>
> On Sun, Aug 4, 2013 at 11:43 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Sir i tried running it from cmd prompt and it showed me
>> Cannot open pid file C:/Program Files/FreeSWITCH/run/freeswitch.pid
>> On Aug 4, 2013 1:13 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:
>>
>>> Well, run it under a command prompt window and see what the console is
>>> telling you. I.e. do not start it up from Windows Explorer.****
>>>
>>> ** **
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish
>>> Mishra
>>> *Sent:* Sunday, August 4, 2013 12:13 AM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] Freeswitch installation****
>>>
>>> ** **
>>>
>>> Sir i have already installed VC2010 EXPRESS...****
>>>
>>> On Aug 4, 2013 12:40 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:****
>>>
>>> Install the VC2012 and the VC2010 Runtime before you install the
>>> FreeSWITCH windows binaries****
>>>
>>>  ****
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish
>>> Mishra
>>> *Sent:* Sunday, August 4, 2013 12:03 AM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* [Freeswitch-users] Freeswitch installation****
>>>
>>>  ****
>>>
>>> Sir i have downloaded the msi file for windows of freeswitch and also
>>> ran the setup file to install freeswitch...but when i double click on the
>>> freeswitchconsole.exe file from the windows explorer it automatically
>>> closes...kindly help with this...
>>> Ashish mishra****
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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****
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Sun Aug  4 18:30:59 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 4 Aug 2013 07:30:59 -0700
Subject: [Freeswitch-users] Freeswitch installation
In-Reply-To: <CAMvhjJuq=xY9ySsmNDFAxZAvk6mci4yZ4UeC4-bU7D1GTUgGMQ@mail.gmail.com>
References: <CAMvhjJvS+_PA_Ou2WKxPnscp31y+7U8Fx5K0ZC3rgqLi6PnwiQ@mail.gmail.com>
	<004d01ce90e1$1b78b560$526a2020$@v10networks.ca>
	<CAMvhjJuAkxgmo=VS8YNayZvCR8gK1F1BW_JZOXJ4wwSrR7Xx9A@mail.gmail.com>
	<005801ce90e5$6f12b2d0$4d381870$@v10networks.ca>
	<CAMvhjJv9u5mcUDPxUch0mNsEZ_o1POuP1quM9uUPMM7K2hgtWA@mail.gmail.com>
	<CALtno0-EZ9kKjAUo2qOXgWcTH7ADK750FXK1go0L9i8wjg9UGg@mail.gmail.com>
	<CAMvhjJuq=xY9ySsmNDFAxZAvk6mci4yZ4UeC4-bU7D1GTUgGMQ@mail.gmail.com>
Message-ID: <CAMvhjJtd_PJUN2K3eGY-f0Um_3hiPPm9UREck5b5rf8=Qb9bEQ@mail.gmail.com>

Thank you sir...problem solved...but can you help me on how to enable
freeswitch to use gsm modem
On Aug 4, 2013 7:48 PM, "Rafal Gwizdala" <rafal.gwizdala at gmail.com> wrote:

Run the console in administrator mode (right click on command line/run as
administrator). Then start freeswitch.console.exe
or just configure FS as a windows service under local system account


On Sun, Aug 4, 2013 at 11:43 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Sir i tried running it from cmd prompt and it showed me
> Cannot open pid file C:/Program Files/FreeSWITCH/run/freeswitch.pid
> On Aug 4, 2013 1:13 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:
>
>> Well, run it under a command prompt window and see what the console is
>> telling you. I.e. do not start it up from Windows Explorer.****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish
>> Mishra
>> *Sent:* Sunday, August 4, 2013 12:13 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Freeswitch installation****
>>
>> ** **
>>
>> Sir i have already installed VC2010 EXPRESS...****
>>
>> On Aug 4, 2013 12:40 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:****
>>
>> Install the VC2012 and the VC2010 Runtime before you install the
>> FreeSWITCH windows binaries****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish
>> Mishra
>> *Sent:* Sunday, August 4, 2013 12:03 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] Freeswitch installation****
>>
>>  ****
>>
>> Sir i have downloaded the msi file for windows of freeswitch and also ran
>> the setup file to install freeswitch...but when i double click on the
>> freeswitchconsole.exe file from the windows explorer it automatically
>> closes...kindly help with this...
>> Ashish mishra****
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 imitev at c3i.bg  Sun Aug  4 09:53:31 2013
From: imitev at c3i.bg (Ivan Mitev)
Date: Sun, 04 Aug 2013 08:53:31 +0300
Subject: [Freeswitch-users] garbled audio with G726-32,
 other codecs are fine
In-Reply-To: <003901ce90b9$749f0090$5ddd01b0$@v10networks.ca>
References: <51FD0E8B.3060702@c3i.bg>
	<CAHgk4BVNA+NOg5ChyYxSddDxhMxgSmWVe1tQ7NiCPRKo6M5BCw@mail.gmail.com>
	<003901ce90b9$749f0090$5ddd01b0$@v10networks.ca>
Message-ID: <51FDEC5B.2010307@c3i.bg>

Thanks for the suggestion but I'm testing with G726-32, not AAL2-G726-32 
; so bitpacking shouldn't be used. By the way when I tested with AAL2 to 
the linphone client I only got cracks and whitenoise, I've forgot to 
mention that in my post.

That said I've tried to uncomment and set <param name="bitpacking" 
value="none"/> in internal.xml ("none" is a wild guess - I couldn't find 
any doc on values accepted by this parameter), but that doesn't help.

Speex: the ATAs don't support it. And being stubborn I'd like to 
understand what's wrong with G726 :)


On 08/04/2013 05:22 AM, Jeff Leung wrote:
>
> You can turn off G726 AAC bit-packing in spandsp.conf.xml.
>
> By the way, there are other codecs out there you can try. SPEEX comes 
> to mind if all your endpoints don?t deal with the PSTN.
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Brian Foster
> *Sent:* Saturday, August 3, 2013 2:37 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] garbled audio with G726-32, other 
> codecs are fine
>
> AAC bitpacking by any chance? I thought I had a similar issue, 
> happened so long ago I cant remember what I did.
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com <mailto:bdfoster at davri.com>
> Indianapolis, Indiana
>
> Sent from a mobile device.
>
> On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev at c3i.bg 
> <mailto:imitev at c3i.bg>> wrote:
>
> Hello
>
> I'm migrating an office setup from asterisk to FS and in the process I
> was considering using G726-32 for some bandwidth starved remote
> endpoints. However I only get metallic/garbled audio with that codec
> even when simply playing moh to the endpoint, while other codecs work
> fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds
> marginally better but still garbled and really worse than G711.
>
> The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest
> (centos6 64bit host). But please don't shoot ! :) - I know about virtual
> environment limitations but for these tests the host is only lightly
> loaded, there aren't any calls to the FS instance except my tests, and
> the fact that it works with other codecs makes me think that
> virtualization is not the issue here. I may be wrong though.
>
> Is there any guide for debugging that kind of problem before reverting
> to a fresh install on bare-metal with the latest HEAD ? Until now I've
> tried:
>
> - improving timers ; but the default soft timer (which I guess uses
> timerd) works best. The time interval between sent packets on a tcpdump
> trace looks identical to the output of "timer_test", so that doesn't
> seem to be a network/jitter problem. And there's no problem with other
> codecs, but maybe G726-XX is specific. For info the guest's clocksource
> is kvm_clock, while the host uses tsc.
>
> - using different endpoints: the production ones are Linksys PAP2
> ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the
> same thing happens with linphone on a fedora 19 laptop.
>
> A call with rtp media going through FS without transcoding - G726-32 to
> G726-32 - works perfectly (I can't hear the difference with G711). The
> problem is only when there's transcoding to G726 (from wav for moh, or
> from any other codec when bridging). I've looked at the wiki, posts,
> changelogs, jira, ..., but am a bit at a loss now.
>
> Any pointers ?
>
> Except that little problem, FS rocks, and I'm happy I can finally ditch
> asterisk. Kudos to the core devs and contributors.
>
> Ivan
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 wampir990 at gmail.com  Sun Aug  4 16:50:41 2013
From: wampir990 at gmail.com (Jacek)
Date: Sun, 04 Aug 2013 14:50:41 +0200
Subject: [Freeswitch-users] Skype "Problem witch Audio Playback" - with
	skypopen-oss module.
Message-ID: <51FE4E21.8030808@gmail.com>

Hi

I have a problem with skypopem-oss kernel module.
After installation, the audio is normal, the next time, the system
starts, Skype has a "problem with audio playback", and do not make any
calls.

FreeSWITCH version:
User-Agent: 20130802T191223Z FreeSWITCH-mod_sofia/1.2.12 git ~ ~ 1e3bfef390

Log in failed calls via Skype:
http://pastebin.com/raw.php?i=eKWFGDUy

Kernel & Skypopen:
modinfo skypopen
filename:       /lib/modules/3.9.9-gr2/misc/skypopen.ko
license:        Dual BSD/GPL
author:         Original: Alessandro Rubini, Jonathan Corbet. Modified
by: Giovanni Maruzzelli for FreeSWITCH skypopen
srcversion:     8AA469D7BEB12A38BBDDA0C
depends:        
vermagic:       3.9.9-gr2 SMP preempt mod_unload modversions REFCOUNT
STACKLEAK_PLUGIN GRSEC
parm:           skypopen_major:int
parm:           skypopen_minor:int
parm:           skypopen_nr_devs:int


Is it possible to work mod_skypopen by Alsa sound, using mod_alsa or
mod_portaudio?

I have  in the office server that is 6 lines Skype, the old Freeswitch
1.0.4 everything worked, with the new is a problem.
Perhaps the module does not work very stable due to the presence of an
enhanced kernel Grsecurity & Pax, maybe the problem is the Gentoo
Hardened and hardened gcc-compiler with PIE & SPP -  gcc version 4.6.3
(Gentoo Hardened 4.6.3 p1.5, pie-0.5.2) ,
any case, I would prefer the whole shall use Alsa instead skypopen module.

An additional reason is that Microsoft blocked Skype 1.4, and at any
time can block the use of Skype 2.0, and this is the latest version of
the OSS.


For ALSA, even the latest version of Skype 4.1 works correctly.
6, or even 40 Skype installation can be done on separate user accounts
in separate folders to $HOME, each with a separate $HOME / .asoundrc,
and a separate session dbus.

That's how it worked for me on an earlier version of FreeSWITCH

By the way I would advise to use the Skype communication via dbus,
rather than X11, it is much more stable solution, and more predictable,
taking into account the Xorg will soon be replaced by Wayland and Mir,
and transportation Skype X11 may stop working.

Cheers


Sorry for my English, my native  language is Polish.
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From gmaruzz at gmail.com  Sun Aug  4 19:11:51 2013
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 4 Aug 2013 17:11:51 +0200
Subject: [Freeswitch-users] Skype "Problem witch Audio Playback" - with
 skypopen-oss module.
In-Reply-To: <51FE4E21.8030808@gmail.com>
References: <51FE4E21.8030808@gmail.com>
Message-ID: <CALXCt0pbzx7AAqgH_39X2yJfFdGDD7xQqj-RaUPPyi6xfKqk_Q@mail.gmail.com>

you probably have something that interferes with /dev/dsp
check that you have no modules *snd* loaded and that /dev/dsp is correctly
created by the script
no alsa no nothing
no problems with skype client whatsoever
folow strictly the wikipage for skypopen, use only supported OS, do not
install anything sound related (alsa, oss, portaudio, whatever) and you're
gold.
eg: follow the wikipage, Luke (or the source, if you're brave ;) )

On Sunday, August 4, 2013, Jacek <wampir990 at gmail.com> wrote:
> Hi
>
> I have a problem with skypopem-oss kernel module.
> After installation, the audio is normal, the next time, the system
starts, Skype has a "problem with audio playback", and do not make any
calls.
>
> FreeSWITCH version:
> User-Agent: 20130802T191223Z FreeSWITCH-mod_sofia/1.2.12 git ~ ~
1e3bfef390
>
> Log in failed calls via Skype:
> http://pastebin.com/raw.php?i=eKWFGDUy
>
> Kernel & Skypopen:
> modinfo skypopen
> filename:       /lib/modules/3.9.9-gr2/misc/skypopen.ko
> license:        Dual BSD/GPL
> author:         Original: Alessandro Rubini, Jonathan Corbet. Modified
by: Giovanni Maruzzelli for FreeSWITCH skypopen
> srcversion:     8AA469D7BEB12A38BBDDA0C
> depends:
> vermagic:       3.9.9-gr2 SMP preempt mod_unload modversions REFCOUNT
STACKLEAK_PLUGIN GRSEC
> parm:           skypopen_major:int
> parm:           skypopen_minor:int
> parm:           skypopen_nr_devs:int
>
>
> Is it possible to work mod_skypopen by Alsa sound, using mod_alsa or
mod_portaudio?
>
> I have  in the office server that is 6 lines Skype, the old Freeswitch
1.0.4 everything worked, with the new is a problem.
> Perhaps the module does not work very stable due to the presence of an
enhanced kernel Grsecurity & Pax, maybe the problem is the Gentoo Hardened
and hardened gcc-compiler with PIE & SPP -  gcc version 4.6.3 (Gentoo
Hardened 4.6.3 p1.5, pie-0.5.2) ,
> any case, I would prefer the whole shall use Alsa instead skypopen module.
>
> An additional reason is that Microsoft blocked Skype 1.4, and at any time
can block the use of Skype 2.0, and this is the latest version of the OSS.
>
>
> For ALSA, even the latest version of Skype 4.1 works correctly.
> 6, or even 40 Skype installation can be done on separate user accounts in
separate folders to $HOME, each with a separate $HOME / .asoundrc, and a
separate session dbus.
>
> That's how it worked for me on an earlier version of FreeSWITCH
>
> By the way I would advise to use the Skype communication via dbus, rather
than X11, it is much more stable solution, and more predictable, taking
into account the Xorg will soon be replaced by Wayland and Mir, and
transportation Skype X11 may stop working.
>
> Cheers
>
>
> Sorry for my English, my native  language is Polish.
>

-- 
Sent from Gmail Mobile
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From frankjr at mcpeekdodge.com  Sun Aug  4 19:42:10 2013
From: frankjr at mcpeekdodge.com (Frank Busalacchi Jr)
Date: Sun, 4 Aug 2013 15:42:10 +0000
Subject: [Freeswitch-users] Call Intercept
In-Reply-To: <7D5E4145-4B7E-4856-80A7-D93A31258AD4@gmail.com>
References: <C066384B0C1F1E4B9B870F050D7F1A9D44B103EC@ADAMA.mcpeekdodge.local>
	<7D5E4145-4B7E-4856-80A7-D93A31258AD4@gmail.com>
Message-ID: <C066384B0C1F1E4B9B870F050D7F1A9D44B10A33@ADAMA.mcpeekdodge.local>

Thanks for the replies.

First off, I got my scripting languages confused.  My intercept script is actually done in perl, not lua (See below).

Also,

                Loyd said: ?I use this way and it always works session:execute("set","intercept_unbridged_only=true")?

I?m trying to make my intercept script be able to intercept any call that is either ringing, or on hold.  I don?t want to be able to intercept a live call.  If I set ?intercept_unbridged_only? it seems to me that  I will be unable to pick up a ringing call?  Worth a try though?

Chris said:  That channel variable was fetched in the routing state before the lua application was executed.   You can either transfer to a new extension to execute the intercept or execute the intercept in the lua script.

If I am understanding you correctly, you are referring to the ?intercept-uuid? variable I am setting in the perl script.  I agree with you regarding the routing state vs. the execute state, but In the pastebin that I posted, line 259 shows that the perl script has made its choice of what UUID to ?intercept?, and in line 299, the XML dialplan actually calls intercept with the UUID the script suggested.  Also,  I originally was calling intercept in the perl script, but when I couldn?t get it to work, I switched to the combo method below in a desperate attempt for success.  Am I misinterpreting this somehow?

***** XML Dialplan Portion of intercept.... *****

<include>

    <extension name="intercept-ext">
      <condition field="destination_number" expression="^\*\*(\d{4})$">
        <action application="set" data="intercept_unbridged_only=false"/>
        <action application="set" data="intercept_unanswered_only=false"/>
        <action application="perl" data="${base_dir}/scripts/myintercept2.pl $1"/>
        <action application="answer"/>
        <action application="intercept" data="${intercept-uuid}"/>
        <action application="sleep" data="2000"/>
      </condition>
    </extension>

</include>

****  Perl Script to find UUID of extension to intercept *****

use strict;
use Switch;
use Data::Dumper;
use XML::Simple;
use POSIX qw(strftime);
our $session;

{
    my $debug = "true";

    my $dest_target = $ARGV[0];
    my $api = new freeswitch::API();

    my $uuid = $session->getVariable('uuid');

    my $detailed_calls = $api->executeString("show detailed_calls as xml");
    my $xml = new XML::Simple;
    my $chan_xml = $xml->XMLin($detailed_calls,KeyAttr => 'row', ForceArray => ['row']);

    my %candidates;
    my $reTarget = "sofia\/internal\/(sip:)*" . $dest_target . ".*";
    my $result;


    #--------------------------------------
    if ($debug eq "true") {
        freeswitch::consoleLog("INFO","\n----------intercept debug info-------------\n");
        freeswitch::consoleLog("INFO",$detailed_calls);
        freeswitch::consoleLog("INFO","\n----------end intercept debug info-------------\n");
    }

    #-- Loop through all the rows from show detailed_calls and put em in a hash...
    #--  Ringing lines are our first choice so make sure their hash key is "sortably" lower than a held call
    #--  Since we will sort the hash, and take the first sorted entry as the uuid to snag...

    foreach my $row (@{$chan_xml->{row}}) {

        if ($row->{callstate} eq "RINGING" && $row->{name} =~ $reTarget ) {
            $candidates{ $row->{created_epoch} - 1000 }  = $row->{uuid};

            if($debug eq "true") {
                freeswitch::consoleLog("INFO","Found intercept candidate.\nRINGING line for target:" . $dest_target .
                                       " uuid:" . $row->{uuid} . " call-uuid:" . $row->{call_uuid} . "\n");
            }
        }

        elsif ($row->{callstate} eq "HELD" && $row->{name} =~ $reTarget ) {
            $candidates{ $row->{b_created_epoch} } = $row->{b_uuid};

            if($debug eq "true") {
                freeswitch::consoleLog("INFO","Found intercept candidate.\nHELD line for target:" . $dest_target .
                                       " b_uuid:" . $row->{b_uuid} . " b_call-uuid:" . $row->{b_call_uuid} . "\n");
            }
        }

        elsif ($row->{b_callstate} eq "HELD" && $row->{b_name} =~ $reTarget ) {
            $candidates{ $row->{created_epoch} } = $row->{uuid};
            if($debug eq "true") {
                freeswitch::consoleLog("INFO","Found intercept candidate.\n  HELD line for target:" . $dest_target .
                                       " uuid:" . $row->{uuid} . " call-uuid:" . $row->{call_uuid} . "\n");
            }
        }
    }

    if( keys( %candidates ) == 0) {
        $result = "[Intercept] No suitable channel to intercept on extension $dest_target.";
        freeswitch::consoleLog("INFO","$result\n");
    }
    else {
        my $user = $session->getVariable('caller_id_name');
        my $cid_num = $session->getVariable('caller_id_number');
        my $winner_uuid;

        for my $winner ( sort keys %candidates ) {
            $winner_uuid = $candidates{$winner};
            last;
        }

        $session->setAutoHangup(0);
        freeswitch::consoleLog("INFO","INTERCEPT called by Ext:" .$cid_num. " INTERCEPTING Ext:" .$dest_target. " uuid:".$winner_uuid . "\n");
        $session->setVariable("intercept-uuid",$winner_uuid);
    }
    return 1;
}

-F

From: cmrienzo at gmail.com [mailto:cmrienzo at gmail.com]
Sent: Sunday, August 04, 2013 3:50 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call Intercept

That channel variable was fetched in the routing state before the lua application was executed.   You can either transfer to a new extension to execute the intercept or execute the intercept in the lua script.

Chris


On Aug 3, 2013, at 11:51 PM, Frank Busalacchi Jr <frankjr at mcpeekdodge.com<mailto:frankjr at mcpeekdodge.com>> wrote:
Hi everyone!  First and foremost, thanks for a great piece of software, and all the time you guys still put into this program day in and day out.  Thank you!

I'm having difficulty implementing an "intercept" system, and I'm thinking that I have to be "conceptually" misunderstanding something.  Towards those ends, I'm hoping someone can get me back on track, I?m sick of banging my head on the wall to try to make this work.

So here is my scenario:

1. A call (Call A) comes into the server from my voip provider on a DID.

2. My XML dialplan transfers the call to an extension in the XML dialplan of 1111.

3. My XML dialplan recognizes 1111 as a local extension, and executes a LUA script to handle the actual bridging of Call A to the appropriate devices.  I use the LUA script because I do some logic to decide whether to ring just the user's SIP phone, the user's cell phone, or a combination of both simultaneously.  Bottom line is that the LUA script does something like this after building the originate string in dialString:

session2 = freeswitch.Session(dialString);

   if session2:answered() == true then
      freeswitch.bridge(session,session2);
      return true;
   else
      return false;
   end

---All of the above works as expected ---

My issue is that when the LUA script starts ringing the sip phone registered as 1111, I want to 'intercept' that call from a different extension.  So here is what I do:

1.  Extension 1112 hears that his co-worker's extension 1111 is ringing, and wants to answer it.  He dials **1111

2.  My XML dialplan recognizes the request to pickup/intercept the call ringing at 1111, and calls a LUA script to figure out the uuid of the ringing phone.

3.  The LUA script sets a channel variable with the UUID it choose as the appropriate candidate to intercept, and exits.

4.  The XML dialplan resumes control, and does a <action application="intercept" data="${intercept-uuid}"/> .

5.  The calls aren't intercepted/bridged.

A pastebin of the log is here: http://pastebin.freeswitch.org/21267

What am I missing conceptually here?


Thanks,
Frank

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From lloyd.aloysius at sunteltech.ca  Sun Aug  4 20:00:30 2013
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Sun, 4 Aug 2013 12:00:30 -0400
Subject: [Freeswitch-users] Call Intercept
In-Reply-To: <C066384B0C1F1E4B9B870F050D7F1A9D44B10A33@ADAMA.mcpeekdodge.local>
References: <C066384B0C1F1E4B9B870F050D7F1A9D44B103EC@ADAMA.mcpeekdodge.local>
	<7D5E4145-4B7E-4856-80A7-D93A31258AD4@gmail.com>
	<C066384B0C1F1E4B9B870F050D7F1A9D44B10A33@ADAMA.mcpeekdodge.local>
Message-ID: <CAGvuWYaaRqw8oAvxJzEAavsMiTfrsyMXRrMrjkha=jtSR0GE2A@mail.gmail.com>

scripting language is not an issue.

Change the following

        <action application="set" data="intercept_unbridged_only=false"/>***
*

to

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


comment the following


        <action application="set" data="intercept_unanswered_only=false"/>


You can pickup a ringing phone.

*
*

* *


On Sun, Aug 4, 2013 at 11:42 AM, Frank Busalacchi Jr <
frankjr at mcpeekdodge.com> wrote:

>  Thanks for the replies.****
>
> ** **
>
> First off, I got my scripting languages confused.  My intercept script is
> actually done in perl, not lua (See below).****
>
> ** **
>
> Also,****
>
> ** **
>
>                 Loyd said: ?I use this way and it always
> works session:execute("set","intercept_unbridged_only=true")?****
>
> ** **
>
> I?m trying to make my intercept script be able to intercept any call that
> is either ringing, or on hold.  I don?t want to be able to intercept a live
> call.  If I set ?intercept_unbridged_only? it seems to me that  I will be
> unable to pick up a ringing call?  Worth a try though?****
>
> ** **
>
> Chris said:  That channel variable was fetched in the routing state
> before the lua application was executed.   You can either transfer to a new
> extension to execute the intercept or execute the intercept in the lua
> script. ****
>
> ** **
>
> If I am understanding you correctly, you are referring to the
> ?intercept-uuid? variable I am setting in the perl script.  I agree with
> you regarding the routing state vs. the execute state, but In the pastebin
> that I posted, line 259 shows that the perl script has made its choice of
> what UUID to ?intercept?, and in line 299, the XML dialplan actually calls
> intercept with the UUID the script suggested.  Also,  I originally was
> calling intercept in the perl script, but when I couldn?t get it to work, I
> switched to the combo method below in a desperate attempt for success.  Am
> I misinterpreting this somehow?****
>
> ** **
>
> ***** XML Dialplan Portion of intercept.... *********
>
> ** **
>
> <include>****
>
> ** **
>
>     <extension name="intercept-ext">****
>
>       <condition field="destination_number" expression="^\*\*(\d{4})$">***
> *
>
>         <action application="set" data="intercept_unbridged_only=false"/>*
> ***
>
>         <action application="set" data="intercept_unanswered_only=false"/>
> ****
>
>         <action application="perl" data="${base_dir}/scripts/
> myintercept2.pl $1"/>****
>
>         <action application="answer"/>****
>
>         <action application="intercept" data="${intercept-uuid}"/>****
>
>         <action application="sleep" data="2000"/>****
>
>       </condition>****
>
>     </extension>****
>
> ** **
>
> </include>****
>
> ** **
>
> ****  Perl Script to find UUID of extension to intercept *********
>
> ** **
>
> use strict;****
>
> use Switch;****
>
> use Data::Dumper;****
>
> use XML::Simple;****
>
> use POSIX qw(strftime);****
>
> our $session;****
>
> ** **
>
> {****
>
>     my $debug = "true";****
>
> ** **
>
>     my $dest_target = $ARGV[0];****
>
>     my $api = new freeswitch::API();****
>
> ** **
>
>     my $uuid = $session->getVariable('uuid');****
>
> ** **
>
>     my $detailed_calls = $api->executeString("show detailed_calls as xml");
> ****
>
>     my $xml = new XML::Simple;****
>
>     my $chan_xml = $xml->XMLin($detailed_calls,KeyAttr => 'row',
> ForceArray => ['row']);****
>
> ** **
>
>     my %candidates;****
>
>     my $reTarget = "sofia\/internal\/(sip:)*" . $dest_target . ".*";****
>
>     my $result;****
>
> ** **
>
> ** **
>
>     #--------------------------------------****
>
>     if ($debug eq "true") {****
>
>         freeswitch::consoleLog("INFO","\n----------intercept debug
> info-------------\n");****
>
>         freeswitch::consoleLog("INFO",$detailed_calls);****
>
>         freeswitch::consoleLog("INFO","\n----------end intercept debug
> info-------------\n");****
>
>     }****
>
> ** **
>
>     #-- Loop through all the rows from show detailed_calls and put em in a
> hash...****
>
>     #--  Ringing lines are our first choice so make sure their hash key is
> "sortably" lower than a held call****
>
>     #--  Since we will sort the hash, and take the first sorted entry as
> the uuid to snag...****
>
> ** **
>
>     foreach my $row (@{$chan_xml->{row}}) {****
>
> ** **
>
>         if ($row->{callstate} eq "RINGING" && $row->{name} =~ $reTarget ) {
> ****
>
>             $candidates{ $row->{created_epoch} - 1000 }  = $row->{uuid};**
> **
>
> ** **
>
>             if($debug eq "true") {****
>
>                 freeswitch::consoleLog("INFO","Found intercept
> candidate.\nRINGING line for target:" . $dest_target .****
>
>                                        " uuid:" . $row->{uuid} . "
> call-uuid:" . $row->{call_uuid} . "\n");****
>
>             }****
>
>         }****
>
> ** **
>
>         elsif ($row->{callstate} eq "HELD" && $row->{name} =~ $reTarget ) {
> ****
>
>             $candidates{ $row->{b_created_epoch} } = $row->{b_uuid};****
>
> ** **
>
>             if($debug eq "true") {****
>
>                 freeswitch::consoleLog("INFO","Found intercept
> candidate.\nHELD line for target:" . $dest_target .****
>
>                                        " b_uuid:" . $row->{b_uuid} . "
> b_call-uuid:" . $row->{b_call_uuid} . "\n");****
>
>             }****
>
>         }****
>
> ** **
>
>         elsif ($row->{b_callstate} eq "HELD" && $row->{b_name} =~
> $reTarget ) {****
>
>             $candidates{ $row->{created_epoch} } = $row->{uuid};****
>
>             if($debug eq "true") {****
>
>                 freeswitch::consoleLog("INFO","Found intercept
> candidate.\n  HELD line for target:" . $dest_target .****
>
>                                        " uuid:" . $row->{uuid} . "
> call-uuid:" . $row->{call_uuid} . "\n");****
>
>             }****
>
>         }****
>
>     }****
>
> ** **
>
>     if( keys( %candidates ) == 0) {****
>
>         $result = "[Intercept] No suitable channel to intercept on
> extension $dest_target.";****
>
>         freeswitch::consoleLog("INFO","$result\n");****
>
>     }****
>
>     else {****
>
>         my $user = $session->getVariable('caller_id_name');****
>
>         my $cid_num = $session->getVariable('caller_id_number');****
>
>         my $winner_uuid;****
>
> ** **
>
>         for my $winner ( sort keys %candidates ) {****
>
>             $winner_uuid = $candidates{$winner};****
>
>             last;****
>
>         }****
>
> ** **
>
>         $session->setAutoHangup(0);****
>
>         freeswitch::consoleLog("INFO","INTERCEPT called by Ext:"
> .$cid_num. " INTERCEPTING Ext:" .$dest_target. " uuid:".$winner_uuid .
> "\n");****
>
>         $session->setVariable("intercept-uuid",$winner_uuid);****
>
>     }****
>
>     return 1;****
>
> }****
>
> ** **
>
> -F****
>
> ** **
>
> *From:* cmrienzo at gmail.com [mailto:cmrienzo at gmail.com]
> *Sent:* Sunday, August 04, 2013 3:50 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Call Intercept****
>
> ** **
>
> That channel variable was fetched in the routing state before the lua
> application was executed.   You can either transfer to a new extension to
> execute the intercept or execute the intercept in the lua script. ****
>
> ** **
>
> Chris****
>
> ** **
>
>
> On Aug 3, 2013, at 11:51 PM, Frank Busalacchi Jr <frankjr at mcpeekdodge.com>
> wrote:****
>
>  Hi everyone!  First and foremost, thanks for a great piece of software,
> and all the time you guys still put into this program day in and day out.
> Thank you!****
>
>  ****
>
> I'm having difficulty implementing an "intercept" system, and I'm thinking
> that I have to be "conceptually" misunderstanding something.  Towards those
> ends, I'm hoping someone can get me back on track, I?m sick of banging my
> head on the wall to try to make this work.****
>
>  ****
>
> So here is my scenario:****
>
>  ****
>
> 1. A call (Call A) comes into the server from my voip provider on a DID.**
> **
>
>  ****
>
> 2. My XML dialplan transfers the call to an extension in the XML dialplan
> of 1111.****
>
>  ****
>
> 3. My XML dialplan recognizes 1111 as a local extension, and executes a
> LUA script to handle the actual bridging of Call A to the appropriate
> devices.  I use the LUA script because I do some logic to decide whether to
> ring just the user's SIP phone, the user's cell phone, or a combination of
> both simultaneously.  Bottom line is that the LUA script does something
> like this after building the originate string in dialString:****
>
>  ****
>
> session2 = freeswitch.Session(dialString);****
>
>  ****
>
>    if session2:answered() == true then****
>
>       freeswitch.bridge(session,session2);****
>
>       return true;****
>
>    else****
>
>       return false;****
>
>    end****
>
>  ****
>
> ---All of the above works as expected ---****
>
>  ****
>
> My issue is that when the LUA script starts ringing the sip phone
> registered as 1111, I want to 'intercept' that call from a different
> extension.  So here is what I do:****
>
>  ****
>
> 1.  Extension 1112 hears that his co-worker's extension 1111 is ringing,
> and wants to answer it.  He dials **1111****
>
>  ****
>
> 2.  My XML dialplan recognizes the request to pickup/intercept the call
> ringing at 1111, and calls a LUA script to figure out the uuid of the
> ringing phone.****
>
>  ****
>
> 3.  The LUA script sets a channel variable with the UUID it choose as the
> appropriate candidate to intercept, and exits.****
>
>  ****
>
> 4.  The XML dialplan resumes control, and does a <action
> application="intercept" data="${intercept-uuid}"/> .****
>
>  ****
>
> 5.  The calls aren't intercepted/bridged.****
>
>  ****
>
> A pastebin of the log is here: http://pastebin.freeswitch.org/21267 ****
>
>  ****
>
> What am I missing conceptually here?****
>
>  ****
>
>  ****
>
> Thanks,****
>
> Frank****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sravi123 at yahoo.com  Mon Aug  5 00:52:00 2013
From: sravi123 at yahoo.com (Ravi)
Date: Mon, 05 Aug 2013 02:22:00 +0530
Subject: [Freeswitch-users] Help with PRI Configuration - Bharti Airtel -
	Sangoma 101DE - Freeswitch
Message-ID: <51FEBEF0.6090500@yahoo.com>

Hello Everyone !

I am from India. I have recently taken a PRI connection from Bharti 
Airtel, one of the service providers. I have installed the following:

Cent OS
Freeswitch, FreeTDM
Sangoma Card


I think, I have followed all the instructions. I am struggling to 
configure Freeswitch to start using the PRI connection. This is what I 
have from the freeswitch cook book, to configure the gateway: we need 
username/password, server address or IP and port.

When I checked with Airtel, they are telling me that they only give 
username/password and IP address details for an internet connection and 
not for PRI lines.

Has anyone here in the list, tried using an indian service provider and 
configured in Freeswitch?
Has anyone done it for Airtel ?
Or please help me to figure out how to configure the PRI so as to make 
inbound and outbound calls ??

Any help is much appreciated.

Thanks.
Ravi
+91-7502029000



From krice at freeswitch.org  Mon Aug  5 01:38:03 2013
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 4 Aug 2013 16:38:03 -0500
Subject: [Freeswitch-users] Help with PRI Configuration - Bharti Airtel
	- Sangoma 101DE - Freeswitch
In-Reply-To: <51FEBEF0.6090500@yahoo.com>
References: <51FEBEF0.6090500@yahoo.com>
Message-ID: <B757A1DB-090B-479E-A18D-001A7B4C0F6D@freeswitch.org>

you should review the instructions for setting up PRI on the fs wiki... a prindoes not have a username or a password

Ken
Sent from my iPad

On Aug 4, 2013, at 15:52, Ravi <sravi123 at yahoo.com> wrote:

> Hello Everyone !
> 
> I am from India. I have recently taken a PRI connection from Bharti 
> Airtel, one of the service providers. I have installed the following:
> 
> Cent OS
> Freeswitch, FreeTDM
> Sangoma Card
> 
> 
> I think, I have followed all the instructions. I am struggling to 
> configure Freeswitch to start using the PRI connection. This is what I 
> have from the freeswitch cook book, to configure the gateway: we need 
> username/password, server address or IP and port.
> 
> When I checked with Airtel, they are telling me that they only give 
> username/password and IP address details for an internet connection and 
> not for PRI lines.
> 
> Has anyone here in the list, tried using an indian service provider and 
> configured in Freeswitch?
> Has anyone done it for Airtel ?
> Or please help me to figure out how to configure the PRI so as to make 
> inbound and outbound calls ??
> 
> Any help is much appreciated.
> 
> Thanks.
> Ravi
> +91-7502029000
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 wampir990 at gmail.com  Mon Aug  5 01:39:09 2013
From: wampir990 at gmail.com (Jacek)
Date: Sun, 04 Aug 2013 23:39:09 +0200
Subject: [Freeswitch-users] Skype "Problem witch Audio Playback" - with
 skypopen-oss module.
In-Reply-To: <CALXCt0pbzx7AAqgH_39X2yJfFdGDD7xQqj-RaUPPyi6xfKqk_Q@mail.gmail.com>
References: <51FE4E21.8030808@gmail.com>
	<CALXCt0pbzx7AAqgH_39X2yJfFdGDD7xQqj-RaUPPyi6xfKqk_Q@mail.gmail.com>
Message-ID: <51FEC9FD.8000409@gmail.com>

Sorry for the earlier post.

I'm used to that all kernel modules are
make me the device in / dev and set permissions.
Skypopen do not do this, so I added init command
mknod / dev / dsp c 15 3
and / dev / dsp was created, but because the system umask is 0022, so
the / dev / dsp was the owner root: root and permissions 644

When I manually changed the permissions and the owner / dev / dsp
now look like this:

ls-l / dev / dsp
crw-rw-rw- 1 root freeswitch 15, 3 08-04 23:10 /dev/dsp

  - Sype oss works fine.

I'll add myself to the startup scripts to automatically be set to the
owner and group permissions. ;)


  ;)

Cheers



W dniu 04.08.2013 17:11, Giovanni Maruzzelli pisze:
> you probably have something that interferes with /dev/dsp
> check that you have no modules *snd* loaded and that /dev/dsp is
> correctly created by the script
> no alsa no nothing
> no problems with skype client whatsoever
> folow strictly the wikipage for skypopen, use only supported OS, do
> not install anything sound related (alsa, oss, portaudio, whatever)
> and you're gold.
> eg: follow the wikipage, Luke (or the source, if you're brave ;) )
>
> On Sunday, August 4, 2013, Jacek <wampir990 at gmail.com
> <mailto:wampir990 at gmail.com>> wrote:
> > Hi
> >
> > I have a problem with skypopem-oss kernel module.
> > After installation, the audio is normal, the next time, the system
> starts, Skype has a "problem with audio playback", and do not make any
> calls.
> >
> > FreeSWITCH version:
> > User-Agent: 20130802T191223Z FreeSWITCH-mod_sofia/1.2.12 git ~ ~
> 1e3bfef390
> >
> > Log in failed calls via Skype:
> > http://pastebin.com/raw.php?i=eKWFGDUy
> >
> > Kernel & Skypopen:
> > modinfo skypopen
> > filename:       /lib/modules/3.9.9-gr2/misc/skypopen.ko
> > license:        Dual BSD/GPL
> > author:         Original: Alessandro Rubini, Jonathan Corbet.
> Modified by: Giovanni Maruzzelli for FreeSWITCH skypopen
> > srcversion:     8AA469D7BEB12A38BBDDA0C
> > depends:        
> > vermagic:       3.9.9-gr2 SMP preempt mod_unload modversions
> REFCOUNT STACKLEAK_PLUGIN GRSEC
> > parm:           skypopen_major:int
> > parm:           skypopen_minor:int
> > parm:           skypopen_nr_devs:int
> >
> >
> > Is it possible to work mod_skypopen by Alsa sound, using mod_alsa or
> mod_portaudio?
> >
> > I have  in the office server that is 6 lines Skype, the old
> Freeswitch 1.0.4 everything worked, with the new is a problem.
> > Perhaps the module does not work very stable due to the presence of
> an enhanced kernel Grsecurity & Pax, maybe the problem is the Gentoo
> Hardened and hardened gcc-compiler with PIE & SPP -  gcc version 4.6.3
> (Gentoo Hardened 4.6.3 p1.5, pie-0.5.2) ,
> > any case, I would prefer the whole shall use Alsa instead skypopen
> module.
> >
> > An additional reason is that Microsoft blocked Skype 1.4, and at any
> time can block the use of Skype 2.0, and this is the latest version of
> the OSS.
> >
> >
> > For ALSA, even the latest version of Skype 4.1 works correctly.
> > 6, or even 40 Skype installation can be done on separate user
> accounts in separate folders to $HOME, each with a separate $HOME /
> .asoundrc, and a separate session dbus.
> >
> > That's how it worked for me on an earlier version of FreeSWITCH
> >
> > By the way I would advise to use the Skype communication via dbus,
> rather than X11, it is much more stable solution, and more
> predictable, taking into account the Xorg will soon be replaced by
> Wayland and Mir, and transportation Skype X11 may stop working.
> >
> > Cheers
> >
> >
> > Sorry for my English, my native  language is Polish.
> >
>
> -- 
> Sent from Gmail Mobile
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 beffa at ieee.org  Mon Aug  5 02:51:22 2013
From: beffa at ieee.org (Federico Beffa)
Date: Mon, 5 Aug 2013 00:51:22 +0200
Subject: [Freeswitch-users] mod_dingaling <error type="modify">
Message-ID: <CAKrPhPN=h+uZX=8p5DL2FszsSZC7L7JEJ1as4oc+zpYFDTaSww@mail.gmail.com>

Hi,

I'm having some difficulties making freeswitch talk to gmail chat.
Signalling and text chat work both ways, but have troubles with audio. I'm
behind NAT and use STUN to traverse it.

In a problematic case I observe the following error in debug mode in the
console, where user1 initiated the conversation with user2


<iq to="user1 at XXX/fsvega0EC80FEA" id="313" type="error" from="user2 at YYY
/VtokiPhn75120521">
  <ses:session type="candidates" id="3759110558"
initiator="user1 at XXX/fsvega0EC80FEA"
xmlns:ses="http://www.google.com/session">
    <ses:candidate name="rtp" address="xxx.xxx.xxx.xxx" port="23751"
username="bSYbFfMybRj8D3xH" password="ri5AOommjwum60f8" preference="1.0"
protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
    <ses:candidate name="rtcp" address="xxx.xxx.xxx.xxx" port="23752"
username="tVI0bVcGkAl7kdQT" password="ay23gfVzUPdaKDW4" preference="1.0"
protocol="udp" type="stun" network="0" generation="0"></ses:candidate>
  </ses:session>
  <error type="modify">
    <sta:bad-request
xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"></sta:bad-request>
    <sta:text xml:lang="en"
xmlns:sta="urn:ietf:params:xml:ns:xmpp-stanzas"></sta:text>
  </error>
</iq>

It appears that user2 would like to modify the session, but mod_dingaling
does not take any action and I do not see any RTP traffic with wireshark.
Is this expected?

A second question is: In the session setup log I see mod_dingaling
advertising a port which is outside the range specified with the
configuration variables rtp-start-port and rtp-end-port in switch.xml. Do
the latter only work with SIP?

Thanks for any advise,
Fede
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From msc at freeswitch.org  Mon Aug  5 05:42:09 2013
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 4 Aug 2013 18:42:09 -0700
Subject: [Freeswitch-users] Help with PRI Configuration - Bharti Airtel
 - Sangoma 101DE - Freeswitch
In-Reply-To: <B757A1DB-090B-479E-A18D-001A7B4C0F6D@freeswitch.org>
References: <51FEBEF0.6090500@yahoo.com>
	<B757A1DB-090B-479E-A18D-001A7B4C0F6D@freeswitch.org>
Message-ID: <CAKzWOxUDrQ=bw5wPsVCwAkO=yZSdYHGjFq6=N=rVJMh8wAfvhw@mail.gmail.com>

Do they supply a "real" PRI or is it an IP connection with a device to
convert to PRI?
On Aug 4, 2013 4:43 PM, "Ken Rice" <krice at freeswitch.org> wrote:

> you should review the instructions for setting up PRI on the fs wiki... a
> prindoes not have a username or a password
>
> Ken
> Sent from my iPad
>
> On Aug 4, 2013, at 15:52, Ravi <sravi123 at yahoo.com> wrote:
>
> > Hello Everyone !
> >
> > I am from India. I have recently taken a PRI connection from Bharti
> > Airtel, one of the service providers. I have installed the following:
> >
> > Cent OS
> > Freeswitch, FreeTDM
> > Sangoma Card
> >
> >
> > I think, I have followed all the instructions. I am struggling to
> > configure Freeswitch to start using the PRI connection. This is what I
> > have from the freeswitch cook book, to configure the gateway: we need
> > username/password, server address or IP and port.
> >
> > When I checked with Airtel, they are telling me that they only give
> > username/password and IP address details for an internet connection and
> > not for PRI lines.
> >
> > Has anyone here in the list, tried using an indian service provider and
> > configured in Freeswitch?
> > Has anyone done it for Airtel ?
> > Or please help me to figure out how to configure the PRI so as to make
> > inbound and outbound calls ??
> >
> > Any help is much appreciated.
> >
> > Thanks.
> > Ravi
> > +91-7502029000
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Mon Aug  5 09:29:43 2013
From: eidevm5 at gmail.com (Peter)
Date: Mon, 5 Aug 2013 15:29:43 +1000
Subject: [Freeswitch-users] One way audio to CME
Message-ID: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>

I currently have successful two way calls (signalling and media) in the
following setup


External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  ->
Kamailio --> Internal Linphone

However, when I try to call a Cisco handset that is registered to CUCM9 via
CME in the following config:

External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> CME
->  CUCM9 -->  Cisco handset

The call signalling appears to be working fine and I can successfully
initiate a call from each end, but the only RTP stream that is working is
from the external Linphone client to the Cisco handset.

Note that CME is being used as a CUBE device, so all SIP and RTP goes via
it.

Looking at the RTP debugs on CME I can see the problem is that the "Media
Dest Addr" is getting set to the external side of the FS SBC rather than
the internal IP address.


I tried setting adding:

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

to the dialplan on the SBC, but it made no difference.


Any suggestions as to what to check next?

Thanks

Peter
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From eidevm5 at gmail.com  Mon Aug  5 11:15:51 2013
From: eidevm5 at gmail.com (Peter)
Date: Mon, 5 Aug 2013 17:15:51 +1000
Subject: [Freeswitch-users] TLS/Freeswitch self signed certs
Message-ID: <CABkS=xockpPROOhnbSaYJxt0uw2bUtKbyYus_CWqEsWz=sjB6Q@mail.gmail.com>

Has anyone managed to get TLS working between Android Linphone and
Freeswitch?

I've done the basic TLS setup as per https://wiki.freeswitch.org/wiki/Tls

I then convert the CA cert from PEM to DER format with:

openssl x509  -inform PEM -outform der -in cafile.pem -out fs.crt

I place fs.crt on a webserver and point my Android browser to it.

When I click on fs.crt, I get the default Android Certificate installer
popup, but it always says:

"Package contains: one user certificate"

ie: it thinks it is a user cert rather than a CA cert.

Android appears to be a real pain to add a CA to its trusted credential
store.

Really interested if anyone has managed to get Android to import the CA
cert.

Thanks

Peter
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From agtmcgarry at gmail.com  Mon Aug  5 11:23:08 2013
From: agtmcgarry at gmail.com (Anthony McGarry)
Date: Mon, 5 Aug 2013 08:23:08 +0100
Subject: [Freeswitch-users] One way audio to CME
In-Reply-To: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>
References: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>
Message-ID: <89239AD2-AE02-4E1C-8692-6E77894D8384@gmail.com>

On cube make sure you specify the source address on your dial-peers
voice-class sip bind media|control 
to the correct side. I have seen one way audio when not set. 

On 5 Aug 2013, at 06:29, Peter <eidevm5 at gmail.com> wrote:

> 
> 
> I currently have successful two way calls (signalling and media) in the following setup
> 
> 
> External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> Kamailio --> Internal Linphone
> 
> However, when I try to call a Cisco handset that is registered to CUCM9 via CME in the following config:
> 
> External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> CME ->  CUCM9 -->  Cisco handset
> 
> The call signalling appears to be working fine and I can successfully  initiate a call from each end, but the only RTP stream that is working is from the external Linphone client to the Cisco handset.   
> 
> Note that CME is being used as a CUBE device, so all SIP and RTP goes via it.
> 
> Looking at the RTP debugs on CME I can see the problem is that the "Media Dest Addr" is getting set to the external side of the FS SBC rather than the internal IP address.
> 
> 
> I tried setting adding:
> 
>             <action application="set" data="disable_rtp_auto_adjust="true" />
> 
> to the dialplan on the SBC, but it made no difference.
> 
> 
> Any suggestions as to what to check next?
> 
> Thanks
> 
> Peter
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 agtmcgarry at gmail.com  Mon Aug  5 11:23:08 2013
From: agtmcgarry at gmail.com (Anthony McGarry)
Date: Mon, 5 Aug 2013 08:23:08 +0100
Subject: [Freeswitch-users] One way audio to CME
In-Reply-To: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>
References: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>
Message-ID: <89239AD2-AE02-4E1C-8692-6E77894D8384@gmail.com>

On cube make sure you specify the source address on your dial-peers
voice-class sip bind media|control 
to the correct side. I have seen one way audio when not set. 

On 5 Aug 2013, at 06:29, Peter <eidevm5 at gmail.com> wrote:

> 
> 
> I currently have successful two way calls (signalling and media) in the following setup
> 
> 
> External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> Kamailio --> Internal Linphone
> 
> However, when I try to call a Cisco handset that is registered to CUCM9 via CME in the following config:
> 
> External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> CME ->  CUCM9 -->  Cisco handset
> 
> The call signalling appears to be working fine and I can successfully  initiate a call from each end, but the only RTP stream that is working is from the external Linphone client to the Cisco handset.   
> 
> Note that CME is being used as a CUBE device, so all SIP and RTP goes via it.
> 
> Looking at the RTP debugs on CME I can see the problem is that the "Media Dest Addr" is getting set to the external side of the FS SBC rather than the internal IP address.
> 
> 
> I tried setting adding:
> 
>             <action application="set" data="disable_rtp_auto_adjust="true" />
> 
> to the dialplan on the SBC, but it made no difference.
> 
> 
> Any suggestions as to what to check next?
> 
> Thanks
> 
> Peter
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mehroz.ashraf85 at gmail.com  Mon Aug  5 11:51:17 2013
From: mehroz.ashraf85 at gmail.com (Mehroz Ashraf)
Date: Mon, 5 Aug 2013 12:51:17 +0500
Subject: [Freeswitch-users] TLS/Freeswitch self signed certs
In-Reply-To: <CABkS=xockpPROOhnbSaYJxt0uw2bUtKbyYus_CWqEsWz=sjB6Q@mail.gmail.com>
References: <CABkS=xockpPROOhnbSaYJxt0uw2bUtKbyYus_CWqEsWz=sjB6Q@mail.gmail.com>
Message-ID: <CAC0j6rfkNbJ7KMKMo7VVfGRG0uox3xZjqaw8heX2=HE09TcrEg@mail.gmail.com>

Why do you want to place the cert on webserver and point android browser?
If you are doing this to download cert into android then that is probably
not the right approach.

I used cafile.pem (without converting it into .der format) and placed the
file in  SD card or phone memory, and point out linphone to get the CA from
the path. You may search in libraries where it need to tell the path.


On Mon, Aug 5, 2013 at 12:15 PM, Peter <eidevm5 at gmail.com> wrote:

> Has anyone managed to get TLS working between Android Linphone and
> Freeswitch?
>
> I've done the basic TLS setup as per https://wiki.freeswitch.org/wiki/Tls
>
> I then convert the CA cert from PEM to DER format with:
>
> openssl x509  -inform PEM -outform der -in cafile.pem -out fs.crt
>
> I place fs.crt on a webserver and point my Android browser to it.
>
> When I click on fs.crt, I get the default Android Certificate installer
> popup, but it always says:
>
> "Package contains: one user certificate"
>
> ie: it thinks it is a user cert rather than a CA cert.
>
> Android appears to be a real pain to add a CA to its trusted credential
> store.
>
> Really interested if anyone has managed to get Android to import the CA
> cert.
>
> Thanks
>
> Peter
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 dp.siddharth at eng.knowlarity.com  Mon Aug  5 12:24:19 2013
From: dp.siddharth at eng.knowlarity.com (DP Siddharth)
Date: Mon, 5 Aug 2013 13:54:19 +0530
Subject: [Freeswitch-users] Issue with export_on_answer with Sangoma
	A108/A108
In-Reply-To: <CAFDf6-9RVt0gf-rkNbVsTsRLSzh7TOJRF=2QLOXhRAMYYiUc+Q@mail.gmail.com>
References: <CAFDf6-9RVt0gf-rkNbVsTsRLSzh7TOJRF=2QLOXhRAMYYiUc+Q@mail.gmail.com>
Message-ID: <CAFDf6-_K165YPdbMSu+1wS4JX239jhCFtnK_975jYzVcvBqZAg@mail.gmail.com>

Hi All,

Moy, can you confirm if this is a known issue? Should I look for dedicated
time for sangoma support?

thanks & regards,
Siddharth


On Tue, Jul 23, 2013 at 2:01 PM, DP Siddharth <
dp.siddharth at eng.knowlarity.com> wrote:

> Hi All,
>
> I am trying to bridge a call & want to play a sound file when legb answers.
>
>
> here is dialplan:
>
> <extension name="incoming">
> <condition field="destination_number" expression="^4948667$">
> <action application="answer"/>
> <action application="bridge"
> data="{execute_on_answer=playback
> /srv/sounds/oriental-insurance-deepawali-demo.wav}freetdm/1/a/9873244228"/>
> <action application="hangup" />
> </condition>
> </extension>
>
> Here I am using Sangoma card & libsng_isdn library (I have tested with
> latest stable FreeSWITCH, wanpipe driver & libsng_isdn).
>
> FreeSWITCH: FreeSWITCH Version 1.0.head (git-9bf2726 2012-03-14 23-26-09
> -0500)
> I have updated with v1.2.stable but same observation
>
> WANPIPE Release: 3.5.28
> libsng_isdn.so.9.9.9, I have now updated it with libsng_isdn.so.7.27.2
>
> I am getting following error:
>
> 2013-07-22 12:44:53.725215 [WARNING] ftmod_wanpipe.c:1033 [s1c2][1:2]
> Failed to read 4096 bytes from sangoma device: No buffer space available
> (-65)
> 2013-07-22 12:44:53.725215 [WARNING] ftdm_io.c:4059 [s1c2][1:2] raw I/O
> read filed
> 2013-07-22 12:44:53.725215 [WARNING] mod_freetdm.c:776 Failed to read from
> channel FreeTDM/1:2/09873244228 device 1:2!
> 2013-07-22 12:44:53.805207 [WARNING] ftmod_wanpipe.c:1033 [s1c2][1:2]
> Failed to read 4096 bytes from sangoma device: No buffer space available
> (-65)
> 2013-07-22 12:44:53.805207 [WARNING] ftdm_io.c:4059 [s1c2][1:2] raw I/O
> read filed
>
>
> I also raised this issue with sangoma (
> http://support.sangoma.com/index.php?/Default/Tickets/Ticket/View/9246)
>
> but got no conclusive response in last 6 months.
>
> Kindly help in resolving same.
>
> thanks & regards,
> Siddharth
>
>
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From stuart.mills3 at btopenworld.com  Mon Aug  5 13:04:48 2013
From: stuart.mills3 at btopenworld.com (Stuart Mills)
Date: Mon, 5 Aug 2013 10:04:48 +0100
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC><000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com><1138329210A14BE4AEE53D7419427A15@PBPC><CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com><CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com><3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC><CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com><B782F8192B3E46608CE6666897F86BEB@PBPC><CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com><CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com>
	<CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com>
Message-ID: <F496A3CCE6994A408714AE2DCE31BAC6@PBPC>

LOL, I guess I should have visited pastebin before asking that question 

I?ve created an account and pasted 2 cdrs - 

http://pastebin.com/wsLnYxhz ? this is the inbound CDR for the call that originates 2 outbound calls using dial at once.

http://pastebin.com/YXkuqWsT ? this is the winner of the 2 outbound calls.

You will see in the inbound paste, the app_log stops at the point of the 2 calls, then it continues in the outbound winners cdr record.

If I just originate one call, this doesn?t happen, all of the app_log data is contained in the inbound cdr record.

Regards,

Stuart

From: Brian Foster 
Sent: Sunday, August 04, 2013 2:32 AM
To: FreeSWITCH Users Help 
Subject: Re: [Freeswitch-users] Call at once and XML-CDR

It's an "if you fail this simple test, part of your problem might be attention to detail" sort of test.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.

On Aug 3, 2013 9:30 PM, "Raymond Chandler" <intralanman at freeswitch.org> wrote:


  On Aug 3, 2013 9:19 PM, "Michael Collins" <msc at freeswitch.org> wrote:
  >
  >  "Speak friend and enter" 

  +1

  Love the LOTR reference :-)

  -Ray 


  _________________________________________________________________________
  Professional FreeSWITCH Consulting Services:
  consulting at freeswitch.org
  http://www.freeswitchsolutions.com

  
  

  Official FreeSWITCH Sites
  http://www.freeswitch.org
  http://wiki.freeswitch.org
  http://www.cluecon.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




--------------------------------------------------------------------------------
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 chang33.tw at gmail.com  Mon Aug  5 14:04:04 2013
From: chang33.tw at gmail.com (Jimmy Chang)
Date: Mon, 05 Aug 2013 18:04:04 +0800
Subject: [Freeswitch-users] H.264 OK SDP
Message-ID: <51FF7894.4040203@gmail.com>

Hi,

We have a softphone client using H.264 codec.
When we made a call to FS, we found that the OK response of FS for the 
INVITE missing something.
There is a SDP media attribute like this from softphone client:
/
Media Attribute (a): fmtp:116 profile-level-id=42000C;packetization-mode=0/

But in the OK SDP of FS, we could't find this info.
The softphone client would choose other resolution for this call.
We know that the H.264 is a pass through video codec in FS.
Our question is that is it possible and how can we add the attribute 
into the OK SDP?

FreeSWITCH Version 1.3.17.

Thanks.

Jimmy
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From sam.h.russell at gmail.com  Mon Aug  5 07:16:18 2013
From: sam.h.russell at gmail.com (Sam Russell)
Date: Mon, 5 Aug 2013 15:16:18 +1200
Subject: [Freeswitch-users] Audio PBX rejects video stream, kills call
Message-ID: <CAJm+voVBFEyUNHg0jeZhKiatE9LcCF-Pj0WKzOMGR+DZnqFf7A@mail.gmail.com>

Hi guys,

I've been using FreeSWITCH for a couple of weeks and built a PBX and B2BUA
out of it. I've seen some interesting things that happen when PBXs get m=
headers that they don't like (such as Asterisk balking when it gets
multiple video streams), and I've seen one recently while calling an old
school PBX.

Here's my build:

Me(Linphone) -> PBX (FreeSWITCH) -> B2BUA (FreeSWITCH) -> Destination

The B2BUA uses proxy_media mode and doesn't have a problem, but the PBX
uses passthrough codecs (so it's reading the SDP packet) and sees a
problem, throwing a 500 error back to my softphone

The outbound packet from the PBX to the B2BUA:

send 1428 bytes to udp/[210.7.45.66]:5060 at 02:46:48.407768:
   ------------------------------------------------------------------------
   INVITE sip:+61269337555 at 210.7.45.66 SIP/2.0
   Via: SIP/2.0/UDP 210.7.46.240;rport;branch=z9hG4bKZrKQvvHNSy64m
   Max-Forwards: 69
   From: "Sam Russell" <sip:+6442820079 at 210.7.46.240>;tag=e722ggarBe7je
   To: <sip:+61269337555 at 210.7.45.66>
   Call-ID: 1d80cd44-781c-1231-8db7-0050568a49fc
   CSeq: 47499340 INVITE
   Contact: <sip:gw+sbc at 210.7.46.240:5060;transport=udp;gw=sbc>
   User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130223T185558Z~28680c5e58
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 453
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "Sam Russell" <sip:+6442820079 at 210.7.46.240
>;party=calling;screen=yes;privacy=off

   v=0
   o=FreeSWITCH 1375638308 1375638309 IN IP4 210.7.46.240
   s=FreeSWITCH
   c=IN IP4 210.7.46.240
   t=0 0
   m=audio 32500 RTP/AVP 98 99 100 0 8 101 13
   a=rtpmap:98 SPEEX/32000
   a=fmtp:98 vbr=on
   a=rtpmap:99 SPEEX/16000
   a=fmtp:99 vbr=on
   a=rtpmap:100 SPEEX/8000
   a=fmtp:100 vbr=on
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-16
   a=ptime:20
   m=video 29910 RTP/AVP 102 103
   a=rtpmap:102 H263-1998/90000
   a=fmtp:102 CIF=1;QCIF=1
   a=rtpmap:103 VP8/90000

The offending packet that it returns:

   ------------------------------------------------------------------------
recv 1357 bytes from udp/[210.7.45.66]:5060 at 02:46:48.739816:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 210.7.46.240;rport=5060;branch=z9hG4bKZrKQvvHNSy64m
   From: "Sam Russell" <sip:+6442820079 at 210.7.46.240>;tag=e722ggarBe7je
   To: <sip:+61269337555 at 210.7.45.66>;tag=teBFaBg5DjBHm
   Call-ID: 1d80cd44-781c-1231-8db7-0050568a49fc
   CSeq: 47499340 INVITE
   Contact: <sip:+61269337555 at 210.7.45.66:5060;transport=udp>
   User-Agent: FreeSWITCH-mod_sofia/1.2.11+git~20130723T222215Z~55b6b8424f
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, conference, presence, dialog, line-seize,
call-info, sla, include-session-description, presence.winfo,
message-summary, refer
   Content-Type: application/sdp
   Content-Disposition: session
   Content-Length: 242
   X-FS-Display-Name: Outbound Call
   X-FS-Display-Number: sip:+61269337555 at 210.7.45.66
   x-inin-crn: 1101488375;loc=%3cRegionDefaultLocation%3e
   X-FS-Support: update_display,send_info
   Remote-Party-ID: "Outbound Call" <sip:+61269337555 at 210.7.45.66
>;party=calling;privacy=off;screen=no

   v=0
   o=FreeSWITCH 914114385 914114386 IN IP4 210.7.45.66
   s=FreeSWITCH
   c=IN IP4 210.7.45.66
   t=0 0
   m=audio 17704 RTP/AVP 8 101
   a=rtpmap:8 PCMA/8000
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   m=video 0 RTP/AVP 96
   a=rtpmap:96 /0


The PBX then balks and sends a 500 error to my softphone

send 704 bytes to udp/[210.7.45.66]:5060 at 02:46:48.740796:
   ------------------------------------------------------------------------
   BYE sip:+61269337555 at 210.7.45.66:5060;transport=udp SIP/2.0
   Via: SIP/2.0/UDP 210.7.46.240;rport;branch=z9hG4bK1a68ZjKvKgKac
   Max-Forwards: 70
   From: "Sam Russell" <sip:+6442820079 at 210.7.46.240>;tag=e722ggarBe7je
   To: <sip:+61269337555 at 210.7.45.66>;tag=teBFaBg5DjBHm
   Call-ID: 1d80cd44-781c-1231-8db7-0050568a49fc
   CSeq: 47499341 BYE
   Contact: <sip:gw+sbc at 210.7.46.240:5060;transport=udp;gw=sbc>
   User-Agent: FreeSWITCH-mod_sofia/1.3.13b+git~20130223T185558Z~28680c5e58
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Reason: SIP;cause=488;text="Incomplete offer/answer"
   Content-Length: 0

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


Then I see this in the logs straight after

2013-08-05 14:46:48.736200 [DEBUG] sofia.c:1431 nua_i_media_error: unknown
event 22: 988 Incomplete offer/answer




The only thing odd about the SDP packet that I get back is the attribute
after the m=video line. I think FreeSWITCH sees this, doesn't ignore the a=
line like it should, and then gets upset with not being able to match
anything before the "/0". RFC 4566 (5.13) says that

"If an attribute is received that is not understood, it MUST be ignored by
the receiver."

So FreeSWITCH should be ignoring any "a=" attributes after it gets a "m="
line with a 0 port, but should also ignore the attribute if it can't match
(but maybe leave a warning/notice in logs)?

Does this look right? Has anyone else seen this, and is there a workaround?

Cheers
Sam
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From khorsmann at gmail.com  Mon Aug  5 14:56:26 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Mon, 5 Aug 2013 12:56:26 +0200
Subject: [Freeswitch-users] Can't call from Firefox 22 to Freeswitch
	using sipml5
In-Reply-To: <CAMKwYLASs++ZxCNiAQ1yJMB+oROwwPAr8MuZ65g4dyQ-6nY_fg@mail.gmail.com>
References: <CAMKwYLASs++ZxCNiAQ1yJMB+oROwwPAr8MuZ65g4dyQ-6nY_fg@mail.gmail.com>
Message-ID: <CAFArqsbMZXD4pK2W65sNOQpgn7TdCR9wec8YFwM6NLYFyftdQw@mail.gmail.com>

Hi all,


i have still the same problem, with 1.4beta branch and master branch.
Firefox 22 still no audio.
Did you get any solution with your Problem?




2013/7/26 Gustavo Salazar <guga.salazar.loor at gmail.com>

>
> Hi,
>
> I want to make calls using sipml5 and freeswitch. I have installed
> freeswitch from the git repository in an ec2 instance with elastic ip
> address, configured sip_profiles/internal.xml with the EIP and here the
> results:
>
> Using Google Chrome and the sipml5 client I register as user 1001 with
> password 1234 , I dial 5000 and I hear the IVR voice, also I can join to a
> conference I created, and hear myself in an echo test dialing 9196. No
> problem with that.
>
> Using Firefox 22 and the same sipml5 client I register as user 1002 and
> password 1234,  then dial 5000, I see the audio files are played in the
> freeswitch cli but I can not hear anything.
>
> I guess the problem is not in my sipml5 client neither Firefox because
> when I use them to register to  webrtc.freeswitch.org there is not any
> problem, I hear the voice after dialing 5000.
>
> I think I am missing some configuration in my Freeswitch server, where I
> also loaded the opus module and set OPUS in the file vars.xml , in the
> parameters global_codec_prefs and outbound_codec_prefs.
>
> I attach the console output from Freeswitch and the console output from
> Firefox
>
> Here the freeswitch output
> http://pastebin.com/S0yJxT1J
>
> Here the Firefox console output
> http://pastebin.com/8kBGdY3J
>
> Thanks in advance for the help
>
>
>
> --
> Gustavo Salazar
>
>
>


-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From jlotery at gmail.com  Mon Aug  5 16:04:57 2013
From: jlotery at gmail.com (James Lotery)
Date: Mon, 5 Aug 2013 14:04:57 +0200
Subject: [Freeswitch-users] message/external-body as content type in SIMPLE
	message
Message-ID: <CAAXEqXB-H=0dThMygvY5ue=UKwmhuMrLkvi7V=YSXiUA4HV+1g@mail.gmail.com>

Hello, I'm using mod_sms in freeswitch for sending messages between
registered users. One thing I'm trying to get working is sharing of images
using SIP SIMPLE by setting the content-type to message/external-body (see
below). However it looks like any parameters after message/external-body
are being filtered out between hitting FS and reaching mod_sms.

If I turn on sip trace in sofia here's what I see:

  ------------------------------------------------------------------------
recv 517 bytes from udp/[xxxxx]:56692 at 11:56:34.486792:
   ------------------------------------------------------------------------
   MESSAGE sip:33979085 at xxxxx SIP/2.0
   Via: SIP/2.0/UDP xxx:56692;rport;branch=z9hG4bK1213589041
   From: <sip:1009 at xxxxx>;tag=133921508
   To: <sip:33979085 at xxxxx>
   Call-ID: 908346813
   CSeq: 20 MESSAGE
   Content-Type: message/external-body; access-type=URL; URL="
https://xxxxxxx/51ff92f903de3_6b94294d169daab4785f.jpg"
   Max-Forwards: 70
   Date: Mon Aug  5 13:56:01 2013
   Content-Length: 0


In this example all I do in my chatplan is execute a lua script that in
turn executes the send function of mod_sms without changing anything. Here
is what is sent from FS :

  ------------------------------------------------------------------------
send 1281 bytes to udp/[xxxx]:1024 at 11:56:34.489567:
   ------------------------------------------------------------------------
   MESSAGE sip:33979085 at xxxxx:1024 SIP/2.0
   Via: SIP/2.0/UDP xxxxx;rport;branch=z9hG4bKN90r1v1yvyt2j
   Route: <sip:33979085 at xxxxx
:1024>;app-id=622464153529;pn-type=google;pn-tok=APA91bH9NTVgzIyoSzaN7c4sf69gYMPd2acTQlLKEWt20_sX3acdti9EXBtpXfsHfi6RMwlq-zERhxwKmhu6pepI6rOCRZ6p5ZFC15Y-sgGyJ6HYRU93fi1SebSrsO8h679IOXSrxDTq;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-call-snd=ring.caf;pn-msg-snd=msg.caf;line=153ea365cc40ca1
   Max-Forwards: 70
   From: <sip:1009 at vxxxxx>;tag=133921508
   To: <sip:33979085 at xxxxx
:1024>;app-id=622464153529;pn-type=google;pn-tok=APA91bH9NTVgzIyoSzaN7c4sf69gYMPd2acTQlLKEWt20_sX3acdti9EXBtpXfsHfi6RMwlq-zERhxwKmhu6pepI6rOCRZ6p5ZFC15Y-sgGyJ6HYRU93fi1SebSrsO8h679IOXSrxDTq;pn-msg-str=IM_MSG;pn-call-str=IC_MSG;pn-call-snd=ring.caf;pn-msg-snd=msg.caf;line=153ea365cc40ca1
   Call-ID: 55302023-3114-432b-8b90-30a5d7e66706
   CSeq: 47515157 MESSAGE
   Contact: <sip:xxxxx>
   User-Agent: FreeSWITCH-mod_sofia/1.5.5b+git~20130729T215516Z~511efc5cf0
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
   Supported: timer, precondition, path, replaces
   Content-Type: message/external-body
   Content-Length: 0
   X-FS-Sending-Message: 9dd08617-6b31-47dd-8f6f-ff46abf521b6


As you can see everything after message/extern-body has been removed in the
content type. What do I have to do so that this comes through to mod_sms
intact?

At which point passing through FS would this get filtered out? It is in the
SIP trace but not there when it gets to mod_sms it seems.

Any ideas ?

Thanks in adavnce

- J
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From shahrzad.aziminia at gmail.com  Mon Aug  5 16:24:20 2013
From: shahrzad.aziminia at gmail.com (Shahrzad A.)
Date: Mon, 5 Aug 2013 14:24:20 +0200
Subject: [Freeswitch-users] Issue with JSSIP + Freeswitch
Message-ID: <CAA7Zpyo5rGpo5jdXjK35-kuH52wJ_10k9KFn=OudDpHvBNqbvw@mail.gmail.com>

Hi  everyone

I'm having latest version of Freeswitch installed on Ubuntu 12.04.2 LTS
 with the latest version of Openssl ('OpenSSL 1.0.1e 11 Feb 2013')
I'm using the default configuration and just uncommentated the ' <param
name="ws-binding"  value=":5066"/> ' in internal.xml in order to have the
support for webrtc.
As the client I'm having JSSIP, the latest version with the adjustment to
have (DtlsSrtpKeyAgreement:true). Both my clients and server located in the
same network without having any firewalls therefore I'm not using any STUN
server. I can make call but there is no sound! and after setting
debig_level 10 and dial from extension 1003 to 5000 I'm having the
following Alert in console and no sound:

[ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not
ready!)

its the same case if I dial from extension 1002 to 1005! (Both are JSSIP
clients)
If I call from extension 1000 which is a SIP client set on my iPhone and
call 1003 I'm having the following alerts but voice on both JSSIP client
and SIP client:

2013-08-05 14:17:05.222446 [ALERT] switch_rtp.c:4563 sofia/internal/
1000 at 10.0.14.16:5060 timer while HOT
2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/
1000 at 10.0.14.16:5060 Hot Hit 1

And from extension 1003 (JSSIP) to extension 1000 ( SIP on Iphone) No voice
and I'm getting the following alerts:

2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4546 sofia/internal/
sip:1000 at 10.0.14.182:5065 Hot Hit 4
2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/
sip:1000 at 10.0.14.182:5065 timer while HOT
2013-08-05 14:19:57.132441 [ALERT] switch_rtp.c:5672 Skip sending audio
packet 172 bytes (dtls not ready!)

If I use bypass media or proxy media I will have voice on both JSSIP
clients but cant ring any SIP clients and again no voice if I call 5000!

Wonder to know if there is any special setting required on FreeSwitch or
its and issue from JSSIP?

Thanks in advanced!

Sherry
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From wampir990 at gmail.com  Mon Aug  5 17:03:01 2013
From: wampir990 at gmail.com (Jacek)
Date: Mon, 05 Aug 2013 15:03:01 +0200
Subject: [Freeswitch-users] Skype "Problem witch Audio Playback" - with
	skypopen-oss module.
Message-ID: <51FFA285.2050608@gmail.com>

Hi

How should it look like fs_cli command that automatically connects the
call using the following method:
A call to the user,
when he receives a call (pick up the phone), hears the message about the
implementation of the merger, and then will be automatically connected
with the number of user B.

I mean the shape of the script command.

Is it possible to do this with *fs_cli**  -**x*, or do I have to combine
the script, which will implement a similar action.

Cheers


Sorry for my English, my native  language is Polish.
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From wampir990 at gmail.com  Mon Aug  5 17:05:16 2013
From: wampir990 at gmail.com (Jacek)
Date: Mon, 05 Aug 2013 15:05:16 +0200
Subject: [Freeswitch-users] How to set up a connection between two users by
	using fs_cli?
Message-ID: <51FFA30C.1070707@gmail.com>

Hi

How should it look like fs_cli command that automatically connects the
call using the following method:
A call to the user,
when he receives a call (pick up the phone), hears the message about the
implementation of the merger, and then will be automatically connected
with the number of user B.

I mean the shape of the script command.

Is it possible to do this with *fs_cli**  -**x*, or do I have to combine
the script, which will implement a similar action.

Cheers


Sorry for my English, my native  language is Polish.
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From fs.user at fordior.net  Mon Aug  5 17:38:30 2013
From: fs.user at fordior.net (EL)
Date: Mon, 5 Aug 2013 15:38:30 +0200
Subject: [Freeswitch-users] FreeSWITCH -- 30 Second call drop
In-Reply-To: <20130801211304.35C9957E002@mail.mydcs.ca>
References: <20130801204721.GC20191@0rdior.com>
	<20130801211304.35C9957E002@mail.mydcs.ca>
Message-ID: <20130805133830.GA32253@0rdior.com>

Hello Paul,

I tried your checklist, but that wasn't the solution in my setup.

For everyone who's running FS on a public server (WAN ip) in
combination with a VPN:

----
sip_profiles/internal.xml:

<param name="rtp-ip" value="your-internal-VPN-ip"/>
<param name="sip-ip" value="your-internal-VPN-ip"/>
<param name="ext-sip-ip" value="your-internal-VPN-ip"/>
<param name="ext-rtp-ip" value="your-internal-VPN-ip"/>
----

My mistake was giving both ext-sip-ip and ext-rtp-ip my WAN IP
address.  After several dumpcaps and analysing these with wireshark, I
noticed a change in the 'Via:' address of the SIP Message Header.
Instead of the internal VPN ip address, it changed to my WAN address.
So the ACK wasn't answered and therefor the call was disconnected
after 32 seconds.

(Please note that all these values in the sip_profiles/external.xml
need to have the WAN ip address.)

Maybe this information will help someone else some day...

@jackal:
Thanks for pointing out some usefull informations regarding ICMP and MTU.

Cheers!
-- 
EL



From krice at freeswitch.org  Mon Aug  5 18:26:01 2013
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 5 Aug 2013 09:26:01 -0500
Subject: [Freeswitch-users] How to set up a connection between two users
	by using fs_cli?
In-Reply-To: <51FFA30C.1070707@gmail.com>
References: <51FFA30C.1070707@gmail.com>
Message-ID: <FCAFD075-A636-4DA0-8AEB-9CA4FE258A6E@freeswitch.org>

See the originate command.  You can do this from fs_cli or via ESL from a scripting language like perl or ruby etc

Sent from my iPhone

On Aug 5, 2013, at 8:05 AM, Jacek <wampir990 at gmail.com> wrote:

> Hi
> 
> How should it look like fs_cli command that automatically connects          the call using the following method:
> A call to the user,
> when he receives a call (pick up the phone), hears the message about the       implementation of the merger, and then will be automatically connected with the number of user B.
> 
> I mean the shape of the script command.
> 
> Is it possible to do this with fs_cli  -x, or do I have to combine the script, which will implement a similar action.
> 
> Cheers
> 
> 
> Sorry for my English, my native  language is Polish.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 gmail.com  Mon Aug  5 18:46:25 2013
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 5 Aug 2013 16:46:25 +0200
Subject: [Freeswitch-users] Skype "Problem witch Audio Playback" - with
 skypopen-oss module.
In-Reply-To: <51FFA285.2050608@gmail.com>
References: <51FFA285.2050608@gmail.com>
Message-ID: <CALXCt0qjBHmScVgZCwCwRYHHNFZWwAy3-+wumTnge+KD4cterA@mail.gmail.com>

please open a new mail thread, so others can read and give their opinions.

I would use one of the many scripting options, or the dialplan.

-giovanni

On Monday, August 5, 2013, Jacek <wampir990 at gmail.com> wrote:
> Hi
>
> How should it look like fs_cli command that automatically connects the
call using the following method:
> A call to the user,
> when he receives a call (pick up the phone), hears the message about the
implementation of the merger, and then will be automatically connected with
the number of user B.
>
> I mean the shape of the script command.
>
> Is it possible to do this with fs_cli  -x, or do I have to combine the
script, which will implement a similar action.
>
> Cheers
>
>
> Sorry for my English, my native  language is Polish.
>

-- 
Sent from Gmail Mobile
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From itsme.kunnu at gmail.com  Mon Aug  5 18:57:31 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Mon, 5 Aug 2013 20:27:31 +0530
Subject: [Freeswitch-users] Softphone
In-Reply-To: <CAMvhjJsP+vY1q3s0bDHdOfiDh37Wwk1g1SSsvi-8fJVNcj4YUw@mail.gmail.com>
References: <CAMvhjJsP+vY1q3s0bDHdOfiDh37Wwk1g1SSsvi-8fJVNcj4YUw@mail.gmail.com>
Message-ID: <CAMvhjJs6zF6Th__8y4qBKDJ2vQrxnvNdibArbsa9mDZbzF1jJg@mail.gmail.com>

Sir i have installed xlite soft phone on my windows 8 pc and i am trying to
connect to the freeswitch server running on my ubuntu machine but when i am
trying to enable the account it displays an error that "account not enabled
contact ur administrator" also i receive the same errror when i connect my
windows 8 pc running xlite softphone with a windows 7 machine acting as
server through a local LAN.

Thank you
Ashish Mishra
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From guga.salazar.loor at gmail.com  Mon Aug  5 19:16:06 2013
From: guga.salazar.loor at gmail.com (Gustavo Salazar)
Date: Mon, 5 Aug 2013 10:16:06 -0500
Subject: [Freeswitch-users] Can't call from Firefox 22 to Freeswitch
	using sipml5
In-Reply-To: <CAFArqsbMZXD4pK2W65sNOQpgn7TdCR9wec8YFwM6NLYFyftdQw@mail.gmail.com>
References: <CAMKwYLASs++ZxCNiAQ1yJMB+oROwwPAr8MuZ65g4dyQ-6nY_fg@mail.gmail.com>
	<CAFArqsbMZXD4pK2W65sNOQpgn7TdCR9wec8YFwM6NLYFyftdQw@mail.gmail.com>
Message-ID: <CAMKwYLAdOvXUz0XLcJX653MwDZbYaoGO=q7kH8QYm7xx0urGBA@mail.gmail.com>

Hi Karsten,

I haven't solved the problem, it works only with Google Chrome.  I  tested
the master branch and the version 1.4 with the same results. ..




2013/8/5 Karsten Horsmann <khorsmann at gmail.com>

> Hi all,
>
>
> i have still the same problem, with 1.4beta branch and master branch.
> Firefox 22 still no audio.
> Did you get any solution with your Problem?
>
>
>
>
> 2013/7/26 Gustavo Salazar <guga.salazar.loor at gmail.com>
>
>>
>> Hi,
>>
>> I want to make calls using sipml5 and freeswitch. I have installed
>> freeswitch from the git repository in an ec2 instance with elastic ip
>> address, configured sip_profiles/internal.xml with the EIP and here the
>> results:
>>
>> Using Google Chrome and the sipml5 client I register as user 1001 with
>> password 1234 , I dial 5000 and I hear the IVR voice, also I can join to a
>> conference I created, and hear myself in an echo test dialing 9196. No
>> problem with that.
>>
>> Using Firefox 22 and the same sipml5 client I register as user 1002 and
>> password 1234,  then dial 5000, I see the audio files are played in the
>> freeswitch cli but I can not hear anything.
>>
>> I guess the problem is not in my sipml5 client neither Firefox because
>> when I use them to register to  webrtc.freeswitch.org there is not any
>> problem, I hear the voice after dialing 5000.
>>
>> I think I am missing some configuration in my Freeswitch server, where I
>> also loaded the opus module and set OPUS in the file vars.xml , in the
>> parameters global_codec_prefs and outbound_codec_prefs.
>>
>> I attach the console output from Freeswitch and the console output from
>> Firefox
>>
>> Here the freeswitch output
>> http://pastebin.com/S0yJxT1J
>>
>> Here the Firefox console output
>> http://pastebin.com/8kBGdY3J
>>
>> Thanks in advance for the help
>>
>>
>>
>> --
>> Gustavo Salazar
>>
>>
>>
>
>
> --
> Mit freundlichen Gr??en
> *Karsten Horsmann*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Gustavo Salazar
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From khorsmann at gmail.com  Mon Aug  5 20:59:07 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Mon, 5 Aug 2013 18:59:07 +0200
Subject: [Freeswitch-users] Can't call from Firefox 22 to Freeswitch
	using sipml5
In-Reply-To: <CAMKwYLAdOvXUz0XLcJX653MwDZbYaoGO=q7kH8QYm7xx0urGBA@mail.gmail.com>
References: <CAMKwYLASs++ZxCNiAQ1yJMB+oROwwPAr8MuZ65g4dyQ-6nY_fg@mail.gmail.com>
	<CAFArqsbMZXD4pK2W65sNOQpgn7TdCR9wec8YFwM6NLYFyftdQw@mail.gmail.com>
	<CAMKwYLAdOvXUz0XLcJX653MwDZbYaoGO=q7kH8QYm7xx0urGBA@mail.gmail.com>
Message-ID: <CAFArqsb5XcHnUM4_ju-39BVxDuc59bCD9vWLhA+QSsoU-BKEDg@mail.gmail.com>

Hi Gustavo,


i think there am more than one Problem with WebRTC + sipml5 + Firefox 22
(or higher).

First, i saw an dtls error and many "use 1.0.1e openssl" hints.

You can check this with openssl client. Here a good and a broken example:

openssl s_client -dtls1 -connect webrtc.freeswitch.org:7443
CONNECTED(00000003)


openssl s_client -dtls1 -connect some.broken.host:7443
CONNECTED(00000003)
write:errno=111
no peer certificate available
No client certificate CA names sent
SSL handshake has read 0 bytes and written 0 bytes
New, (NONE), Cipher is (NONE)
Secure Renegotiation IS NOT supported
Compression: NONE
Expansion: NONE
SSL-Session:
    Protocol  : DTLSv1
    Cipher    : 0000
    Session-ID:
    Session-ID-ctx:
    Master-Key:
    Key-Arg   : None
    Krb5 Principal: None
    PSK identity: None
    PSK identity hint: None
    Start Time: 1375716502
    Timeout   : 7200 (sec)
    Verify return code: 0 (ok)


Then you can check https://webrtc.freeswitch.org/sipml5/ with Firefox 22.
It works like a charm.

I thought then, maybe some voodoo was made on the sipml5 code. But there
was a few settings insert. And i test exact the same sipml5 code with my
test WebRTC FreeSWITCHes, it doesnt work with Firefox.

Only Chrome works.

In Firefox Webdeveloper Console i got only an SDP Warning of the sipml5
Client:

({name:"INVALID_SESSION_DESCRIPTION", message:"Could not negotiate answer
SDP; cause = ERR | SDP Parsing Error:  Warning: Unrecognized attribute
(msid-semantic)  | SDP Parsing Error:  Warning: Unrecognized attribute
(ssrc)  | SDP Parsing Error:  Warning: Unrecognized attribute (ssrc)  | SDP
Parsing Error:  Warning: Unrecognized attribute (ssrc)  | SDP Parsing
Error:  Warning: Unrecognized attribute (ssrc) ",
__exposedProps__:{name:"rw", message:"rw"}})

Maybe the developers give us another hint?

Cheers...


2013/8/5 Gustavo Salazar <guga.salazar.loor at gmail.com>

> Hi Karsten,
>
> I haven't solved the problem, it works only with Google Chrome.  I  tested
> the master branch and the version 1.4 with the same results. ..
>
>


-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From anthony.minessale at gmail.com  Mon Aug  5 21:16:18 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 5 Aug 2013 12:16:18 -0500
Subject: [Freeswitch-users] Can't call from Firefox 22 to Freeswitch
	using sipml5
In-Reply-To: <CAFArqsb5XcHnUM4_ju-39BVxDuc59bCD9vWLhA+QSsoU-BKEDg@mail.gmail.com>
References: <CAMKwYLASs++ZxCNiAQ1yJMB+oROwwPAr8MuZ65g4dyQ-6nY_fg@mail.gmail.com>
	<CAFArqsbMZXD4pK2W65sNOQpgn7TdCR9wec8YFwM6NLYFyftdQw@mail.gmail.com>
	<CAMKwYLAdOvXUz0XLcJX653MwDZbYaoGO=q7kH8QYm7xx0urGBA@mail.gmail.com>
	<CAFArqsb5XcHnUM4_ju-39BVxDuc59bCD9vWLhA+QSsoU-BKEDg@mail.gmail.com>
Message-ID: <CAKbxfG9YBOzX40shFOCbs2_at3zXGOdEeGT2_vygLpnpmbT_3A@mail.gmail.com>

try going to your wss address in firefox as an https addr.
Replace wss:// with https://  and paste it in the location toolbar.

You may get the trust warning and have to set exception on the cert.  They
won't tell you this when you hit it as wss and it will not work silently
forever.
Once you trust the cert over https it will allow it for subsequent hits as
wss.




On Mon, Aug 5, 2013 at 11:59 AM, Karsten Horsmann <khorsmann at gmail.com>wrote:

> Hi Gustavo,
>
>
> i think there am more than one Problem with WebRTC + sipml5 + Firefox 22
> (or higher).
>
> First, i saw an dtls error and many "use 1.0.1e openssl" hints.
>
> You can check this with openssl client. Here a good and a broken example:
>
> openssl s_client -dtls1 -connect webrtc.freeswitch.org:7443
> CONNECTED(00000003)
>
>
> openssl s_client -dtls1 -connect some.broken.host:7443
> CONNECTED(00000003)
> write:errno=111
> no peer certificate available
> No client certificate CA names sent
> SSL handshake has read 0 bytes and written 0 bytes
> New, (NONE), Cipher is (NONE)
> Secure Renegotiation IS NOT supported
> Compression: NONE
> Expansion: NONE
> SSL-Session:
>     Protocol  : DTLSv1
>     Cipher    : 0000
>     Session-ID:
>     Session-ID-ctx:
>     Master-Key:
>     Key-Arg   : None
>     Krb5 Principal: None
>     PSK identity: None
>     PSK identity hint: None
>     Start Time: 1375716502
>     Timeout   : 7200 (sec)
>     Verify return code: 0 (ok)
>
>
> Then you can check https://webrtc.freeswitch.org/sipml5/ with Firefox 22.
> It works like a charm.
>
> I thought then, maybe some voodoo was made on the sipml5 code. But there
> was a few settings insert. And i test exact the same sipml5 code with my
> test WebRTC FreeSWITCHes, it doesnt work with Firefox.
>
> Only Chrome works.
>
> In Firefox Webdeveloper Console i got only an SDP Warning of the sipml5
> Client:
>
> ({name:"INVALID_SESSION_DESCRIPTION", message:"Could not negotiate answer
> SDP; cause = ERR | SDP Parsing Error:  Warning: Unrecognized attribute
> (msid-semantic)  | SDP Parsing Error:  Warning: Unrecognized attribute
> (ssrc)  | SDP Parsing Error:  Warning: Unrecognized attribute (ssrc)  | SDP
> Parsing Error:  Warning: Unrecognized attribute (ssrc)  | SDP Parsing
> Error:  Warning: Unrecognized attribute (ssrc) ",
> __exposedProps__:{name:"rw", message:"rw"}})
>
> Maybe the developers give us another hint?
>
> Cheers...
>
>
> 2013/8/5 Gustavo Salazar <guga.salazar.loor at gmail.com>
>
>> Hi Karsten,
>>
>> I haven't solved the problem, it works only with Google Chrome.  I
>>  tested the master branch and the version 1.4 with the same results. ..
>>
>>
>
>
> --
> Mit freundlichen Gr??en
> *Karsten Horsmann*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From sravi123 at yahoo.com  Mon Aug  5 18:18:15 2013
From: sravi123 at yahoo.com (Ravi S)
Date: Mon, 05 Aug 2013 19:48:15 +0530
Subject: [Freeswitch-users] Help with PRI Configuration - Bharti Airtel
 - Sangoma 101DE - Freeswitch
In-Reply-To: <CAKzWOxUDrQ=bw5wPsVCwAkO=yZSdYHGjFq6=N=rVJMh8wAfvhw@mail.gmail.com>
References: <51FEBEF0.6090500@yahoo.com>
	<B757A1DB-090B-479E-A18D-001A7B4C0F6D@freeswitch.org>
	<CAKzWOxUDrQ=bw5wPsVCwAkO=yZSdYHGjFq6=N=rVJMh8wAfvhw@mail.gmail.com>
Message-ID: <51FFB427.7040604@yahoo.com>

Michael,

Thank you for your response. The PRI is supplied through copper cables. 
A modem receives it and through an RJ45 cable, I connect it to the 
Freeswitch server. Per the service provider it is a PRI connection. I 
will check with them to confirm that this the case.

Thanks.
Ravi

On 05/08/13 7:12 AM, Michael Collins wrote:
>
> Do they supply a "real" PRI or is it an IP connection with a device to 
> convert to PRI?
>
> On Aug 4, 2013 4:43 PM, "Ken Rice" <krice at freeswitch.org 
> <mailto:krice at freeswitch.org>> wrote:
>
>     you should review the instructions for setting up PRI on the fs
>     wiki... a prindoes not have a username or a password
>
>     Ken
>     Sent from my iPad
>
>     On Aug 4, 2013, at 15:52, Ravi <sravi123 at yahoo.com
>     <mailto:sravi123 at yahoo.com>> wrote:
>
>     > Hello Everyone !
>     >
>     > I am from India. I have recently taken a PRI connection from Bharti
>     > Airtel, one of the service providers. I have installed the
>     following:
>     >
>     > Cent OS
>     > Freeswitch, FreeTDM
>     > Sangoma Card
>     >
>     >
>     > I think, I have followed all the instructions. I am struggling to
>     > configure Freeswitch to start using the PRI connection. This is
>     what I
>     > have from the freeswitch cook book, to configure the gateway: we
>     need
>     > username/password, server address or IP and port.
>     >
>     > When I checked with Airtel, they are telling me that they only give
>     > username/password and IP address details for an internet
>     connection and
>     > not for PRI lines.
>     >
>     > Has anyone here in the list, tried using an indian service
>     provider and
>     > configured in Freeswitch?
>     > Has anyone done it for Airtel ?
>     > Or please help me to figure out how to configure the PRI so as
>     to make
>     > inbound and outbound calls ??
>     >
>     > Any help is much appreciated.
>     >
>     > Thanks.
>     > Ravi
>     > +91-7502029000 <tel:%2B91-7502029000>
>     >
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     > http://www.freeswitchsolutions.com
>     >
>     > 
>     > 
>     >
>     > Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://wiki.freeswitch.org
>     > http://www.cluecon.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
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sravi123 at yahoo.com  Mon Aug  5 20:36:12 2013
From: sravi123 at yahoo.com (Ravi)
Date: Mon, 5 Aug 2013 09:36:12 -0700 (PDT)
Subject: [Freeswitch-users] Help with PRI Configuration - Bharti Airtel
	- Sangoma 101DE - Freeswitch
In-Reply-To: <CAKzWOxUDrQ=bw5wPsVCwAkO=yZSdYHGjFq6=N=rVJMh8wAfvhw@mail.gmail.com>
References: <51FEBEF0.6090500@yahoo.com>	<B757A1DB-090B-479E-A18D-001A7B4C0F6D@freeswitch.org>
	<CAKzWOxUDrQ=bw5wPsVCwAkO=yZSdYHGjFq6=N=rVJMh8wAfvhw@mail.gmail.com>
Message-ID: <1375720572.96851.YahooMailNeo@web160505.mail.bf1.yahoo.com>

Thanks Ken,
Previously I did not have trouble installing the Sangoma card. Now when I try to go through the FS freetdm wiki, this is the error that I am getting:

[root at bfree-server wanpipe-7.0.5]# make freetdm
?? Error linux headers/source not found: /lib/modules/2.6.32-358.14.1.el6.x86_64/build !

make: *** [_checksrc] Error 1


any reason why this is happening ??

Thanks.
Ravi




________________________________
 From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Monday, August 5, 2013 7:12 AM
Subject: Re: [Freeswitch-users] Help with PRI Configuration - Bharti Airtel - Sangoma 101DE - Freeswitch
 


Do they supply a "real" PRI or is it an IP connection with a device to convert to PRI?
On Aug 4, 2013 4:43 PM, "Ken Rice" <krice at freeswitch.org> wrote:

you should review the instructions for setting up PRI on the fs wiki... a prindoes not have a username or a password
>
>Ken
>Sent from my iPad
>
>On Aug 4, 2013, at 15:52, Ravi <sravi123 at yahoo.com> wrote:
>
>> Hello Everyone !
>>
>> I am from India. I have recently taken a PRI connection from Bharti
>> Airtel, one of the service providers. I have installed the following:
>>
>> Cent OS
>> Freeswitch, FreeTDM
>> Sangoma Card
>>
>>
>> I think, I have followed all the instructions. I am struggling to
>> configure Freeswitch to start using the PRI connection. This is what I
>> have from the freeswitch cook book, to configure the gateway: we need
>> username/password, server address or IP and port.
>>
>> When I checked with Airtel, they are telling me that they only give
>> username/password and IP address details for an internet connection and
>> not for PRI lines.
>>
>> Has anyone here in the list, tried using an indian service provider and
>> configured in Freeswitch?
>> Has anyone done it for Airtel ?
>> Or please help me to figure out how to configure the PRI so as to make
>> inbound and outbound calls ??
>>
>> Any help is much appreciated.
>>
>> Thanks.
>> Ravi
>> +91-7502029000
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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
>
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 csiwek at orange-vallee.net  Mon Aug  5 19:45:30 2013
From: csiwek at orange-vallee.net (Cezary Siwek)
Date: Mon, 05 Aug 2013 16:45:30 +0100
Subject: [Freeswitch-users] Can't call from Firefox 22 to Freeswitch
 using sipml5
In-Reply-To: <CAMKwYLAdOvXUz0XLcJX653MwDZbYaoGO=q7kH8QYm7xx0urGBA@mail.gmail.com>
References: <CAMKwYLASs++ZxCNiAQ1yJMB+oROwwPAr8MuZ65g4dyQ-6nY_fg@mail.gmail.com>
	<CAFArqsbMZXD4pK2W65sNOQpgn7TdCR9wec8YFwM6NLYFyftdQw@mail.gmail.com>
	<CAMKwYLAdOvXUz0XLcJX653MwDZbYaoGO=q7kH8QYm7xx0urGBA@mail.gmail.com>
Message-ID: <51FFC89A.4020807@orange-vallee.net>

Hi,

Try Firefox 23 (nightly build). It should have better webrtc support.

Regards


On 05/08/2013 16:16, Gustavo Salazar wrote:
> Hi Karsten,
>
> I haven't solved the problem, it works only with Google Chrome.  I 
>  tested the master branch and the version 1.4 with the same results. ..
>
>
>
>
> 2013/8/5 Karsten Horsmann <khorsmann at gmail.com 
> <mailto:khorsmann at gmail.com>>
>
>     Hi all,
>
>
>     i have still the same problem, with 1.4beta branch and master
>     branch. Firefox 22 still no audio.
>     Did you get any solution with your Problem?
>
>
>
>
>     2013/7/26 Gustavo Salazar <guga.salazar.loor at gmail.com
>     <mailto:guga.salazar.loor at gmail.com>>
>
>
>         Hi,
>
>         I want to make calls using sipml5 and freeswitch. I have
>         installed freeswitch from the git repository in an ec2
>         instance with elastic ip address, configured
>         sip_profiles/internal.xml with the EIP and here the results:
>
>         Using Google Chrome and the sipml5 client I register as user
>         1001 with password 1234 , I dial 5000 and I hear the IVR
>         voice, also I can join to a conference I created, and hear
>         myself in an echo test dialing 9196. No problem with that.
>
>         Using Firefox 22 and the same sipml5 client I register as user
>         1002 and password 1234,  then dial 5000, I see the audio files
>         are played in the freeswitch cli but I can not hear anything.
>
>         I guess the problem is not in my sipml5 client neither Firefox
>         because when I use them to register to webrtc.freeswitch.org
>         <http://webrtc.freeswitch.org> there is not any problem, I
>         hear the voice after dialing 5000.
>
>         I think I am missing some configuration in my Freeswitch
>         server, where I also loaded the opus module and set OPUS in
>         the file vars.xml , in the parameters global_codec_prefs
>         and outbound_codec_prefs.
>
>         I attach the console output from Freeswitch and the console
>         output from Firefox
>
>         Here the freeswitch output
>         http://pastebin.com/S0yJxT1J
>
>         Here the Firefox console output
>         http://pastebin.com/8kBGdY3J
>
>         Thanks in advance for the help
>
>
>
>         -- 
>         Gustavo Salazar
>
>
>
>
>
>     -- 
>     Mit freundlichen Gr??en
>     *Karsten Horsmann*
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
>
> -- 
> Gustavo Salazar
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 khorsmann at gmail.com  Mon Aug  5 21:36:17 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Mon, 5 Aug 2013 19:36:17 +0200
Subject: [Freeswitch-users] Can't call from Firefox 22 to Freeswitch
	using sipml5
In-Reply-To: <CAKbxfG9YBOzX40shFOCbs2_at3zXGOdEeGT2_vygLpnpmbT_3A@mail.gmail.com>
References: <CAMKwYLASs++ZxCNiAQ1yJMB+oROwwPAr8MuZ65g4dyQ-6nY_fg@mail.gmail.com>
	<CAFArqsbMZXD4pK2W65sNOQpgn7TdCR9wec8YFwM6NLYFyftdQw@mail.gmail.com>
	<CAMKwYLAdOvXUz0XLcJX653MwDZbYaoGO=q7kH8QYm7xx0urGBA@mail.gmail.com>
	<CAFArqsb5XcHnUM4_ju-39BVxDuc59bCD9vWLhA+QSsoU-BKEDg@mail.gmail.com>
	<CAKbxfG9YBOzX40shFOCbs2_at3zXGOdEeGT2_vygLpnpmbT_3A@mail.gmail.com>
Message-ID: <CAFArqsb1M=vSMTGtg8spw-ZQKM+p1Lp-bqz=G+aCowLFkU59MA@mail.gmail.com>

Hello Anthony,


i did that and Firefox dont said its an untrused certificate. Its an
trusted certificate.
I made an debug-Logfile and test it with git-master.

d1268e81036ce8ce00de8ee22f387cdbf43a7203

switch_rtp.c:5668 Skip sending audio packet 88 bytes (ice not ready!)
is to see in the logfile over and over again, if the rtp should send.

FreeSWITCH Debug-Logfile:
*http://pastebin.freeswitch.org/21270*

[19:17:03.572] ({name:"INVALID_SESSION_DESCRIPTION", message:"Could not
negotiate answer SDP; cause = ERR | SDP Parsing Error:  Warning:
Unrecognized attribute (msid-semantic)  | SDP Parsing Error:  Warning:
Unrecognized attribute (ssrc)  | SDP Parsing Error:  Warning: Unrecognized
attribute (ssrc)  | SDP Parsing Error:  Warning: Unrecognized attribute
(ssrc)  | SDP Parsing Error:  Warning: Unrecognized attribute (ssrc) ",
__exposedProps__:{name:"rw", message:"rw"}})

Firefox 22 Webdeveloper-Logfile:
*http://pastebin.freeswitch.org/21268*


Hope everything is in there to take a look.

Cheers.
Karsten

2013/8/5 Anthony Minessale <anthony.minessale at gmail.com>

> try going to your wss address in firefox as an https addr.
> Replace wss:// with https://  and paste it in the location toolbar.
>
> You may get the trust warning and have to set exception on the cert.  They
> won't tell you this when you hit it as wss and it will not work silently
> forever.
> Once you trust the cert over https it will allow it for subsequent hits as
> wss.
>
>
>
>
>
> --
> Anthony Minessale II
>
> 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From jack at livecall.com  Mon Aug  5 22:15:21 2013
From: jack at livecall.com (Jack)
Date: Mon, 05 Aug 2013 11:15:21 -0700
Subject: [Freeswitch-users] DTMF issues
In-Reply-To: <1375381751492-7593499.post@n2.nabble.com>
References: <1375381751492-7593499.post@n2.nabble.com>
Message-ID: <51FFEBB9.5020209@livecall.com>

We are having a similar experience and found that our numbers that were 
ported through Level3 were not transmitting DTMF reliably . The solution 
was to have our provider port them through Peatec.
Hope that helps..
Jack


On 8/1/2013 11:29 AM, baskar wrote:
> Hi All,
>
> Our current existing set up has three different SIP service providers and
> all three were working fine, but all of a sudden on one of our SIP service
> provider we started to have DTMF issues. We followed up with the problematic
> service provider and they kept saying that there is no issue at their end,
> now we have connected the non working SIP link to our test server and we are
> not having any DTMF issues while using it on the test server. I have
> attached log on pastebin.
>
> Freeswitch Live server log
> http://pastebin.com/mQ27rDts
>
> Freeswitch test server log
> http://pastebin.com/4BEeUcSu
>
> Please some one help me to resolve this issues.
>
> Thanks,
> N.Baskar
>
>
>
>
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/DTMF-issues-tp7593499.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 khorsmann at gmail.com  Mon Aug  5 22:19:33 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Mon, 5 Aug 2013 20:19:33 +0200
Subject: [Freeswitch-users] Can't call from Firefox 22 to Freeswitch
	using sipml5
In-Reply-To: <51FFC89A.4020807@orange-vallee.net>
References: <CAMKwYLASs++ZxCNiAQ1yJMB+oROwwPAr8MuZ65g4dyQ-6nY_fg@mail.gmail.com>
	<CAFArqsbMZXD4pK2W65sNOQpgn7TdCR9wec8YFwM6NLYFyftdQw@mail.gmail.com>
	<CAMKwYLAdOvXUz0XLcJX653MwDZbYaoGO=q7kH8QYm7xx0urGBA@mail.gmail.com>
	<51FFC89A.4020807@orange-vallee.net>
Message-ID: <CAFArqsYXECis-KFXstadKGSNjHhs-oePr-UbSYy67bqKndDuHg@mail.gmail.com>

Hi Cezary,


but the question is, why Firefox 22 works with sipml5 +
https://webrtc.freeswitch.org/sipml5/
and *not* with my installation?



2013/8/5 Cezary Siwek <csiwek at orange-vallee.net>

>  Hi,
>
> Try Firefox 23 (nightly build). It should have better webrtc support.
>
> Regards
>
>

-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From anthony.minessale at gmail.com  Mon Aug  5 22:52:40 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 5 Aug 2013 13:52:40 -0500
Subject: [Freeswitch-users] Can't call from Firefox 22 to Freeswitch
	using sipml5
In-Reply-To: <CAFArqsYXECis-KFXstadKGSNjHhs-oePr-UbSYy67bqKndDuHg@mail.gmail.com>
References: <CAMKwYLASs++ZxCNiAQ1yJMB+oROwwPAr8MuZ65g4dyQ-6nY_fg@mail.gmail.com>
	<CAFArqsbMZXD4pK2W65sNOQpgn7TdCR9wec8YFwM6NLYFyftdQw@mail.gmail.com>
	<CAMKwYLAdOvXUz0XLcJX653MwDZbYaoGO=q7kH8QYm7xx0urGBA@mail.gmail.com>
	<51FFC89A.4020807@orange-vallee.net>
	<CAFArqsYXECis-KFXstadKGSNjHhs-oePr-UbSYy67bqKndDuHg@mail.gmail.com>
Message-ID: <CAKbxfG8JiU20-Atc58Wyt3pHSoUptsecMgoQq47iyu-DV2aHgA@mail.gmail.com>

The ice not ready message comes from not getting any stun messages to the
candidate ip/port sent in the 200ok.
Maybe do a pcap and see if the firefox is sending any stun packets to that
addr.




On Mon, Aug 5, 2013 at 1:19 PM, Karsten Horsmann <khorsmann at gmail.com>wrote:

> Hi Cezary,
>
>
> but the question is, why Firefox 22 works with sipml5 +
> https://webrtc.freeswitch.org/sipml5/
> and *not* with my installation?
>
>
>
> 2013/8/5 Cezary Siwek <csiwek at orange-vallee.net>
>
>>  Hi,
>>
>> Try Firefox 23 (nightly build). It should have better webrtc support.
>>
>> Regards
>>
>>
>
> --
> Mit freundlichen Gr??en
> *Karsten Horsmann*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From khorsmann at gmail.com  Mon Aug  5 23:58:38 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Mon, 5 Aug 2013 21:58:38 +0200
Subject: [Freeswitch-users] Can't call from Firefox 22 to Freeswitch
	using sipml5
In-Reply-To: <CAKbxfG8JiU20-Atc58Wyt3pHSoUptsecMgoQq47iyu-DV2aHgA@mail.gmail.com>
References: <CAMKwYLASs++ZxCNiAQ1yJMB+oROwwPAr8MuZ65g4dyQ-6nY_fg@mail.gmail.com>
	<CAFArqsbMZXD4pK2W65sNOQpgn7TdCR9wec8YFwM6NLYFyftdQw@mail.gmail.com>
	<CAMKwYLAdOvXUz0XLcJX653MwDZbYaoGO=q7kH8QYm7xx0urGBA@mail.gmail.com>
	<51FFC89A.4020807@orange-vallee.net>
	<CAFArqsYXECis-KFXstadKGSNjHhs-oePr-UbSYy67bqKndDuHg@mail.gmail.com>
	<CAKbxfG8JiU20-Atc58Wyt3pHSoUptsecMgoQq47iyu-DV2aHgA@mail.gmail.com>
Message-ID: <CAFArqsYho0u56kmenj9PHPSjAMykpVSbh_1fC6KGd1tf5u0iKg@mail.gmail.com>

Hello Anthony,

i made pcaps from the FS-Server and a Firefox 22 Client. Filtered on STUN
and ICMP.

https://webrtc.0x300.de/pcap/client-stun.pcap  31K
https://webrtc.0x300.de/pcap/server-stun.pcap   8K

It seems that my gateway sends icmp (port unreachable), but i dunno why -
whats so diffrent between the Chrome doing and Firefox?

Regards

2013/8/5 Anthony Minessale <anthony.minessale at gmail.com>

> The ice not ready message comes from not getting any stun messages to the
> candidate ip/port sent in the 200ok.
> Maybe do a pcap and see if the firefox is sending any stun packets to that
> addr.
>
>
>

-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From khorsmann at gmail.com  Tue Aug  6 00:07:38 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Mon, 5 Aug 2013 22:07:38 +0200
Subject: [Freeswitch-users] Softphone
In-Reply-To: <CAMvhjJs6zF6Th__8y4qBKDJ2vQrxnvNdibArbsa9mDZbzF1jJg@mail.gmail.com>
References: <CAMvhjJsP+vY1q3s0bDHdOfiDh37Wwk1g1SSsvi-8fJVNcj4YUw@mail.gmail.com>
	<CAMvhjJs6zF6Th__8y4qBKDJ2vQrxnvNdibArbsa9mDZbzF1jJg@mail.gmail.com>
Message-ID: <CAFArqsZ7eWoMnWnF2QQ0eagVsh8EcQraUY6w5srLoChe74b9Dg@mail.gmail.com>

Hi Ashish,

connect with ssh to your FreeSWITCH Server and run "fs_cli" to see what
happen if you try to xlite your box.
Visite the fine docu
https://wiki.freeswitch.org/wiki/Getting_Started_Guide#directory and take a
look if your credentials are correct.


2013/8/5 Ashish Mishra <itsme.kunnu at gmail.com>

> Sir i have installed xlite soft phone on my windows 8 pc and i am trying
> to connect to the freeswitch server running on my ubuntu machine but when i
> am trying to enable the account it displays an error that "account not
> enabled contact ur administrator" also i receive the same errror when i
> connect my windows 8 pc running xlite softphone with a windows 7 machine
> acting as server through a local LAN.
>
> Thank you
> Ashish Mishra
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From anthony.minessale at gmail.com  Tue Aug  6 00:16:48 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 5 Aug 2013 15:16:48 -0500
Subject: [Freeswitch-users] Can't call from Firefox 22 to Freeswitch
	using sipml5
In-Reply-To: <CAFArqsYho0u56kmenj9PHPSjAMykpVSbh_1fC6KGd1tf5u0iKg@mail.gmail.com>
References: <CAMKwYLASs++ZxCNiAQ1yJMB+oROwwPAr8MuZ65g4dyQ-6nY_fg@mail.gmail.com>
	<CAFArqsbMZXD4pK2W65sNOQpgn7TdCR9wec8YFwM6NLYFyftdQw@mail.gmail.com>
	<CAMKwYLAdOvXUz0XLcJX653MwDZbYaoGO=q7kH8QYm7xx0urGBA@mail.gmail.com>
	<51FFC89A.4020807@orange-vallee.net>
	<CAFArqsYXECis-KFXstadKGSNjHhs-oePr-UbSYy67bqKndDuHg@mail.gmail.com>
	<CAKbxfG8JiU20-Atc58Wyt3pHSoUptsecMgoQq47iyu-DV2aHgA@mail.gmail.com>
	<CAFArqsYho0u56kmenj9PHPSjAMykpVSbh_1fC6KGd1tf5u0iKg@mail.gmail.com>
Message-ID: <CAKbxfG-bSG7sgZN039WaZ5GhpcR9sP19hy+-D2FkweMzEURnhw@mail.gmail.com>

Make sure its going to the ports that are advertised in the sdp and that
there are no nat or firewall issues in between.


On Mon, Aug 5, 2013 at 2:58 PM, Karsten Horsmann <khorsmann at gmail.com>wrote:

> Hello Anthony,
>
> i made pcaps from the FS-Server and a Firefox 22 Client. Filtered on STUN
> and ICMP.
>
> https://webrtc.0x300.de/pcap/client-stun.pcap  31K
> https://webrtc.0x300.de/pcap/server-stun.pcap   8K
>
> It seems that my gateway sends icmp (port unreachable), but i dunno why -
> whats so diffrent between the Chrome doing and Firefox?
>
> Regards
>
>
> 2013/8/5 Anthony Minessale <anthony.minessale at gmail.com>
>
>> The ice not ready message comes from not getting any stun messages to the
>> candidate ip/port sent in the 200ok.
>> Maybe do a pcap and see if the firefox is sending any stun packets to
>> that addr.
>>
>>
>>
>
> --
> Mit freundlichen Gr??en
> *Karsten Horsmann*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From khorsmann at gmail.com  Tue Aug  6 00:41:31 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Mon, 5 Aug 2013 22:41:31 +0200
Subject: [Freeswitch-users] Can't call from Firefox 22 to Freeswitch
	using sipml5
In-Reply-To: <CAKbxfG-bSG7sgZN039WaZ5GhpcR9sP19hy+-D2FkweMzEURnhw@mail.gmail.com>
References: <CAMKwYLASs++ZxCNiAQ1yJMB+oROwwPAr8MuZ65g4dyQ-6nY_fg@mail.gmail.com>
	<CAFArqsbMZXD4pK2W65sNOQpgn7TdCR9wec8YFwM6NLYFyftdQw@mail.gmail.com>
	<CAMKwYLAdOvXUz0XLcJX653MwDZbYaoGO=q7kH8QYm7xx0urGBA@mail.gmail.com>
	<51FFC89A.4020807@orange-vallee.net>
	<CAFArqsYXECis-KFXstadKGSNjHhs-oePr-UbSYy67bqKndDuHg@mail.gmail.com>
	<CAKbxfG8JiU20-Atc58Wyt3pHSoUptsecMgoQq47iyu-DV2aHgA@mail.gmail.com>
	<CAFArqsYho0u56kmenj9PHPSjAMykpVSbh_1fC6KGd1tf5u0iKg@mail.gmail.com>
	<CAKbxfG-bSG7sgZN039WaZ5GhpcR9sP19hy+-D2FkweMzEURnhw@mail.gmail.com>
Message-ID: <CAFArqsZA1DtMoK_OuEDEjcOVJ+mEeF_g+a9-K5Jf2sWf7aDFFA@mail.gmail.com>

Hi Anthony,

how is the best way to check that? With wireshark most stuff is ssl unlike
normal UDP/SIP stuff.

Iam still confused, that it works with webrtc.freeswitch.org at work and at
home, but not with my example setup.

I checked for iptables (everything fine), and its no cloud virtual foo
where the server is running.

Thanks in advance,
Regards

2013/8/5 Anthony Minessale <anthony.minessale at gmail.com>

> Make sure its going to the ports that are advertised in the sdp and that
> there are no nat or firewall issues in between.
>
>


-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From itsme.kunnu at gmail.com  Tue Aug  6 01:00:45 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Tue, 6 Aug 2013 02:30:45 +0530
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
Message-ID: <CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>

When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i
have installed freeswitch) it gives me the following error :
fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
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From anthony.minessale at gmail.com  Tue Aug  6 01:02:12 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 5 Aug 2013 16:02:12 -0500
Subject: [Freeswitch-users] Can't call from Firefox 22 to Freeswitch
	using sipml5
In-Reply-To: <CAFArqsZA1DtMoK_OuEDEjcOVJ+mEeF_g+a9-K5Jf2sWf7aDFFA@mail.gmail.com>
References: <CAMKwYLASs++ZxCNiAQ1yJMB+oROwwPAr8MuZ65g4dyQ-6nY_fg@mail.gmail.com>
	<CAFArqsbMZXD4pK2W65sNOQpgn7TdCR9wec8YFwM6NLYFyftdQw@mail.gmail.com>
	<CAMKwYLAdOvXUz0XLcJX653MwDZbYaoGO=q7kH8QYm7xx0urGBA@mail.gmail.com>
	<51FFC89A.4020807@orange-vallee.net>
	<CAFArqsYXECis-KFXstadKGSNjHhs-oePr-UbSYy67bqKndDuHg@mail.gmail.com>
	<CAKbxfG8JiU20-Atc58Wyt3pHSoUptsecMgoQq47iyu-DV2aHgA@mail.gmail.com>
	<CAFArqsYho0u56kmenj9PHPSjAMykpVSbh_1fC6KGd1tf5u0iKg@mail.gmail.com>
	<CAKbxfG-bSG7sgZN039WaZ5GhpcR9sP19hy+-D2FkweMzEURnhw@mail.gmail.com>
	<CAFArqsZA1DtMoK_OuEDEjcOVJ+mEeF_g+a9-K5Jf2sWf7aDFFA@mail.gmail.com>
Message-ID: <CAKbxfG_GUuHwT_V_H0HYRPA=sQNp9VGu9txtWXYZ=A0OspR65g@mail.gmail.com>

Your example might have different network conditions than the demo.
You can check by enabling sip trace on FS and bring up the call and look at
the ports that are advertised out.
You can try turning off selinux or iptables as a test.




On Mon, Aug 5, 2013 at 3:41 PM, Karsten Horsmann <khorsmann at gmail.com>wrote:

> Hi Anthony,
>
> how is the best way to check that? With wireshark most stuff is ssl unlike
> normal UDP/SIP stuff.
>
> Iam still confused, that it works with webrtc.freeswitch.org at work and
> at home, but not with my example setup.
>
> I checked for iptables (everything fine), and its no cloud virtual foo
> where the server is running.
>
> Thanks in advance,
> Regards
>
>
> 2013/8/5 Anthony Minessale <anthony.minessale at gmail.com>
>
>> Make sure its going to the ports that are advertised in the sdp and that
>> there are no nat or firewall issues in between.
>>
>>
>
>
> --
> Mit freundlichen Gr??en
> *Karsten Horsmann*
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From itsme.kunnu at gmail.com  Tue Aug  6 01:04:00 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Tue, 6 Aug 2013 02:34:00 +0530
Subject: [Freeswitch-users] Softphone
In-Reply-To: <CAFArqsZ7eWoMnWnF2QQ0eagVsh8EcQraUY6w5srLoChe74b9Dg@mail.gmail.com>
References: <CAMvhjJsP+vY1q3s0bDHdOfiDh37Wwk1g1SSsvi-8fJVNcj4YUw@mail.gmail.com>
	<CAMvhjJs6zF6Th__8y4qBKDJ2vQrxnvNdibArbsa9mDZbzF1jJg@mail.gmail.com>
	<CAFArqsZ7eWoMnWnF2QQ0eagVsh8EcQraUY6w5srLoChe74b9Dg@mail.gmail.com>
Message-ID: <CAMvhjJv_zk_TbA+qFwa4_PJkmAbTNqe9VV8dTdgDCYDD60+uOw@mail.gmail.com>

Sir i am getting an error when i am launching fs_cli on my ubuntu
machine...the error being:
fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
Hi Ashish,

connect with ssh to your FreeSWITCH Server and run "fs_cli" to see what
happen if you try to xlite your box.
Visite the fine docu
https://wiki.freeswitch.org/wiki/Getting_Started_Guide#directory and take a
look if your credentials are correct.
Hi Ashish,

connect with ssh to your FreeSWITCH Server and run "fs_cli" to see what
happen if you try to xlite your box.
Visite the fine docu
https://wiki.freeswitch.org/wiki/Getting_Started_Guide#directory and take a
look if your credentials are correct.


2013/8/5 Ashish Mishra <itsme.kunnu at gmail.com>

> Sir i have installed xlite soft phone on my windows 8 pc and i am trying
> to connect to the freeswitch server running on my ubuntu machine but when i
> am trying to enable the account it displays an error that "account not
> enabled contact ur administrator" also i receive the same errror when i
> connect my windows 8 pc running xlite softphone with a windows 7 machine
> acting as server through a local LAN.
>
> Thank you
> Ashish Mishra
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Mit freundlichen Gr??en
*Karsten Horsmann*

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 mishehu at freeswitch.org  Tue Aug  6 01:38:11 2013
From: mishehu at freeswitch.org (Yossi Neiman)
Date: Mon, 05 Aug 2013 16:38:11 -0500
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
Message-ID: <52001B43.9080901@freeswitch.org>

Do you have mod_event_socket loaded?

-Yossi

On 08/05/2013 04:00 PM, Ashish Mishra wrote:
>
> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which 
> i have installed freeswitch) it gives me the following error :
> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bob at experient.com  Tue Aug  6 02:01:16 2013
From: bob at experient.com (Bob McCarthy)
Date: Mon, 5 Aug 2013 16:01:16 -0600
Subject: [Freeswitch-users] How to attche custom "content-type" in
	Freeswitch SIP message
In-Reply-To: <50D08AA095C54A7188B2C7E9F833D199@gmail.com>
References: <201305250918139069169@163.com>
	<50D08AA095C54A7188B2C7E9F833D199@gmail.com>
Message-ID: <01d401ce9227$4ff92170$efeb6450$@experient.com>

Is there another way to create a mutipart body ? I am using this successfully to create an invite but it breaks the CDR-XML log.

<action application="set"><![CDATA[sip_mp_application/pidf+xml=${geolocationbyvalue}]]></action>

 

The ?/? and the ?+? break the CDR-XML when the sip_mp_application/pidf+xml variable is converted into an XML tag

FS-5403 fixed the reading of the ?/? but does not address the writing and the ?+? is also an issue

I have left a comment on jira, I am not sure if I need to open a new one.

 

Bob McCarthy

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Seven Du
Sent: Friday, May 24, 2013 8:39 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] How to attche custom "content-type" in Freeswitch SIP message

 

FS defined SOFIA_MULTIPART_PREFIX  which should can support multipart body. not sure if you can find the wiki. maybe you should look the source code and update the wiki when you find out.

 

I had tried that but I forget the procedure. it should be some chan vars like sip_mp_ ...

 

-- 

Seven Du

http://www.freeswitch.org.cn

http://about.me/dujinfang

http://www.dujinfang.com

 

Sent with Sparrow <http://www.sparrowmailapp.com/?sig> 

 

On Saturday, May 25, 2013 at 9:18 AM, xiaofengcanyuexp at 163.com <mailto:xiaofengcanyuexp at 163.com>  wrote:

Dear support,

 

I''m trying to encapsulate my private "application/isup" in the SIP Msg. Normally, it should like below example.

 

It firstly addresses "Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR". And then can write private "content-type" like "application/sdp".

Now I can see the application/sdp is encapsulated via variable "switch_r_sdp".

Is there anyway to encapsulated other customized "content-type"? 

 

Appreciate to get your reply.

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

Here is an example of SIP message which encapsulated "applicaiton/sdp" and "application/isup".

 

INVITE sip: 87896677 at dance.com <mailto:87896677 at dance.com> ;user=phone;SIP/2.0

From: "Caller"<sip:678923001 at skynetwork.com <mailto:678923001 at skynetwork.com> >

To: <sip:87896677 at dance.com <mailto:87896677 at dance.com> >;user=phone

Call-ID: QRLVLNKeKx-WDHAuwlEkR-EwhPPcTHOP at skynetwork.com <mailto:QRLVLNKeKx-WDHAuwlEkR-EwhPPcTHOP at skynetwork.com> 

Content-Type: multipart/mixed;boundary=QRLVLNKeKxWDHAuwlEkR

MIME-Version: 1.0

Content-Length: 433

--QRLVLNKeKxWDHAuwlEkR

Content-Type: application/sdp

User-Agent: ENSR2.5.46.6-IS2-RMRG36-RG20-CPO487

Content-Length: 142

 

v=0

o=- 1706944438 1706944438 IN IP4 192.168.1.105

s=ENSResip

t=0 0

m=audio 6793 RTP/AVP 0

a=rtpmap:0 PCMU/8000

--QRLVLNKeKxWDHAuwlEkR

Content-Type: application/isup; version=ansi;base=ansi00

Content-Disposition: signal; handling=optional

 

01 00 60 01 0a 03 05 0b 02 c0 90 06 03 10 78 98 66 77 0a 07 83 13 76 98 32 00 0f 00 

--QRLVLNKeKxWDHAuwlEkR--

 

 

Thanks

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

2013-05-25

 

 

 

 

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org <mailto:consulting at freeswitch.org> 

http://www.freeswitchsolutions.com

 





 

Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.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

 

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From guga.salazar.loor at gmail.com  Tue Aug  6 02:07:18 2013
From: guga.salazar.loor at gmail.com (Gustavo Salazar)
Date: Mon, 5 Aug 2013 17:07:18 -0500
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <52001B43.9080901@freeswitch.org>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<52001B43.9080901@freeswitch.org>
Message-ID: <CAMKwYLCtQV1wCVzPxwhNj2d23eK__uPDYPb98p0ZiJLRqx0GKQ@mail.gmail.com>

Is Freeswitch running?
I have seen a similar error when I tried to start the cli and Freeswitch
was not running .



2013/8/5 Yossi Neiman <mishehu at freeswitch.org>

>  Do you have mod_event_socket loaded?
>
> -Yossi
>
>
> On 08/05/2013 04:00 PM, Ashish Mishra wrote:
>
> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i
> have installed freeswitch) it gives me the following error :
> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Gustavo Salazar
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From dp.siddharth at eng.knowlarity.com  Tue Aug  6 04:40:59 2013
From: dp.siddharth at eng.knowlarity.com (DP Siddharth)
Date: Tue, 6 Aug 2013 06:10:59 +0530
Subject: [Freeswitch-users] Help with PRI Configuration - Bharti Airtel
 - Sangoma 101DE - Freeswitch
In-Reply-To: <51FFB427.7040604@yahoo.com>
References: <51FEBEF0.6090500@yahoo.com>
	<B757A1DB-090B-479E-A18D-001A7B4C0F6D@freeswitch.org>
	<CAKzWOxUDrQ=bw5wPsVCwAkO=yZSdYHGjFq6=N=rVJMh8wAfvhw@mail.gmail.com>
	<51FFB427.7040604@yahoo.com>
Message-ID: <CAFDf6-9WT3HrK5JZYYDFvG+t4pSF9qqxi6gr128j-JCX34-ttQ@mail.gmail.com>

Just check if linux kernel source available on your server? Wanpipe driver
needs that for compilation.

To have sangoma working you also need lbsng_isdn.so or libpri installed.

Install wanpipe-> configure it through wancfg_fs-> install libsng_isdn.so->
compile and install mod_freetdm -> reload mod_freetdm in freeswitch.

Thanks & Regards,
Siddharth
On Aug 5, 2013 11:00 PM, "Ravi S" <sravi123 at yahoo.com> wrote:

>  Michael,
>
> Thank you for your response. The PRI is supplied through copper cables. A
> modem receives it and through an RJ45 cable, I connect it to the Freeswitch
> server. Per the service provider it is a PRI connection. I will check with
> them to confirm that this the case.
>
> Thanks.
> Ravi
>
> On 05/08/13 7:12 AM, Michael Collins wrote:
>
> Do they supply a "real" PRI or is it an IP connection with a device to
> convert to PRI?
> On Aug 4, 2013 4:43 PM, "Ken Rice" <krice at freeswitch.org> wrote:
>
>> you should review the instructions for setting up PRI on the fs wiki... a
>> prindoes not have a username or a password
>>
>> Ken
>> Sent from my iPad
>>
>> On Aug 4, 2013, at 15:52, Ravi <sravi123 at yahoo.com> wrote:
>>
>> > Hello Everyone !
>> >
>> > I am from India. I have recently taken a PRI connection from Bharti
>> > Airtel, one of the service providers. I have installed the following:
>> >
>> > Cent OS
>> > Freeswitch, FreeTDM
>> > Sangoma Card
>> >
>> >
>> > I think, I have followed all the instructions. I am struggling to
>> > configure Freeswitch to start using the PRI connection. This is what I
>> > have from the freeswitch cook book, to configure the gateway: we need
>> > username/password, server address or IP and port.
>> >
>> > When I checked with Airtel, they are telling me that they only give
>> > username/password and IP address details for an internet connection and
>> > not for PRI lines.
>> >
>> > Has anyone here in the list, tried using an indian service provider and
>> > configured in Freeswitch?
>> > Has anyone done it for Airtel ?
>> > Or please help me to figure out how to configure the PRI so as to make
>> > inbound and outbound calls ??
>> >
>> > Any help is much appreciated.
>> >
>> > Thanks.
>> > Ravi
>> > +91-7502029000
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 zhulizhong at live.com  Tue Aug  6 04:59:59 2013
From: zhulizhong at live.com (James zhu)
Date: Tue, 6 Aug 2013 00:59:59 +0000
Subject: [Freeswitch-users] Help with PRI Configuration - Bharti Airtel
 - Sangoma 101DE - Freeswitch
In-Reply-To: <51FEBEF0.6090500@yahoo.com>
References: <51FEBEF0.6090500@yahoo.com>
Message-ID: <BLU171-W115A0A279C706960B044C06B55D0@phx.gbl>

hello:this is a link you can refer:http://wiki.voip88.com/index.php/Linux/Freeswitch/Sangoma/PRIplease try to verify the CRC4 for specific service providers.

Best regards,

James.zhu

website: www.hiastar.com

> Date: Mon, 5 Aug 2013 02:22:00 +0530
> From: sravi123 at yahoo.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Help with PRI Configuration - Bharti Airtel -	Sangoma 101DE - Freeswitch
> 
> Hello Everyone !
> 
> I am from India. I have recently taken a PRI connection from Bharti 
> Airtel, one of the service providers. I have installed the following:
> 
> Cent OS
> Freeswitch, FreeTDM
> Sangoma Card
> 
> 
> I think, I have followed all the instructions. I am struggling to 
> configure Freeswitch to start using the PRI connection. This is what I 
> have from the freeswitch cook book, to configure the gateway: we need 
> username/password, server address or IP and port.
> 
> When I checked with Airtel, they are telling me that they only give 
> username/password and IP address details for an internet connection and 
> not for PRI lines.
> 
> Has anyone here in the list, tried using an indian service provider and 
> configured in Freeswitch?
> Has anyone done it for Airtel ?
> Or please help me to figure out how to configure the PRI so as to make 
> inbound and outbound calls ??
> 
> Any help is much appreciated.
> 
> Thanks.
> Ravi
> +91-7502029000
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at davri.com  Tue Aug  6 06:26:33 2013
From: bdfoster at davri.com (Brian Foster)
Date: Mon, 5 Aug 2013 22:26:33 -0400
Subject: [Freeswitch-users] IVR - Dial by extension
Message-ID: <CAHgk4BXkW2e94Hm3Tx46FpSzWRTGS15vV4pQ3qKv96TogWze6A@mail.gmail.com>

In the dialplan, I use something like this as a condition for the Local
Extension "extension":

<condition field="${user_exists id ${destination_number} $${domain}}"
expression="^true$">

I was wondering if there was something you could do to have a similar
behavior when creating IVR menus. Has anyone tried this? Our extensions
don't fit a particular format, other than 4 digits, so checking beforehand
whether the user id exists would be ideal before transferring.

Thank you,

Brian Foster
Project Manager/Owner's Representative
Davri Investments, Incorporated
P: +1-317-787-2686
M: +1-317-600-9753
Indianapolis, Indiana
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From mahesh.katta at flexydial.com  Tue Aug  6 07:46:35 2013
From: mahesh.katta at flexydial.com (mahesh katta)
Date: Tue, 6 Aug 2013 03:46:35 +0000
Subject: [Freeswitch-users] Help with PRI Configuration - Bharti Airtel
 - Sangoma 101DE - Freeswitch
In-Reply-To: <CAFDf6-9WT3HrK5JZYYDFvG+t4pSF9qqxi6gr128j-JCX34-ttQ@mail.gmail.com>
References: <51FEBEF0.6090500@yahoo.com>
	<B757A1DB-090B-479E-A18D-001A7B4C0F6D@freeswitch.org>
	<CAKzWOxUDrQ=bw5wPsVCwAkO=yZSdYHGjFq6=N=rVJMh8wAfvhw@mail.gmail.com>
	<51FFB427.7040604@yahoo.com>
	<CAFDf6-9WT3HrK5JZYYDFvG+t4pSF9qqxi6gr128j-JCX34-ttQ@mail.gmail.com>
Message-ID: <CALzn+-7fgxqbtvSw5ftQi-p_HkyWOx0D+Hwh10gP6UfonpDXdg@mail.gmail.com>

Hey,

You need to connect TDMCARD (Digium,Sangoma) any digital which is connect
to PRI modem, then you have to install Drivers for this card, load
mod_freetdm and follow the wiki.

Thanking you.

Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
C-91/92, 3rd Floor,Sector-2, Noida 201301
GSM +91.99993 45699 | Phone +91.12.0431.0581
Web http://www.buzzworks.com


On Tue, Aug 6, 2013 at 12:40 AM, DP Siddharth <
dp.siddharth at eng.knowlarity.com> wrote:

> Just check if linux kernel source available on your server? Wanpipe driver
> needs that for compilation.
>
> To have sangoma working you also need lbsng_isdn.so or libpri installed.
>
> Install wanpipe-> configure it through wancfg_fs-> install
> libsng_isdn.so-> compile and install mod_freetdm -> reload mod_freetdm in
> freeswitch.
>
> Thanks & Regards,
> Siddharth
> On Aug 5, 2013 11:00 PM, "Ravi S" <sravi123 at yahoo.com> wrote:
>
>>  Michael,
>>
>> Thank you for your response. The PRI is supplied through copper cables. A
>> modem receives it and through an RJ45 cable, I connect it to the Freeswitch
>> server. Per the service provider it is a PRI connection. I will check with
>> them to confirm that this the case.
>>
>> Thanks.
>> Ravi
>>
>> On 05/08/13 7:12 AM, Michael Collins wrote:
>>
>> Do they supply a "real" PRI or is it an IP connection with a device to
>> convert to PRI?
>> On Aug 4, 2013 4:43 PM, "Ken Rice" <krice at freeswitch.org> wrote:
>>
>>> you should review the instructions for setting up PRI on the fs wiki...
>>> a prindoes not have a username or a password
>>>
>>> Ken
>>> Sent from my iPad
>>>
>>> On Aug 4, 2013, at 15:52, Ravi <sravi123 at yahoo.com> wrote:
>>>
>>> > Hello Everyone !
>>> >
>>> > I am from India. I have recently taken a PRI connection from Bharti
>>> > Airtel, one of the service providers. I have installed the following:
>>> >
>>> > Cent OS
>>> > Freeswitch, FreeTDM
>>> > Sangoma Card
>>> >
>>> >
>>> > I think, I have followed all the instructions. I am struggling to
>>> > configure Freeswitch to start using the PRI connection. This is what I
>>> > have from the freeswitch cook book, to configure the gateway: we need
>>> > username/password, server address or IP and port.
>>> >
>>> > When I checked with Airtel, they are telling me that they only give
>>> > username/password and IP address details for an internet connection and
>>> > not for PRI lines.
>>> >
>>> > Has anyone here in the list, tried using an indian service provider and
>>> > configured in Freeswitch?
>>> > Has anyone done it for Airtel ?
>>> > Or please help me to figure out how to configure the PRI so as to make
>>> > inbound and outbound calls ??
>>> >
>>> > Any help is much appreciated.
>>> >
>>> > Thanks.
>>> > Ravi
>>> > +91-7502029000
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.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
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Tue Aug  6 08:31:49 2013
From: eidevm5 at gmail.com (Peter)
Date: Tue, 6 Aug 2013 14:31:49 +1000
Subject: [Freeswitch-users] TLS/Freeswitch self signed certs
In-Reply-To: <CAC0j6rfkNbJ7KMKMo7VVfGRG0uox3xZjqaw8heX2=HE09TcrEg@mail.gmail.com>
References: <CABkS=xockpPROOhnbSaYJxt0uw2bUtKbyYus_CWqEsWz=sjB6Q@mail.gmail.com>
	<CAC0j6rfkNbJ7KMKMo7VVfGRG0uox3xZjqaw8heX2=HE09TcrEg@mail.gmail.com>
Message-ID: <CABkS=xqg_Nw=X97szavNccZOT+cHA5PZ2sQ0-Tg=7DW+cqZTrg@mail.gmail.com>

The reason I put it on a webserver is mostly for convenience to make it
easier to install.

I tried copying cafile.pem to /sdcard on a Galaxy Note II, but when I try
the "Install from device storage" option, it just comes back with:

"No certificate file found on SD card"



On Mon, Aug 5, 2013 at 5:51 PM, Mehroz Ashraf <mehroz.ashraf85 at gmail.com>wrote:

> Why do you want to place the cert on webserver and point android browser?
> If you are doing this to download cert into android then that is probably
> not the right approach.
>
> I used cafile.pem (without converting it into .der format) and placed the
> file in  SD card or phone memory, and point out linphone to get the CA from
> the path. You may search in libraries where it need to tell the path.
>
>
> On Mon, Aug 5, 2013 at 12:15 PM, Peter <eidevm5 at gmail.com> wrote:
>
>> Has anyone managed to get TLS working between Android Linphone and
>> Freeswitch?
>>
>> I've done the basic TLS setup as per https://wiki.freeswitch.org/wiki/Tls
>>
>> I then convert the CA cert from PEM to DER format with:
>>
>> openssl x509  -inform PEM -outform der -in cafile.pem -out fs.crt
>>
>> I place fs.crt on a webserver and point my Android browser to it.
>>
>> When I click on fs.crt, I get the default Android Certificate installer
>> popup, but it always says:
>>
>> "Package contains: one user certificate"
>>
>> ie: it thinks it is a user cert rather than a CA cert.
>>
>> Android appears to be a real pain to add a CA to its trusted credential
>> store.
>>
>> Really interested if anyone has managed to get Android to import the CA
>> cert.
>>
>> Thanks
>>
>> Peter
>>
>> ____________________________________________________________
>>
>>
>>
>
> ____________________________________________________________
>
>
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From eidevm5 at gmail.com  Tue Aug  6 08:35:54 2013
From: eidevm5 at gmail.com (Peter)
Date: Tue, 6 Aug 2013 14:35:54 +1000
Subject: [Freeswitch-users] One way audio to CME
In-Reply-To: <89239AD2-AE02-4E1C-8692-6E77894D8384@gmail.com>
References: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>
	<89239AD2-AE02-4E1C-8692-6E77894D8384@gmail.com>
Message-ID: <CABkS=xo7NQoTNpeR0dMrLkeBndypfPmTbPEEbUonMu_NJ7wfAA@mail.gmail.com>

Thanks for replying Anthony.

Keep in mind that I have very little experience with Cisco products, so I
may be missing something fundamental.

As far as I can see

voice-class sip bind media source-interface ....

is just used to bind the SIP or media stream to the appropriate interface
on the CUBE.

My issue is that the CUBE is trying to initiate the return RTP stream to
the external interface (instead of the internal interface) on the
Freeswitch SBC.

Is my understanding of the sip bind media command correct?

Thanks

Peter


On Mon, Aug 5, 2013 at 5:23 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:

> On cube make sure you specify the source address on your dial-peers
> voice-class sip bind media|control
> to the correct side. I have seen one way audio when not set.
>
> On 5 Aug 2013, at 06:29, Peter <eidevm5 at gmail.com> wrote:
>
> >
> >
> > I currently have successful two way calls (signalling and media) in the
> following setup
> >
> >
> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  ->
> Kamailio --> Internal Linphone
> >
> > However, when I try to call a Cisco handset that is registered to CUCM9
> via CME in the following config:
> >
> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> CME
> ->  CUCM9 -->  Cisco handset
> >
> > The call signalling appears to be working fine and I can successfully
>  initiate a call from each end, but the only RTP stream that is working is
> from the external Linphone client to the Cisco handset.
> >
> > Note that CME is being used as a CUBE device, so all SIP and RTP goes
> via it.
> >
> > Looking at the RTP debugs on CME I can see the problem is that the
> "Media Dest Addr" is getting set to the external side of the FS SBC rather
> than the internal IP address.
> >
> >
> > I tried setting adding:
> >
> >             <action application="set"
> data="disable_rtp_auto_adjust="true" />
> >
> > to the dialplan on the SBC, but it made no difference.
> >
> >
> > Any suggestions as to what to check next?
> >
> > Thanks
> >
> > Peter
> >
> > _________________________________________________________________________
>
>
>
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From yehavi.bourvine at gmail.com  Tue Aug  6 09:02:10 2013
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 6 Aug 2013 08:02:10 +0300
Subject: [Freeswitch-users] A few problems with latest GIT
Message-ID: <CAAtWKeEW+ngi=7xq=0LOuYSPg4_JUwzFjqDBMMheskDy40bFvg@mail.gmail.com>

Hello,

  I am running FS version 1.0.x and trying to move to 1.2/1.4/latest-git. I
encounter a few problems which prevents me from using these versions, and
would like to know whether someone else is having it. It is quite hard to
create debugging information, as with full logging the behaviour is
slightly different than lower logging level. The problems appear only on my
production system, around noon (when the number of active sessins goes
beyond 50-60 sessions):


   - The voice path is lost in the middle of the session (Jira FS-5616).
   From TCPdump I see that FS receive the audio from both sides but stops
   sending it; there is nothing meaningfull in the logs.
   - FS starts to get sluggish, fs_cli is stuck, SIP messages are processed
   in delay (5-30 seconds). This goes away by itself after a few minutes.
   - sessions are left "in use" after the users hangup, and the number of
   active sessions starts rising. Doing uuid_kill succeeds once (next times
   fail with "no such uuid", but "show channels" still shows it.

The most frustrating side is that I do not know what exact debugging tools
to use in order to submit meaningfull data to FS developers.

                     Thanks! __Yehavi:
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From eidevm5 at gmail.com  Tue Aug  6 10:16:24 2013
From: eidevm5 at gmail.com (Peter)
Date: Tue, 6 Aug 2013 16:16:24 +1000
Subject: [Freeswitch-users] TLS/Freeswitch self signed certs
In-Reply-To: <CABkS=xqg_Nw=X97szavNccZOT+cHA5PZ2sQ0-Tg=7DW+cqZTrg@mail.gmail.com>
References: <CABkS=xockpPROOhnbSaYJxt0uw2bUtKbyYus_CWqEsWz=sjB6Q@mail.gmail.com>
	<CAC0j6rfkNbJ7KMKMo7VVfGRG0uox3xZjqaw8heX2=HE09TcrEg@mail.gmail.com>
	<CABkS=xqg_Nw=X97szavNccZOT+cHA5PZ2sQ0-Tg=7DW+cqZTrg@mail.gmail.com>
Message-ID: <CABkS=xqE=3D_PuSVPLWXM8gFzf_7+RPYjj4PTNAUUTU=F1iyog@mail.gmail.com>

Finally figured out the issue was related to the gentls_cert script was
generating an openssl template that didn't have the required x509v3
extensions set.

I modified the script where it generates config.tpl to add

                        x509_extensions = v3_ca

to the [req] section, then I added the section:

                        [ v3_ca ]
                        subjectKeyIdentifier=hash
                        authorityKeyIdentifier=keyid:always,issuer
                        basicConstraints=CA:TRUE

Now when you issue:

openssl x509 -noout -inform pem -text -in cafile.pem

you'll see the following section:

        X509v3 extensions:
            X509v3 Subject Key Identifier:
                02:0A:A8:D0:5C:23:7C:8B:C4:EF:79:11:C7:0C:A8:86:71:15:59:D5
            X509v3 Authority Key Identifier:

keyid:02:0A:A8:D0:5C:23:7C:8B:C4:EF:79:11:C7:0C:A8:86:71:15:59:D5

            X509v3 Basic Constraints:
                CA:TRUE

If these are present, then Android will treat the cert as a standard user
cert.

Then it was a simple matter of copying cafile.pem to cafile.crt on the
sdcard on the Android device and using the "install from device storage"
option.

When the cert installer dialog comes up, it will now detect cafile.crt as a
CA cert and not user cert.

Hope this helps other people, as cert management on Android is a right pain
in the $#%^.

Peter



On Tue, Aug 6, 2013 at 2:31 PM, Peter <eidevm5 at gmail.com> wrote:

> The reason I put it on a webserver is mostly for convenience to make it
> easier to install.
>
> I tried copying cafile.pem to /sdcard on a Galaxy Note II, but when I try
> the "Install from device storage" option, it just comes back with:
>
> "No certificate file found on SD card"
>
>
>
> On Mon, Aug 5, 2013 at 5:51 PM, Mehroz Ashraf <mehroz.ashraf85 at gmail.com>wrote:
>
>> Why do you want to place the cert on webserver and point android browser?
>> If you are doing this to download cert into android then that is probably
>> not the right approach.
>>
>> I used cafile.pem (without converting it into .der format) and placed the
>> file in  SD card or phone memory, and point out linphone to get the CA from
>> the path. You may search in libraries where it need to tell the path.
>>
>>
>> On Mon, Aug 5, 2013 at 12:15 PM, Peter <eidevm5 at gmail.com> wrote:
>>
>>> Has anyone managed to get TLS working between Android Linphone and
>>> Freeswitch?
>>>
>>> I've done the basic TLS setup as per
>>> https://wiki.freeswitch.org/wiki/Tls
>>>
>>> I then convert the CA cert from PEM to DER format with:
>>>
>>> openssl x509  -inform PEM -outform der -in cafile.pem -out fs.crt
>>>
>>> I place fs.crt on a webserver and point my Android browser to it.
>>>
>>> When I click on fs.crt, I get the default Android Certificate installer
>>> popup, but it always says:
>>>
>>> "Package contains: one user certificate"
>>>
>>> ie: it thinks it is a user cert rather than a CA cert.
>>>
>>> Android appears to be a real pain to add a CA to its trusted credential
>>> store.
>>>
>>> Really interested if anyone has managed to get Android to import the CA
>>> cert.
>>>
>>> Thanks
>>>
>>> Peter
>>>
>>> ____________________________________________________________
>>>
>>>
>>>
>>
>> ____________________________________________________________
>>
>>
>
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From zhulizhong at live.com  Tue Aug  6 10:42:14 2013
From: zhulizhong at live.com (James zhu)
Date: Tue, 6 Aug 2013 06:42:14 +0000
Subject: [Freeswitch-users] Can not make out going call from sangoma A200
In-Reply-To: <CAHgk4BVdu6Dk3P=rrRBKquiydLUUGBUxSHauTv73tx=XPq=pjA@mail.gmail.com>
References: <BLU171-W3672C8625546F122FFC013B5510@phx.gbl>,
	<CAHgk4BXX1nArMXv-T5CNzLbjh_wyn0fjzYNOUCaAPsUE14KNdA@mail.gmail.com>,
	<BLU171-W58FF5C141AECC333E5BD0FB5520@phx.gbl>,
	<CAHgk4BVdu6Dk3P=rrRBKquiydLUUGBUxSHauTv73tx=XPq=pjA@mail.gmail.com>
Message-ID: <BLU171-W488ED06FE9240BE8CD3149B55D0@phx.gbl>

hello:I contacted Sangoma, but I do not have any feedback yet. I think it might be a bugfor analog card with freeSWITCH.  freetdm  partially works with sangoma A200. There are two issues:1) Can not get callerid from IO:  only shows this:Initializing cid data! the callerid is emptyeven I set to CN.2) Can not make outgoing calls by the port that I used to make outgoing calls. even I replaced a FXO module, the problem is same. I use same syntax as follow:http://blog.hiastar.com/?p=276 Hope someone from the FreeSWITCH community to clarify that. 
Best regards,

James.zhu

website: www.hiastar.com

Date: Fri, 2 Aug 2013 23:03:47 -0400
From: bdfoster at davri.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can not make out going call from sangoma A200

I apologize, I misread your email. Hopefully someone who knows FreeTDM will come around. Have you contacted Sangoma? They are the maintainers of FreeTDM as far as I know. You might try that route. They are very familiar with freeswitch im sure :).

Thank you,
Brian Foster

Project Manager/Owner's Rep.

Davri Investments, Inc.

O: 317-787-2686 x2102

M: 317-600-9753

E: bdfoster at davri.com

Indianapolis, Indiana
Sent from a mobile device.
On Aug 2, 2013 10:39 PM, "James zhu" <zhulizhong at live.com> wrote:




thanks, Brian.  actually the 2:1 is idle because the port can accept incoming call. afterI reload freetdm or restart FS, I still can not make outgoing calls. If the port physicallyfailed, how come I can make incoming call use the same port. confused. thanks again.

Best regards,

James.zhu

website: www.hiastar.com

Date: Fri, 2 Aug 2013 14:22:29 -0400
From: bdfoster at davri.com
To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] Can not make out going call from sangoma A200

That's the expected behavior if 2:1 is in use. So that means you use another port to place your outbound call. Tips on how to do that, however, I can't really produce. Hopefully someone with more experience with FreeTDM can chime in.


Thank you,
Brian Foster

Project Manager/Owner's Rep.

Davri Investments, Inc.

O: 317-787-2686 x2102

M: 317-600-9753

E: bdfoster at davri.com

Indianapolis, Indiana
Sent from a mobile device.
On Aug 2, 2013 3:09 AM, "James zhu" <zhulizhong at live.com> wrote:




hello:I installed freeswitch and sangoma A200 with 2 FXOs. i set the tone group to CNand opermode=china. the A200 can make accept incoming calls from FreeTDM/2:1 , but I use same port to make out going port, the shows the port is CONGESTION.

--------------------------system log-----------------------------------------------2013-08-02 14:36:42.003531 [DEBUG] mod_freetdm.c:640 [2:1] FreeTDM/2:1/13811737925 CHANNEL HANGUP EXIT

2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:48 FreeTDM/2:1/13811737925 Standard HANGUP, cause: NORMAL_CIRCUIT_CONGESTION2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:679 (FreeTDM/2:1/13811737925) State HANGUP going to sleep

2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:692 (FreeTDM/2:1/13811737925) Callstate Change DOWN -> HANGUP2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:447 (FreeTDM/2:1/13811737925) State Change CS_HANGUP -> CS_REPORTING

2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925) Running State Change CS_REPORTING

2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State REPORTING2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:92 FreeTDM/2:1/13811737925 Standard REPORTING, cause: NORMAL_CIRCUIT_CONGESTION

2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State REPORTING going to sleep2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:441 (FreeTDM/2:1/13811737925) State Change CS_REPORTING -> CS_DESTROY

2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1549 Session 37 (FreeTDM/2:1/13811737925) Locked, Waiting on external entities

2013-08-02 14:36:42.023531 [DEBUG] switch_ivr_originate.c:3632 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]2013-08-02 14:36:42.023531 [INFO] mod_dptools.c:3192 Originate Failed.  Cause: NORMAL_CIRCUIT_CONGESTION

2013-08-02 14:36:42.023531 [NOTICE] switch_channel.c:4593 Hangup sofia/internal/1008 at 192.168.0.173 [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]2013-08-02 14:36:42.023531 [DEBUG] switch_channel.c:3135 Send signal sofia/internal/1008 at 192.168.0.173 [KILL]

2013-08-02 14:36:42.023531 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal---------------------------------------------the FS is download from git and wanpipe is latest version.  please give a help for that issue.


Best regards,

James.zhu

website: www.hiastar.com 		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.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




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.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




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 agtmcgarry at gmail.com  Tue Aug  6 11:13:29 2013
From: agtmcgarry at gmail.com (Anthony McGarry)
Date: Tue, 6 Aug 2013 08:13:29 +0100
Subject: [Freeswitch-users] One way audio to CME
In-Reply-To: <CABkS=xo7NQoTNpeR0dMrLkeBndypfPmTbPEEbUonMu_NJ7wfAA@mail.gmail.com>
References: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>
	<89239AD2-AE02-4E1C-8692-6E77894D8384@gmail.com>
	<CABkS=xo7NQoTNpeR0dMrLkeBndypfPmTbPEEbUonMu_NJ7wfAA@mail.gmail.com>
Message-ID: <9C8EC1A4-7DAC-4226-BDD6-F63A9AFD107F@gmail.com>

Hi Peter,

Because the calls are fine when using Kamailio I'm assuming your sip profiles are fine and you FS SBC config is fine. Are you using the same profiles?
Yes you are correct. Have you added the commands? Add them as a first step.
Send on a 'debug ccsip messages' 

Anthony 



On 6 Aug 2013, at 05:35, Peter <eidevm5 at gmail.com> wrote:

> Thanks for replying Anthony.
> 
> Keep in mind that I have very little experience with Cisco products, so I may be missing something fundamental.
> 
> As far as I can see
> 
> voice-class sip bind media source-interface ....
> 
> is just used to bind the SIP or media stream to the appropriate interface on the CUBE.
> 
> My issue is that the CUBE is trying to initiate the return RTP stream to the external interface (instead of the internal interface) on the Freeswitch SBC.
> 
> Is my understanding of the sip bind media command correct?
> 
> Thanks
> 
> Peter
> 
> 
> On Mon, Aug 5, 2013 at 5:23 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
>> On cube make sure you specify the source address on your dial-peers
>> voice-class sip bind media|control
>> to the correct side. I have seen one way audio when not set.
>> 
>> On 5 Aug 2013, at 06:29, Peter <eidevm5 at gmail.com> wrote:
>> 
>> >
>> >
>> > I currently have successful two way calls (signalling and media) in the following setup
>> >
>> >
>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> Kamailio --> Internal Linphone
>> >
>> > However, when I try to call a Cisco handset that is registered to CUCM9 via CME in the following config:
>> >
>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> CME ->  CUCM9 -->  Cisco handset
>> >
>> > The call signalling appears to be working fine and I can successfully  initiate a call from each end, but the only RTP stream that is working is from the external Linphone client to the Cisco handset.
>> >
>> > Note that CME is being used as a CUBE device, so all SIP and RTP goes via it.
>> >
>> > Looking at the RTP debugs on CME I can see the problem is that the "Media Dest Addr" is getting set to the external side of the FS SBC rather than the internal IP address.
>> >
>> >
>> > I tried setting adding:
>> >
>> >             <action application="set" data="disable_rtp_auto_adjust="true" />
>> >
>> > to the dialplan on the SBC, but it made no difference.
>> >
>> >
>> > Any suggestions as to what to check next?
>> >
>> > Thanks
>> >
>> > Peter
>> >
>> > _________________________________________________________________________
>> 
>> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Tue Aug  6 11:51:50 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Tue, 6 Aug 2013 13:21:50 +0530
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAMKwYLCtQV1wCVzPxwhNj2d23eK__uPDYPb98p0ZiJLRqx0GKQ@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<52001B43.9080901@freeswitch.org>
	<CAMKwYLCtQV1wCVzPxwhNj2d23eK__uPDYPb98p0ZiJLRqx0GKQ@mail.gmail.com>
Message-ID: <CAMvhjJt8v0XVnGqBs5pMgEyNR3tDV6GuMFR5Dsb1x6XMCMO4tw@mail.gmail.com>

Yes my freeswitch is running...
On Aug 6, 2013 3:41 AM, "Gustavo Salazar" <guga.salazar.loor at gmail.com>
wrote:

>
>
> Is Freeswitch running?
> I have seen a similar error when I tried to start the cli and Freeswitch
> was not running .
>
>
>
> 2013/8/5 Yossi Neiman <mishehu at freeswitch.org>
>
>>  Do you have mod_event_socket loaded?
>>
>> -Yossi
>>
>>
>> On 08/05/2013 04:00 PM, Ashish Mishra wrote:
>>
>> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i
>> have installed freeswitch) it gives me the following error :
>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Gustavo Salazar
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Tue Aug  6 11:53:16 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Tue, 6 Aug 2013 13:23:16 +0530
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAMvhjJt8v0XVnGqBs5pMgEyNR3tDV6GuMFR5Dsb1x6XMCMO4tw@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<52001B43.9080901@freeswitch.org>
	<CAMKwYLCtQV1wCVzPxwhNj2d23eK__uPDYPb98p0ZiJLRqx0GKQ@mail.gmail.com>
	<CAMvhjJt8v0XVnGqBs5pMgEyNR3tDV6GuMFR5Dsb1x6XMCMO4tw@mail.gmail.com>
Message-ID: <CAMvhjJtbZTd9_s37EOsem42JsOszYwc9ZEZ5V_k1=+j4CLguaQ@mail.gmail.com>

How to load mod_event_socket...???

Yes my freeswitch is running...
On Aug 6, 2013 3:41 AM, "Gustavo Salazar" <guga.salazar.loor at gmail.com>
wrote:

>
>
> Is Freeswitch running?
> I have seen a similar error when I tried to start the cli and Freeswitch
> was not running .
>
>
>
> 2013/8/5 Yossi Neiman <mishehu at freeswitch.org>
>
>>  Do you have mod_event_socket loaded?
>>
>> -Yossi
>>
>>
>> On 08/05/2013 04:00 PM, Ashish Mishra wrote:
>>
>> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i
>> have installed freeswitch) it gives me the following error :
>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Gustavo Salazar
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at davri.com  Tue Aug  6 11:54:01 2013
From: bdfoster at davri.com (Brian Foster)
Date: Tue, 6 Aug 2013 03:54:01 -0400
Subject: [Freeswitch-users] freeswitch: Syntax error: word unexpected
	(expecting ")")
In-Reply-To: <CAHgk4BVhPAfJ-rdAs2pc2mG-jz86N1BsicEZThq0Y-XCWqxAWA@mail.gmail.com>
References: <CAHgk4BXhFoSaEyXTopMNFsDOwd=5kxFPmq5+h=e=EPbatV-mFw@mail.gmail.com>
	<CAHgk4BVhPAfJ-rdAs2pc2mG-jz86N1BsicEZThq0Y-XCWqxAWA@mail.gmail.com>
Message-ID: <CAHgk4BXaY6gPjhuG-_JR65XUL3waG34V-9DpzY0+wcUTQVNF7A@mail.gmail.com>

Fresh build on a bare metal Dell Precision 490 running Ubuntu 12.04 64-bit,
latest v1.2.stable (5 min ago). When I try to run freeswitch, I get:

freeswitch:1 freeswitch: Syntax error: word unexpected (expecting ")")

I've since deleted everything in /usr/local/freeswitch and
/usr/local/src/freeswitch, and ran the install again. Still have the same
outcome.

The same version was built on another server, no issues. What could be the
problem? Anything I can do to mitigate the issue?

Thank you,

Brian Foster
Project Manager/Owner's Representative
Davri Investments, Incorporated
P: +1-317-787-2686
M: +1-317-600-9753
Indianapolis, Indiana
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From ssinyagin at yahoo.com  Tue Aug  6 12:02:45 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Tue, 6 Aug 2013 01:02:45 -0700 (PDT)
Subject: [Freeswitch-users] IVR - Dial by extension
In-Reply-To: <CAHgk4BXkW2e94Hm3Tx46FpSzWRTGS15vV4pQ3qKv96TogWze6A@mail.gmail.com>
References: <CAHgk4BXkW2e94Hm3Tx46FpSzWRTGS15vV4pQ3qKv96TogWze6A@mail.gmail.com>
Message-ID: <1375776165.13386.YahooMailNeo@web126201.mail.ne1.yahoo.com>

this seems to be a perfect job for a Lua or javascript script. You can program the whole IVR menu in it.






>________________________________
> From: Brian Foster <bdfoster at davri.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Tuesday, August 6, 2013 4:26 AM
>Subject: [Freeswitch-users] IVR - Dial by extension
> 
>
>
>In the dialplan, I use something like this as a condition for the Local Extension "extension":
>
>
><condition field="${user_exists id ${destination_number} $${domain}}" expression="^true$">
>
>
>
>I was wondering if there was something you could do to have a similar behavior when creating IVR menus. Has anyone tried this? Our extensions don't fit a particular format, other than 4 digits, so checking beforehand whether the user id exists would be ideal before transferring.
>
>
>Thank you,
>
>
>Brian Foster
>Project Manager/Owner's Representative
>Davri Investments, Incorporated
>P: +1-317-787-2686
>M: +1-317-600-9753
>Indianapolis, Indiana
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 lists at kavun.ch  Tue Aug  6 12:37:18 2013
From: lists at kavun.ch (Emrah)
Date: Tue, 6 Aug 2013 04:37:18 -0400
Subject: [Freeswitch-users] A few problems with latest GIT
In-Reply-To: <CAAtWKeEW+ngi=7xq=0LOuYSPg4_JUwzFjqDBMMheskDy40bFvg@mail.gmail.com>
References: <CAAtWKeEW+ngi=7xq=0LOuYSPg4_JUwzFjqDBMMheskDy40bFvg@mail.gmail.com>
Message-ID: <BE4C37BE-3FDC-4E12-8718-CD8607A8AC81@kavun.ch>

I confirm the last issue of your list has also happened on my latest GIT.
Your no audio syndrome sounds like it could be connected to my Polycom bridging issue.
As for the sluggishnness of fs_cli I haven't noticed anything or just associated it to my mobile broadband? 
On Aug 6, 2013, at 1:02 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com> wrote:

> Hello,
>  
>   I am running FS version 1.0.x and trying to move to 1.2/1.4/latest-git. I encounter a few problems which prevents me from using these versions, and would like to know whether someone else is having it. It is quite hard to create debugging information, as with full logging the behaviour is slightly different than lower logging level. The problems appear only on my production system, around noon (when the number of active sessins goes beyond 50-60 sessions):
>  
> The voice path is lost in the middle of the session (Jira FS-5616). From TCPdump I see that FS receive the audio from both sides but stops sending it; there is nothing meaningfull in the logs.
> FS starts to get sluggish, fs_cli is stuck, SIP messages are processed in delay (5-30 seconds). This goes away by itself after a few minutes.
> sessions are left "in use" after the users hangup, and the number of active sessions starts rising. Doing uuid_kill succeeds once (next times fail with "no such uuid", but "show channels" still shows it.
> The most frustrating side is that I do not know what exact debugging tools to use in order to submit meaningfull data to FS developers.
>  
>                      Thanks! __Yehavi:
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 govoiper at gmail.com  Tue Aug  6 13:24:59 2013
From: govoiper at gmail.com (SamyGo)
Date: Tue, 6 Aug 2013 02:24:59 -0700
Subject: [Freeswitch-users] Mod_distributor question
Message-ID: <CAJUJwthrNBmEv3FMssDxN6hQjx5iUUQ+zsSZQMEH11tin=T-Zg@mail.gmail.com>

Dear users,

I'm trying to figure out how this mod_distributor works. I've a few
questions to help me get this going.

I've defined two of my gateways in the distributor.conf.xml like this:

<configuration name="distributor.conf" description="Distributor
Configuration">
  <lists>
    <list name="GWLB" total-weight="2">
      <node name="OpenSIPS-A" weight="1"/>
      <node name="OpenSIPS-B" weight="1"/>
    </list>
  </lists>
</configuration>

and so when I reload the distributor from API I get the following warning:

freeswitch at internal> distributor_ctl reload
+ok reloaded.
2013-08-06 14:17:16.652087 [WARNING] mod_distributor.c:201 Total weight
does not add up to total weight 10

So I've changed the total-weight to 10, and set weight of each node as 5.
That gives the same warning as well.

A Warning is fine, but as soon as I just use API to expand the list I get a
proper gateway name for the first two times only and then -err starts
showing up until I reload the distributor.

freeswitch at internal> expand distributor GWLB
OpenSIPS-A
freeswitch at internal> expand distributor GWLB
OpenSIPS-B
freeswitch at internal> expand distributor GWLB
-err
freeswitch at internal> expand distributor GWLB
-err
freeswitch at internal>

The last question is regarding use of this distributor from LUA script.

I call in the distributor like this from my LUA.

session =
freeswitch.Session("{ignore_early_media=true,origination_caller_id_number="..outbound_cli.."}sofia/gateway/${distributor(GWLB)}/"..
phone_number);

and that gives a straight error:

[ERR] mod_sofia.c:4668 Invalid Gateway '%{distributor(GWLB)}'


I hope somebody can share their knowledge on this module and help me move
forward with my project.

Best Regards,
Sammy
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From ahmed at netelsat.net  Tue Aug  6 13:31:46 2013
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Tue, 6 Aug 2013 14:31:46 +0500
Subject: [Freeswitch-users] Mod_distributor question
In-Reply-To: <CAJUJwthrNBmEv3FMssDxN6hQjx5iUUQ+zsSZQMEH11tin=T-Zg@mail.gmail.com>
References: <CAJUJwthrNBmEv3FMssDxN6hQjx5iUUQ+zsSZQMEH11tin=T-Zg@mail.gmail.com>
Message-ID: <CALFFxt0d0fv7kEHKBJTfYu2x+EDDye_M91vxtQEi30BpiTKCHA@mail.gmail.com>

Hi,
Total Weight 2 and nodes weight as 1/1 was fine . you should do first
"reloadxml" and then "distributor_ctl reload"

On Tue, Aug 6, 2013 at 2:24 PM, SamyGo <govoiper at gmail.com> wrote:

> Dear users,
>
> I'm trying to figure out how this mod_distributor works. I've a few
> questions to help me get this going.
>
> I've defined two of my gateways in the distributor.conf.xml like this:
>
> <configuration name="distributor.conf" description="Distributor
> Configuration">
>   <lists>
>     <list name="GWLB" total-weight="2">
>       <node name="OpenSIPS-A" weight="1"/>
>       <node name="OpenSIPS-B" weight="1"/>
>     </list>
>   </lists>
> </configuration>
>
> and so when I reload the distributor from API I get the following warning:
>
> freeswitch at internal> distributor_ctl reload
> +ok reloaded.
> 2013-08-06 14:17:16.652087 [WARNING] mod_distributor.c:201 Total weight
> does not add up to total weight 10
>
> So I've changed the total-weight to 10, and set weight of each node as 5.
> That gives the same warning as well.
>
> A Warning is fine, but as soon as I just use API to expand the list I get
> a proper gateway name for the first two times only and then -err starts
> showing up until I reload the distributor.
>
> freeswitch at internal> expand distributor GWLB
> OpenSIPS-A
> freeswitch at internal> expand distributor GWLB
> OpenSIPS-B
> freeswitch at internal> expand distributor GWLB
> -err
> freeswitch at internal> expand distributor GWLB
> -err
> freeswitch at internal>
>
> The last question is regarding use of this distributor from LUA script.
>
> I call in the distributor like this from my LUA.
>
> session =
> freeswitch.Session("{ignore_early_media=true,origination_caller_id_number="..outbound_cli.."}sofia/gateway/${distributor(GWLB)}/"..
> phone_number);
>
> and that gives a straight error:
>
> [ERR] mod_sofia.c:4668 Invalid Gateway '%{distributor(GWLB)}'
>
>
> I hope somebody can share their knowledge on this module and help me move
> forward with my project.
>
> Best Regards,
> Sammy
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 govoiper at gmail.com  Tue Aug  6 13:42:37 2013
From: govoiper at gmail.com (SamyGo)
Date: Tue, 6 Aug 2013 02:42:37 -0700
Subject: [Freeswitch-users] Mod_distributor question
In-Reply-To: <CALFFxt0d0fv7kEHKBJTfYu2x+EDDye_M91vxtQEi30BpiTKCHA@mail.gmail.com>
References: <CAJUJwthrNBmEv3FMssDxN6hQjx5iUUQ+zsSZQMEH11tin=T-Zg@mail.gmail.com>
	<CALFFxt0d0fv7kEHKBJTfYu2x+EDDye_M91vxtQEi30BpiTKCHA@mail.gmail.com>
Message-ID: <CAJUJwtj3Qh+vD72mZmhrCjORTEQ7YA+aLeZTJjUhQi0Z-=H-AQ@mail.gmail.com>

Dear Saboor,

Thanks for the quick reply, to share my luck with this, I'v even tried a
complete restart of freeswitch after trying reloadxml and sofia module
reloading as well.

The total-weight Warning even if I configured it with 2 made me think that
I'm probably changing in the wrong file but the API command to expand
distributor list gives results for the first two attempts so that is really
weird for me.

Just to add info, this is my FS version:

freeswitch at internal> version
FreeSWITCH Version 1.3.14b+git~20130305T150702Z~57d6495248 (git 57d6495
2013-03-05 15:07:02Z)

BR
Sammy




On Tue, Aug 6, 2013 at 2:31 AM, Ahmed Sboor <ahmed at netelsat.net> wrote:

> Hi,
> Total Weight 2 and nodes weight as 1/1 was fine . you should do first
> "reloadxml" and then "distributor_ctl reload"
>
> On Tue, Aug 6, 2013 at 2:24 PM, SamyGo <govoiper at gmail.com> wrote:
>
>> Dear users,
>>
>> I'm trying to figure out how this mod_distributor works. I've a few
>> questions to help me get this going.
>>
>> I've defined two of my gateways in the distributor.conf.xml like this:
>>
>> <configuration name="distributor.conf" description="Distributor
>> Configuration">
>>   <lists>
>>     <list name="GWLB" total-weight="2">
>>       <node name="OpenSIPS-A" weight="1"/>
>>       <node name="OpenSIPS-B" weight="1"/>
>>     </list>
>>   </lists>
>> </configuration>
>>
>> and so when I reload the distributor from API I get the following warning:
>>
>> freeswitch at internal> distributor_ctl reload
>> +ok reloaded.
>> 2013-08-06 14:17:16.652087 [WARNING] mod_distributor.c:201 Total weight
>> does not add up to total weight 10
>>
>> So I've changed the total-weight to 10, and set weight of each node as 5.
>> That gives the same warning as well.
>>
>> A Warning is fine, but as soon as I just use API to expand the list I get
>> a proper gateway name for the first two times only and then -err starts
>> showing up until I reload the distributor.
>>
>> freeswitch at internal> expand distributor GWLB
>> OpenSIPS-A
>> freeswitch at internal> expand distributor GWLB
>> OpenSIPS-B
>> freeswitch at internal> expand distributor GWLB
>> -err
>> freeswitch at internal> expand distributor GWLB
>> -err
>> freeswitch at internal>
>>
>> The last question is regarding use of this distributor from LUA script.
>>
>> I call in the distributor like this from my LUA.
>>
>> session =
>> freeswitch.Session("{ignore_early_media=true,origination_caller_id_number="..outbound_cli.."}sofia/gateway/${distributor(GWLB)}/"..
>> phone_number);
>>
>> and that gives a straight error:
>>
>> [ERR] mod_sofia.c:4668 Invalid Gateway '%{distributor(GWLB)}'
>>
>>
>> I hope somebody can share their knowledge on this module and help me move
>> forward with my project.
>>
>> Best Regards,
>> Sammy
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ahmed at netelsat.net  Tue Aug  6 14:22:46 2013
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Tue, 6 Aug 2013 15:22:46 +0500
Subject: [Freeswitch-users] Mod_distributor question
In-Reply-To: <CAJUJwtj3Qh+vD72mZmhrCjORTEQ7YA+aLeZTJjUhQi0Z-=H-AQ@mail.gmail.com>
References: <CAJUJwthrNBmEv3FMssDxN6hQjx5iUUQ+zsSZQMEH11tin=T-Zg@mail.gmail.com>
	<CALFFxt0d0fv7kEHKBJTfYu2x+EDDye_M91vxtQEi30BpiTKCHA@mail.gmail.com>
	<CAJUJwtj3Qh+vD72mZmhrCjORTEQ7YA+aLeZTJjUhQi0Z-=H-AQ@mail.gmail.com>
Message-ID: <CALFFxt2_CAoJo9W27hOycpDQsHMmBpxv75qu_a-eMwh0JwcN5g@mail.gmail.com>

Hi Sammy,
its a news for me that even after restarting you are having invalid weight
warnings.
there is only one file to change i-e

/usr/local/freeswitch/conf/autoload_configs/distributor.xml

And reloadxml,distributor_ctl reload should work fine.

can  you recheck file if sample list is still enabled as default configs
come with something like 10 i think.


On Tue, Aug 6, 2013 at 2:42 PM, SamyGo <govoiper at gmail.com> wrote:

> Dear Saboor,
>
> Thanks for the quick reply, to share my luck with this, I'v even tried a
> complete restart of freeswitch after trying reloadxml and sofia module
> reloading as well.
>
> The total-weight Warning even if I configured it with 2 made me think that
> I'm probably changing in the wrong file but the API command to expand
> distributor list gives results for the first two attempts so that is really
> weird for me.
>
> Just to add info, this is my FS version:
>
> freeswitch at internal> version
> FreeSWITCH Version 1.3.14b+git~20130305T150702Z~57d6495248 (git 57d6495
> 2013-03-05 15:07:02Z)
>
> BR
> Sammy
>
>
>
>
> On Tue, Aug 6, 2013 at 2:31 AM, Ahmed Sboor <ahmed at netelsat.net> wrote:
>
>> Hi,
>> Total Weight 2 and nodes weight as 1/1 was fine . you should do first
>> "reloadxml" and then "distributor_ctl reload"
>>
>> On Tue, Aug 6, 2013 at 2:24 PM, SamyGo <govoiper at gmail.com> wrote:
>>
>>> Dear users,
>>>
>>> I'm trying to figure out how this mod_distributor works. I've a few
>>> questions to help me get this going.
>>>
>>> I've defined two of my gateways in the distributor.conf.xml like this:
>>>
>>> <configuration name="distributor.conf" description="Distributor
>>> Configuration">
>>>   <lists>
>>>     <list name="GWLB" total-weight="2">
>>>       <node name="OpenSIPS-A" weight="1"/>
>>>       <node name="OpenSIPS-B" weight="1"/>
>>>     </list>
>>>   </lists>
>>> </configuration>
>>>
>>> and so when I reload the distributor from API I get the following
>>> warning:
>>>
>>> freeswitch at internal> distributor_ctl reload
>>> +ok reloaded.
>>> 2013-08-06 14:17:16.652087 [WARNING] mod_distributor.c:201 Total weight
>>> does not add up to total weight 10
>>>
>>> So I've changed the total-weight to 10, and set weight of each node as
>>> 5. That gives the same warning as well.
>>>
>>> A Warning is fine, but as soon as I just use API to expand the list I
>>> get a proper gateway name for the first two times only and then -err starts
>>> showing up until I reload the distributor.
>>>
>>> freeswitch at internal> expand distributor GWLB
>>> OpenSIPS-A
>>> freeswitch at internal> expand distributor GWLB
>>> OpenSIPS-B
>>> freeswitch at internal> expand distributor GWLB
>>> -err
>>> freeswitch at internal> expand distributor GWLB
>>> -err
>>> freeswitch at internal>
>>>
>>> The last question is regarding use of this distributor from LUA script.
>>>
>>> I call in the distributor like this from my LUA.
>>>
>>> session =
>>> freeswitch.Session("{ignore_early_media=true,origination_caller_id_number="..outbound_cli.."}sofia/gateway/${distributor(GWLB)}/"..
>>> phone_number);
>>>
>>> and that gives a straight error:
>>>
>>> [ERR] mod_sofia.c:4668 Invalid Gateway '%{distributor(GWLB)}'
>>>
>>>
>>> I hope somebody can share their knowledge on this module and help me
>>> move forward with my project.
>>>
>>> Best Regards,
>>> Sammy
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 govoiper at gmail.com  Tue Aug  6 14:41:07 2013
From: govoiper at gmail.com (SamyGo)
Date: Tue, 6 Aug 2013 03:41:07 -0700
Subject: [Freeswitch-users] Mod_distributor question
In-Reply-To: <CALFFxt2_CAoJo9W27hOycpDQsHMmBpxv75qu_a-eMwh0JwcN5g@mail.gmail.com>
References: <CAJUJwthrNBmEv3FMssDxN6hQjx5iUUQ+zsSZQMEH11tin=T-Zg@mail.gmail.com>
	<CALFFxt0d0fv7kEHKBJTfYu2x+EDDye_M91vxtQEi30BpiTKCHA@mail.gmail.com>
	<CAJUJwtj3Qh+vD72mZmhrCjORTEQ7YA+aLeZTJjUhQi0Z-=H-AQ@mail.gmail.com>
	<CALFFxt2_CAoJo9W27hOycpDQsHMmBpxv75qu_a-eMwh0JwcN5g@mail.gmail.com>
Message-ID: <CAJUJwtjEH9dCYoDogNciqeVF9PXsquiMSC9ThMbD4n+jg3oMZA@mail.gmail.com>

Dear Saboor,

That is indeed weird since I did not want to load the test list in the file
and hence used my own list. Reading your comment I put the test list back
in and my first two issues are resolved. No warning and no -err on
expanding the lists now. Its working now; strange for me.

Anyway, the last question still needs some help which is using this in LUA
script.

Best Regards,
Sammy



On Tue, Aug 6, 2013 at 3:22 AM, Ahmed Sboor <ahmed at netelsat.net> wrote:

> Hi Sammy,
> its a news for me that even after restarting you are having invalid weight
> warnings.
> there is only one file to change i-e
>
> /usr/local/freeswitch/conf/autoload_configs/distributor.xml
>
> And reloadxml,distributor_ctl reload should work fine.
>
> can  you recheck file if sample list is still enabled as default configs
> come with something like 10 i think.
>
>
> On Tue, Aug 6, 2013 at 2:42 PM, SamyGo <govoiper at gmail.com> wrote:
>
>> Dear Saboor,
>>
>> Thanks for the quick reply, to share my luck with this, I'v even tried a
>> complete restart of freeswitch after trying reloadxml and sofia module
>> reloading as well.
>>
>> The total-weight Warning even if I configured it with 2 made me think
>> that I'm probably changing in the wrong file but the API command to expand
>> distributor list gives results for the first two attempts so that is really
>> weird for me.
>>
>> Just to add info, this is my FS version:
>>
>> freeswitch at internal> version
>> FreeSWITCH Version 1.3.14b+git~20130305T150702Z~57d6495248 (git 57d6495
>> 2013-03-05 15:07:02Z)
>>
>> BR
>> Sammy
>>
>>
>>
>>
>> On Tue, Aug 6, 2013 at 2:31 AM, Ahmed Sboor <ahmed at netelsat.net> wrote:
>>
>>> Hi,
>>> Total Weight 2 and nodes weight as 1/1 was fine . you should do first
>>> "reloadxml" and then "distributor_ctl reload"
>>>
>>> On Tue, Aug 6, 2013 at 2:24 PM, SamyGo <govoiper at gmail.com> wrote:
>>>
>>>> Dear users,
>>>>
>>>> I'm trying to figure out how this mod_distributor works. I've a few
>>>> questions to help me get this going.
>>>>
>>>> I've defined two of my gateways in the distributor.conf.xml like this:
>>>>
>>>> <configuration name="distributor.conf" description="Distributor
>>>> Configuration">
>>>>   <lists>
>>>>     <list name="GWLB" total-weight="2">
>>>>       <node name="OpenSIPS-A" weight="1"/>
>>>>       <node name="OpenSIPS-B" weight="1"/>
>>>>     </list>
>>>>   </lists>
>>>> </configuration>
>>>>
>>>> and so when I reload the distributor from API I get the following
>>>> warning:
>>>>
>>>> freeswitch at internal> distributor_ctl reload
>>>> +ok reloaded.
>>>> 2013-08-06 14:17:16.652087 [WARNING] mod_distributor.c:201 Total weight
>>>> does not add up to total weight 10
>>>>
>>>> So I've changed the total-weight to 10, and set weight of each node as
>>>> 5. That gives the same warning as well.
>>>>
>>>> A Warning is fine, but as soon as I just use API to expand the list I
>>>> get a proper gateway name for the first two times only and then -err starts
>>>> showing up until I reload the distributor.
>>>>
>>>> freeswitch at internal> expand distributor GWLB
>>>> OpenSIPS-A
>>>> freeswitch at internal> expand distributor GWLB
>>>> OpenSIPS-B
>>>> freeswitch at internal> expand distributor GWLB
>>>> -err
>>>> freeswitch at internal> expand distributor GWLB
>>>> -err
>>>> freeswitch at internal>
>>>>
>>>> The last question is regarding use of this distributor from LUA script.
>>>>
>>>> I call in the distributor like this from my LUA.
>>>>
>>>> session =
>>>> freeswitch.Session("{ignore_early_media=true,origination_caller_id_number="..outbound_cli.."}sofia/gateway/${distributor(GWLB)}/"..
>>>> phone_number);
>>>>
>>>> and that gives a straight error:
>>>>
>>>> [ERR] mod_sofia.c:4668 Invalid Gateway '%{distributor(GWLB)}'
>>>>
>>>>
>>>> I hope somebody can share their knowledge on this module and help me
>>>> move forward with my project.
>>>>
>>>> Best Regards,
>>>> Sammy
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 adahary at gmail.com  Tue Aug  6 15:21:26 2013
From: adahary at gmail.com (adahary)
Date: Tue, 6 Aug 2013 04:21:26 -0700 (PDT)
Subject: [Freeswitch-users] fs_ivrd 'Could not open input file'
Message-ID: <1375788086795-7593663.post@n2.nabble.com>

I'm trying to run an outbound php script using fs_ivrd but keep getting
'Could not open input file' whenever the dialplan redirects to the socket.

I'm trying to run the basic example (as it is) from this wiki:
http://wiki.freeswitch.org/wiki/PHP_ESL#ivrd

$ /usr/local/freeswitch/bin/fs_ivrd -h 127.0.0.1 -p 8084
Starting forking listener.
Could not open input file: 4      <== response to the dialplan

I've already set up permission 755 to the php script:
-rwxr-xr-x 1 root root 678 Aug  6 06:58
/usr/local/freeswitch/scripts/ivrd-demo.php

I've also disabled iptables.

Please advice

Assaf



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/fs-ivrd-Could-not-open-input-file-tp7593663.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From ahmed at netelsat.net  Tue Aug  6 15:31:12 2013
From: ahmed at netelsat.net (Ahmed Sboor)
Date: Tue, 6 Aug 2013 16:31:12 +0500
Subject: [Freeswitch-users] Mod_distributor question
In-Reply-To: <CAJUJwtjEH9dCYoDogNciqeVF9PXsquiMSC9ThMbD4n+jg3oMZA@mail.gmail.com>
References: <CAJUJwthrNBmEv3FMssDxN6hQjx5iUUQ+zsSZQMEH11tin=T-Zg@mail.gmail.com>
	<CALFFxt0d0fv7kEHKBJTfYu2x+EDDye_M91vxtQEi30BpiTKCHA@mail.gmail.com>
	<CAJUJwtj3Qh+vD72mZmhrCjORTEQ7YA+aLeZTJjUhQi0Z-=H-AQ@mail.gmail.com>
	<CALFFxt2_CAoJo9W27hOycpDQsHMmBpxv75qu_a-eMwh0JwcN5g@mail.gmail.com>
	<CAJUJwtjEH9dCYoDogNciqeVF9PXsquiMSC9ThMbD4n+jg3oMZA@mail.gmail.com>
Message-ID: <CALFFxt1aqodqi1Ja8ZG1pTcM9XqxOSLf2c+qi-Mbyjroo6vV4g@mail.gmail.com>

Hi
i am not sure about lua so may be some other friend can help . but one
thing i can see that you are trying to call a freeswitch variable inside a
lua string so may  be its the reason its not doing as you are expecting.
If you share your lua code may be someone here can help .

Thanks
Ahmed


On Tue, Aug 6, 2013 at 3:41 PM, SamyGo <govoiper at gmail.com> wrote:

> Dear Saboor,
>
> That is indeed weird since I did not want to load the test list in the
> file and hence used my own list. Reading your comment I put the test list
> back in and my first two issues are resolved. No warning and no -err on
> expanding the lists now. Its working now; strange for me.
>
> Anyway, the last question still needs some help which is using this in LUA
> script.
>
> Best Regards,
> Sammy
>
>
>
> On Tue, Aug 6, 2013 at 3:22 AM, Ahmed Sboor <ahmed at netelsat.net> wrote:
>
>> Hi Sammy,
>> its a news for me that even after restarting you are having invalid
>> weight warnings.
>> there is only one file to change i-e
>>
>> /usr/local/freeswitch/conf/autoload_configs/distributor.xml
>>
>> And reloadxml,distributor_ctl reload should work fine.
>>
>> can  you recheck file if sample list is still enabled as default configs
>> come with something like 10 i think.
>>
>>
>> On Tue, Aug 6, 2013 at 2:42 PM, SamyGo <govoiper at gmail.com> wrote:
>>
>>> Dear Saboor,
>>>
>>> Thanks for the quick reply, to share my luck with this, I'v even tried a
>>> complete restart of freeswitch after trying reloadxml and sofia module
>>> reloading as well.
>>>
>>> The total-weight Warning even if I configured it with 2 made me think
>>> that I'm probably changing in the wrong file but the API command to expand
>>> distributor list gives results for the first two attempts so that is really
>>> weird for me.
>>>
>>> Just to add info, this is my FS version:
>>>
>>> freeswitch at internal> version
>>> FreeSWITCH Version 1.3.14b+git~20130305T150702Z~57d6495248 (git 57d6495
>>> 2013-03-05 15:07:02Z)
>>>
>>> BR
>>> Sammy
>>>
>>>
>>>
>>>
>>> On Tue, Aug 6, 2013 at 2:31 AM, Ahmed Sboor <ahmed at netelsat.net> wrote:
>>>
>>>> Hi,
>>>> Total Weight 2 and nodes weight as 1/1 was fine . you should do first
>>>> "reloadxml" and then "distributor_ctl reload"
>>>>
>>>> On Tue, Aug 6, 2013 at 2:24 PM, SamyGo <govoiper at gmail.com> wrote:
>>>>
>>>>> Dear users,
>>>>>
>>>>> I'm trying to figure out how this mod_distributor works. I've a few
>>>>> questions to help me get this going.
>>>>>
>>>>> I've defined two of my gateways in the distributor.conf.xml like this:
>>>>>
>>>>> <configuration name="distributor.conf" description="Distributor
>>>>> Configuration">
>>>>>   <lists>
>>>>>     <list name="GWLB" total-weight="2">
>>>>>       <node name="OpenSIPS-A" weight="1"/>
>>>>>       <node name="OpenSIPS-B" weight="1"/>
>>>>>     </list>
>>>>>   </lists>
>>>>> </configuration>
>>>>>
>>>>> and so when I reload the distributor from API I get the following
>>>>> warning:
>>>>>
>>>>> freeswitch at internal> distributor_ctl reload
>>>>> +ok reloaded.
>>>>> 2013-08-06 14:17:16.652087 [WARNING] mod_distributor.c:201 Total
>>>>> weight does not add up to total weight 10
>>>>>
>>>>> So I've changed the total-weight to 10, and set weight of each node as
>>>>> 5. That gives the same warning as well.
>>>>>
>>>>> A Warning is fine, but as soon as I just use API to expand the list I
>>>>> get a proper gateway name for the first two times only and then -err starts
>>>>> showing up until I reload the distributor.
>>>>>
>>>>> freeswitch at internal> expand distributor GWLB
>>>>> OpenSIPS-A
>>>>> freeswitch at internal> expand distributor GWLB
>>>>> OpenSIPS-B
>>>>> freeswitch at internal> expand distributor GWLB
>>>>> -err
>>>>> freeswitch at internal> expand distributor GWLB
>>>>> -err
>>>>> freeswitch at internal>
>>>>>
>>>>> The last question is regarding use of this distributor from LUA script.
>>>>>
>>>>> I call in the distributor like this from my LUA.
>>>>>
>>>>> session =
>>>>> freeswitch.Session("{ignore_early_media=true,origination_caller_id_number="..outbound_cli.."}sofia/gateway/${distributor(GWLB)}/"..
>>>>> phone_number);
>>>>>
>>>>> and that gives a straight error:
>>>>>
>>>>> [ERR] mod_sofia.c:4668 Invalid Gateway '%{distributor(GWLB)}'
>>>>>
>>>>>
>>>>> I hope somebody can share their knowledge on this module and help me
>>>>> move forward with my project.
>>>>>
>>>>> Best Regards,
>>>>> Sammy
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 govoiper at gmail.com  Tue Aug  6 15:52:54 2013
From: govoiper at gmail.com (SamyGo)
Date: Tue, 6 Aug 2013 04:52:54 -0700
Subject: [Freeswitch-users] Mod_distributor question
In-Reply-To: <CALFFxt1aqodqi1Ja8ZG1pTcM9XqxOSLf2c+qi-Mbyjroo6vV4g@mail.gmail.com>
References: <CAJUJwthrNBmEv3FMssDxN6hQjx5iUUQ+zsSZQMEH11tin=T-Zg@mail.gmail.com>
	<CALFFxt0d0fv7kEHKBJTfYu2x+EDDye_M91vxtQEi30BpiTKCHA@mail.gmail.com>
	<CAJUJwtj3Qh+vD72mZmhrCjORTEQ7YA+aLeZTJjUhQi0Z-=H-AQ@mail.gmail.com>
	<CALFFxt2_CAoJo9W27hOycpDQsHMmBpxv75qu_a-eMwh0JwcN5g@mail.gmail.com>
	<CAJUJwtjEH9dCYoDogNciqeVF9PXsquiMSC9ThMbD4n+jg3oMZA@mail.gmail.com>
	<CALFFxt1aqodqi1Ja8ZG1pTcM9XqxOSLf2c+qi-Mbyjroo6vV4g@mail.gmail.com>
Message-ID: <CAJUJwtgbjHk7vM9xCvL11jvtVux_Y6mFCc=X8rk5wf=x-8G=Kg@mail.gmail.com>

Thank you Saboor,

I'm not sure I understood what you meant by calling FS variable inside LUA.
I call my LUA script from luarun BGAPI command over ESL and hence no XML
dialplan loads this script.

I have made this work alternatively by calling "eval ${distributor(GWLB)}"
command from the LUA script using freeswitch.API().

These two lines got me the required gateway:

fs_api = freeswitch.API();
gateway = fs_api:executeString("eval ${distributor(GWLB)}");

I am feeling this as an overhead since its an Auto-dialer LUA application
and is called about couple hundred times a second.

If anyone can advise that'll be great.

Best Regards,
Sammy



On Tue, Aug 6, 2013 at 4:31 AM, Ahmed Sboor <ahmed at netelsat.net> wrote:

> Hi
> i am not sure about lua so may be some other friend can help . but one
> thing i can see that you are trying to call a freeswitch variable inside a
> lua string so may  be its the reason its not doing as you are expecting.
> If you share your lua code may be someone here can help .
>
> Thanks
> Ahmed
>
>
> On Tue, Aug 6, 2013 at 3:41 PM, SamyGo <govoiper at gmail.com> wrote:
>
>> Dear Saboor,
>>
>> That is indeed weird since I did not want to load the test list in the
>> file and hence used my own list. Reading your comment I put the test list
>> back in and my first two issues are resolved. No warning and no -err on
>> expanding the lists now. Its working now; strange for me.
>>
>> Anyway, the last question still needs some help which is using this in
>> LUA script.
>>
>> Best Regards,
>> Sammy
>>
>>
>>
>> On Tue, Aug 6, 2013 at 3:22 AM, Ahmed Sboor <ahmed at netelsat.net> wrote:
>>
>>> Hi Sammy,
>>> its a news for me that even after restarting you are having invalid
>>> weight warnings.
>>> there is only one file to change i-e
>>>
>>> /usr/local/freeswitch/conf/autoload_configs/distributor.xml
>>>
>>> And reloadxml,distributor_ctl reload should work fine.
>>>
>>> can  you recheck file if sample list is still enabled as default configs
>>> come with something like 10 i think.
>>>
>>>
>>> On Tue, Aug 6, 2013 at 2:42 PM, SamyGo <govoiper at gmail.com> wrote:
>>>
>>>> Dear Saboor,
>>>>
>>>> Thanks for the quick reply, to share my luck with this, I'v even tried
>>>> a complete restart of freeswitch after trying reloadxml and sofia module
>>>> reloading as well.
>>>>
>>>> The total-weight Warning even if I configured it with 2 made me think
>>>> that I'm probably changing in the wrong file but the API command to expand
>>>> distributor list gives results for the first two attempts so that is really
>>>> weird for me.
>>>>
>>>> Just to add info, this is my FS version:
>>>>
>>>> freeswitch at internal> version
>>>> FreeSWITCH Version 1.3.14b+git~20130305T150702Z~57d6495248 (git 57d6495
>>>> 2013-03-05 15:07:02Z)
>>>>
>>>> BR
>>>> Sammy
>>>>
>>>>
>>>>
>>>>
>>>> On Tue, Aug 6, 2013 at 2:31 AM, Ahmed Sboor <ahmed at netelsat.net> wrote:
>>>>
>>>>> Hi,
>>>>> Total Weight 2 and nodes weight as 1/1 was fine . you should do first
>>>>> "reloadxml" and then "distributor_ctl reload"
>>>>>
>>>>> On Tue, Aug 6, 2013 at 2:24 PM, SamyGo <govoiper at gmail.com> wrote:
>>>>>
>>>>>> Dear users,
>>>>>>
>>>>>> I'm trying to figure out how this mod_distributor works. I've a few
>>>>>> questions to help me get this going.
>>>>>>
>>>>>> I've defined two of my gateways in the distributor.conf.xml like this:
>>>>>>
>>>>>> <configuration name="distributor.conf" description="Distributor
>>>>>> Configuration">
>>>>>>   <lists>
>>>>>>     <list name="GWLB" total-weight="2">
>>>>>>       <node name="OpenSIPS-A" weight="1"/>
>>>>>>       <node name="OpenSIPS-B" weight="1"/>
>>>>>>     </list>
>>>>>>   </lists>
>>>>>> </configuration>
>>>>>>
>>>>>> and so when I reload the distributor from API I get the following
>>>>>> warning:
>>>>>>
>>>>>> freeswitch at internal> distributor_ctl reload
>>>>>> +ok reloaded.
>>>>>> 2013-08-06 14:17:16.652087 [WARNING] mod_distributor.c:201 Total
>>>>>> weight does not add up to total weight 10
>>>>>>
>>>>>> So I've changed the total-weight to 10, and set weight of each node
>>>>>> as 5. That gives the same warning as well.
>>>>>>
>>>>>> A Warning is fine, but as soon as I just use API to expand the list I
>>>>>> get a proper gateway name for the first two times only and then -err starts
>>>>>> showing up until I reload the distributor.
>>>>>>
>>>>>> freeswitch at internal> expand distributor GWLB
>>>>>> OpenSIPS-A
>>>>>> freeswitch at internal> expand distributor GWLB
>>>>>> OpenSIPS-B
>>>>>> freeswitch at internal> expand distributor GWLB
>>>>>> -err
>>>>>> freeswitch at internal> expand distributor GWLB
>>>>>> -err
>>>>>> freeswitch at internal>
>>>>>>
>>>>>> The last question is regarding use of this distributor from LUA
>>>>>> script.
>>>>>>
>>>>>> I call in the distributor like this from my LUA.
>>>>>>
>>>>>> session =
>>>>>> freeswitch.Session("{ignore_early_media=true,origination_caller_id_number="..outbound_cli.."}sofia/gateway/${distributor(GWLB)}/"..
>>>>>> phone_number);
>>>>>>
>>>>>> and that gives a straight error:
>>>>>>
>>>>>> [ERR] mod_sofia.c:4668 Invalid Gateway '%{distributor(GWLB)}'
>>>>>>
>>>>>>
>>>>>> I hope somebody can share their knowledge on this module and help me
>>>>>> move forward with my project.
>>>>>>
>>>>>> Best Regards,
>>>>>> Sammy
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 georgi_mei at abv.bg  Tue Aug  6 11:56:42 2013
From: georgi_mei at abv.bg (Georgi Stefanov)
Date: Tue, 6 Aug 2013 10:56:42 +0300 (EEST)
Subject: [Freeswitch-users] Bridge outbound calls stay active even after
	hangup
Message-ID: <1042927593.74073.1375775802329.JavaMail.apache@mail21.abv.bg>

 Hello All,

Today I found something, which is strange or maybe it is some kind of an issue

What I am talking about is visible here

freeswitch at internal> uuid_exists a1524f1e-fece-4d4e-b0c6-1a81b02f7978
false
freeswitch at internal> show calls
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch
a1524f1e-fece-4d4e-b0c6-1a81b02f7978,outbound,2013-08-06 09:21:39,1375770099,sofia/internal/sip:LinkSysPhone at 10.102.8.101:5060,CS_EXCHANGE_MEDIA,Outbound Call,LinkSysPhone,10.102.8.101,6001,LinkSysPhone at 10.102.8.58,,ACTIVE,,,SEND,a1524f1e-fece-4d4e-b0c6-1a81b02f7978,mediagw2-stm1.dev.opencode.com,Outbound Call,GeorgiStefanov,,,,,,,,,,,,,,,,,,,

1 total.

freeswitch at internal> uuid_ki

[ uuid_kill]


freeswitch at internal> uuid_kill a1524f1e-fece-4d4e-b0c6-1a81b02f7978
-ERR No such channel!

freeswitch at internal>


1. I have made 2 outbound calls (ex: originate user/linksysphone 6001)
2. I have bridged them with uuid_bridge
3. Either phone hangup
4. Then I execute "show calls" "uuid_exist" and etc.

Am I missing something ?
Or it is an issue ?


From soeren.sprenger at aerea.de  Tue Aug  6 12:08:21 2013
From: soeren.sprenger at aerea.de (=?ISO-8859-1?Q?S=F6ren_Sprenger?=)
Date: Tue, 06 Aug 2013 10:08:21 +0200
Subject: [Freeswitch-users] freeswitch: Syntax error: word unexpected
 (expecting ")")
In-Reply-To: <CAHgk4BXaY6gPjhuG-_JR65XUL3waG34V-9DpzY0+wcUTQVNF7A@mail.gmail.com>
References: <CAHgk4BXhFoSaEyXTopMNFsDOwd=5kxFPmq5+h=e=EPbatV-mFw@mail.gmail.com>
	<CAHgk4BVhPAfJ-rdAs2pc2mG-jz86N1BsicEZThq0Y-XCWqxAWA@mail.gmail.com>
	<CAHgk4BXaY6gPjhuG-_JR65XUL3waG34V-9DpzY0+wcUTQVNF7A@mail.gmail.com>
Message-ID: <5200AEF5.3010400@aerea.de>

Hi,

what does "file /path/to/freeswitch" say which kind binary executable
file it is? Seams your freeswitch is not binary compatible with your
running system...


On 06.08.2013 09:54, Brian Foster wrote:
>
> Fresh build on a bare metal Dell Precision 490 running Ubuntu 12.04
> 64-bit, latest v1.2.stable (5 min ago). When I try to run freeswitch,
> I get:
>
> freeswitch:1 freeswitch: Syntax error: word unexpected (expecting ")")
>
> I've since deleted everything in /usr/local/freeswitch and
> /usr/local/src/freeswitch, and ran the install again. Still have the
> same outcome.
>
> The same version was built on another server, no issues. What could be
> the problem? Anything I can do to mitigate the issue?
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Representative
> Davri Investments, Incorporated
> P: +1-317-787-2686
> M: +1-317-600-9753
> Indianapolis, Indiana
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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


-- 
AereA NetworX UG (haftungsbeschr?nkt)
Im Camisch 8
07768 Kahla
Gesch?ftsf?hrerin: Franziska Sprenger
Handelsregister: Jena B504724
E-Mail: info at aerea.de
Web: http://www.aerea.de
Fon: +49 (0) 36424 760823
Fax: +49 (0) 36651 1390009

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From menendez.garcia at gmail.com  Tue Aug  6 15:32:53 2013
From: menendez.garcia at gmail.com (Javier Menendez)
Date: Tue, 6 Aug 2013 13:32:53 +0200
Subject: [Freeswitch-users] webrtc INCOMPATIBLE_DESTINATION
Message-ID: <CAK59s4y6YHecUr-Te-TMvJoS=ASxthL8FCuYmNYhEmkfQioFmQ@mail.gmail.com>

Hi,

I am trying to make an outbound call to a webrtc softphone using jssip, I
initiate the call from an asterisk box :

[Asterisk] -> [FS] ->[jssip]

I always get an  INCOMPATIBLE_DESTINATION error, looking at the trace logs
I found out that the problem is the codec negotiation but I can not make it
work, AFAIK the call should use the ALAW codec as it is compatible with all
legs involved. this is the asterisk SDP

2013-08-06 13:16:50.892293 [DEBUG] sofia.c:5802 Remote SDP:
v=0
o=root 451068671 451068671 IN IP4 192.168.90.16
s=Asterisk PBX 1.8.16.0
c=IN IP4 192.168.90.16
t=0 0
m=audio 16126 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

And this is the SDP from FS wich is sent to the jssip client

2013-08-06 13:16:50.912323 [DEBUG] sofia_glue.c:1220 Local SDP:
v=0
o=FreeSWITCH 1375766038 1375766039 IN IP4 PUBLIC_IP
s=FreeSWITCH
c=IN IP4 PUBLIC_IP
t=0 0
a=msid-semantic: WMS 9hJmtW2hnx2uG5S2obryUBUKGF35tgps
m=audio 21772 RTP/SAVPF 8 0 101 13
a=rtpmap:101 telephone-event/8000
a=rtcp-mux
a=rtcp:21772 IN IP4 PUBLIC_IP
a=ssrc:3456358538 cname:oZFIMxPZUIkrD3jS
a=ssrc:3456358538 msid:9hJmtW2hnx2uG5S2obryUBUKGF35tgps a0
a=ssrc:3456358538 mslabel:9hJmtW2hnx2uG5S2obryUBUKGF35tgps
a=ssrc:3456358538 label:9hJmtW2hnx2uG5S2obryUBUKGF35tgpsa0
a=ice-ufrag:4BgP20qzvjuPqK7E
a=ice-pwd:ADcc0S2Lqc2T5cB0
a=candidate:3263618716 1 udp 659136 212.230.135.231 21772 typ host
generation 0
a=candidate:3263618716 2 udp 659136 212.230.135.231 21772 typ host
generation 0
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:JbAOcaClBcoNLW6zjoucy5BU0mfS+UQqWcyYh9+7
a=ptime:20
a=sendrecv

Shouldn't it include the PCMA codec? I hace tried to enable it in
configuration but it doesn't work, also tried to set it with

  <action application="export"
data="nolocal:absolute_codec_string=PCMA,PCMU"/>

before bridge, but no luck.

Funny thing is that it is working if i innitiate the call from the jssip
client. [jssip] -> [FS] ->[Asterisk]

Any clue?
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From anthony.minessale at gmail.com  Tue Aug  6 17:25:23 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Aug 2013 08:25:23 -0500
Subject: [Freeswitch-users] A few problems with latest GIT
In-Reply-To: <BE4C37BE-3FDC-4E12-8718-CD8607A8AC81@kavun.ch>
References: <CAAtWKeEW+ngi=7xq=0LOuYSPg4_JUwzFjqDBMMheskDy40bFvg@mail.gmail.com>
	<BE4C37BE-3FDC-4E12-8718-CD8607A8AC81@kavun.ch>
Message-ID: <CAKbxfG-jjkHY2xz1j4AM2GyenRz0nYsn1inNxnm=2LyXYoWC3A@mail.gmail.com>

If you can find a way to reproduce the left over sessions problem, its the
only way to find the cause.  Its not something that happens by default.  It
could be related to something specific you are doing like leaving a lua
script in an endless loop or a bad build.

Try a full clean build of a fresh git checkout against the default configs
and see if you can still reproduce your problem and post the details to
Jira.



On Tue, Aug 6, 2013 at 3:37 AM, Emrah <lists at kavun.ch> wrote:

> I confirm the last issue of your list has also happened on my latest GIT.
> Your no audio syndrome sounds like it could be connected to my Polycom
> bridging issue.
> As for the sluggishnness of fs_cli I haven't noticed anything or just
> associated it to my mobile broadband?
> On Aug 6, 2013, at 1:02 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
> wrote:
>
> Hello,
>
>   I am running FS version 1.0.x and trying to move to 1.2/1.4/latest-git.
> I encounter a few problems which prevents me from using these versions, and
> would like to know whether someone else is having it. It is quite hard to
> create debugging information, as with full logging the behaviour is
> slightly different than lower logging level. The problems appear only on my
> production system, around noon (when the number of active sessins goes
> beyond 50-60 sessions):
>
>
>    - The voice path is lost in the middle of the session (Jira FS-5616).
>    From TCPdump I see that FS receive the audio from both sides but stops
>    sending it; there is nothing meaningfull in the logs.
>    - FS starts to get sluggish, fs_cli is stuck, SIP messages are
>    processed in delay (5-30 seconds). This goes away by itself after a few
>    minutes.
>    - sessions are left "in use" after the users hangup, and the number of
>    active sessions starts rising. Doing uuid_kill succeeds once (next times
>    fail with "no such uuid", but "show channels" still shows it.
>
> The most frustrating side is that I do not know what exact debugging tools
> to use in order to submit meaningfull data to FS developers.
>
>                      Thanks! __Yehavi:
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From anthony.minessale at gmail.com  Tue Aug  6 17:26:50 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Aug 2013 08:26:50 -0500
Subject: [Freeswitch-users] webrtc INCOMPATIBLE_DESTINATION
In-Reply-To: <CAK59s4y6YHecUr-Te-TMvJoS=ASxthL8FCuYmNYhEmkfQioFmQ@mail.gmail.com>
References: <CAK59s4y6YHecUr-Te-TMvJoS=ASxthL8FCuYmNYhEmkfQioFmQ@mail.gmail.com>
Message-ID: <CAKbxfG96LTXzrB92TGNEkFvmTU9bZz1WX=VFK9p+a_Hra9ZDBg@mail.gmail.com>

try prepending the bridge url with {absolute_codec_string=PCMA}


On Tue, Aug 6, 2013 at 6:32 AM, Javier Menendez
<menendez.garcia at gmail.com>wrote:

> Hi,
>
> I am trying to make an outbound call to a webrtc softphone using jssip, I
> initiate the call from an asterisk box :
>
> [Asterisk] -> [FS] ->[jssip]
>
> I always get an  INCOMPATIBLE_DESTINATION error, looking at the trace logs
> I found out that the problem is the codec negotiation but I can not make it
> work, AFAIK the call should use the ALAW codec as it is compatible with all
> legs involved. this is the asterisk SDP
>
> 2013-08-06 13:16:50.892293 [DEBUG] sofia.c:5802 Remote SDP:
> v=0
> o=root 451068671 451068671 IN IP4 192.168.90.16
> s=Asterisk PBX 1.8.16.0
> c=IN IP4 192.168.90.16
> t=0 0
> m=audio 16126 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> And this is the SDP from FS wich is sent to the jssip client
>
> 2013-08-06 13:16:50.912323 [DEBUG] sofia_glue.c:1220 Local SDP:
> v=0
> o=FreeSWITCH 1375766038 1375766039 IN IP4 PUBLIC_IP
> s=FreeSWITCH
> c=IN IP4 PUBLIC_IP
> t=0 0
> a=msid-semantic: WMS 9hJmtW2hnx2uG5S2obryUBUKGF35tgps
> m=audio 21772 RTP/SAVPF 8 0 101 13
> a=rtpmap:101 telephone-event/8000
> a=rtcp-mux
> a=rtcp:21772 IN IP4 PUBLIC_IP
> a=ssrc:3456358538 cname:oZFIMxPZUIkrD3jS
> a=ssrc:3456358538 msid:9hJmtW2hnx2uG5S2obryUBUKGF35tgps a0
> a=ssrc:3456358538 mslabel:9hJmtW2hnx2uG5S2obryUBUKGF35tgps
> a=ssrc:3456358538 label:9hJmtW2hnx2uG5S2obryUBUKGF35tgpsa0
> a=ice-ufrag:4BgP20qzvjuPqK7E
> a=ice-pwd:ADcc0S2Lqc2T5cB0
> a=candidate:3263618716 1 udp 659136 212.230.135.231 21772 typ host
> generation 0
> a=candidate:3263618716 2 udp 659136 212.230.135.231 21772 typ host
> generation 0
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:JbAOcaClBcoNLW6zjoucy5BU0mfS+UQqWcyYh9+7
> a=ptime:20
> a=sendrecv
>
> Shouldn't it include the PCMA codec? I hace tried to enable it in
> configuration but it doesn't work, also tried to set it with
>
>   <action application="export"
> data="nolocal:absolute_codec_string=PCMA,PCMU"/>
>
> before bridge, but no luck.
>
> Funny thing is that it is working if i innitiate the call from the jssip
> client. [jssip] -> [FS] ->[Asterisk]
>
> Any clue?
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From anthony.minessale at gmail.com  Tue Aug  6 17:30:25 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Aug 2013 08:30:25 -0500
Subject: [Freeswitch-users] Bridge outbound calls stay active even after
	hangup
In-Reply-To: <1042927593.74073.1375775802329.JavaMail.apache@mail21.abv.bg>
References: <1042927593.74073.1375775802329.JavaMail.apache@mail21.abv.bg>
Message-ID: <CAKbxfG_=y49pmQQLeen5GfLWA2dWQbNSNhOaFggT0Ak66j8+Lg@mail.gmail.com>

You may need to rebuild to eliminate build skew as a possibility.  I tried
the same recipe on my dev box and I don't see that problem.
There are some others with the same problem but its clearly not something
apparent from a default build env.
Try "make sure" to force a full rebuild and clean git tree.





On Tue, Aug 6, 2013 at 2:56 AM, Georgi Stefanov <georgi_mei at abv.bg> wrote:

>  Hello All,
>
> Today I found something, which is strange or maybe it is some kind of an
> issue
>
> What I am talking about is visible here
>
> freeswitch at internal> uuid_exists a1524f1e-fece-4d4e-b0c6-1a81b02f7978
> false
> freeswitch at internal> show calls
>
> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch
> a1524f1e-fece-4d4e-b0c6-1a81b02f7978,outbound,2013-08-06
> 09:21:39,1375770099,sofia/internal/sip:LinkSysPhone at 10.102.8.101:5060,CS_EXCHANGE_MEDIA,Outbound
> Call,LinkSysPhone,10.102.8.101,6001,LinkSysPhone at 10.102.8.58
> ,,ACTIVE,,,SEND,a1524f1e-fece-4d4e-b0c6-1a81b02f7978,
> mediagw2-stm1.dev.opencode.com,Outbound
> Call,GeorgiStefanov,,,,,,,,,,,,,,,,,,,
>
> 1 total.
>
> freeswitch at internal> uuid_ki
>
> [ uuid_kill]
>
>
> freeswitch at internal> uuid_kill a1524f1e-fece-4d4e-b0c6-1a81b02f7978
> -ERR No such channel!
>
> freeswitch at internal>
>
>
> 1. I have made 2 outbound calls (ex: originate user/linksysphone 6001)
> 2. I have bridged them with uuid_bridge
> 3. Either phone hangup
> 4. Then I execute "show calls" "uuid_exist" and etc.
>
> Am I missing something ?
> Or it is an issue ?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From menendez.garcia at gmail.com  Tue Aug  6 17:41:30 2013
From: menendez.garcia at gmail.com (Javier Menendez)
Date: Tue, 6 Aug 2013 15:41:30 +0200
Subject: [Freeswitch-users] webrtc INCOMPATIBLE_DESTINATION
In-Reply-To: <CAKbxfG96LTXzrB92TGNEkFvmTU9bZz1WX=VFK9p+a_Hra9ZDBg@mail.gmail.com>
References: <CAK59s4y6YHecUr-Te-TMvJoS=ASxthL8FCuYmNYhEmkfQioFmQ@mail.gmail.com>
	<CAKbxfG96LTXzrB92TGNEkFvmTU9bZz1WX=VFK9p+a_Hra9ZDBg@mail.gmail.com>
Message-ID: <CAK59s4yjYoEDVJJyEx2CE0z=tqn_aSNvg76TXBVrFUPR3as=gQ@mail.gmail.com>

No luck :(

EXECUTE sofia/internal/asterisk at 192.168.90.16export(nolocal:absolute_codec_string=PCMA,PCMU)
2013-08-06 15:37:47.312306 [DEBUG] switch_channel.c:1222 EXPORT
(export_vars) (REMOTE ONLY) [absolute_codec_string]=[PCMA,PCMU]
EXECUTE sofia/internal/asterisk at 192.168.90.16bridge({absolute_codec_string=PCMA}sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid
;transport=ws;fs_nat=yes;fs_path=sip%3A4st0031l%40PUBLIC_IP%3A38333%3Btransport%3Dws)
2013-08-06 15:37:47.312306 [DEBUG] switch_channel.c:1176 sofia/internal/
asterisk at 192.168.90.16 EXPORTING[export_vars]
[absolute_codec_string]=[PCMA,PCMU] to event
2013-08-06 15:37:47.312306 [DEBUG] switch_ivr_originate.c:2050 Parsing
global variables
2013-08-06 15:37:47.312306 [DEBUG] switch_event.c:1615 Parsing variable
[absolute_codec_string]=[PCMA]
2013-08-06 15:37:47.312306 [NOTICE] switch_channel.c:1030 New Channel
sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid[617a9f0e-fe9d-11e2-b1fd-b761e46dca72]
2013-08-06 15:37:47.312306 [DEBUG] mod_sofia.c:4420
(sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid) State Change CS_NEW ->
CS_INIT
2013-08-06 15:37:47.312306 [DEBUG] switch_core_session.c:1341 Send signal
sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid [BREAK]
2013-08-06 15:37:47.312306 [DEBUG] switch_core_state_machine.c:416
(sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid) Running State Change
CS_INIT
2013-08-06 15:37:47.312306 [DEBUG] switch_core_state_machine.c:455
(sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid) State INIT
2013-08-06 15:37:47.312306 [DEBUG] mod_sofia.c:87
sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid SOFIA INIT
2013-08-06 15:37:47.312306 [DEBUG] switch_core_media.c:681 Set Local Key [1
AES_CM_128_HMAC_SHA1_80 inline:SvDPyP7iDr8+xmeRvVRrrqHinvGB/Ht8+VArR6Ro]
2013-08-06 15:37:47.312306 [DEBUG] switch_core_media.c:681 Set Local Key [1
AES_CM_128_HMAC_SHA1_80 inline:wUFqlClOJwLZaHzE+QlEQfVQEx979Gx4e7BCbl56]
2013-08-06 15:37:47.312306 [DEBUG] sofia_glue.c:1191
sip:4st0031l at PUBLIC_IP:38333;transport=ws
Setting proxy route to sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid
2013-08-06 15:37:47.312306 [DEBUG] sofia_glue.c:1220 Local SDP:
v=0
o=FreeSWITCH 1375775801 1375775802 IN IP4 PUBLIC_IP
s=FreeSWITCH
c=IN IP4 PUBLIC_IP
t=0 0
a=msid-semantic: WMS B54oKgHfL2pj5wroyvgO0c1ghIMukBaL
m=audio 20466 RTP/SAVPF 8 101 13
a=rtpmap:101 telephone-event/8000
a=rtcp-mux
a=rtcp:20466 IN IP4 PUBLIC_IP
a=ssrc:3456367315 cname:w406axcLqrVlOdzc
a=ssrc:3456367315 msid:B54oKgHfL2pj5wroyvgO0c1ghIMukBaL a0
a=ssrc:3456367315 mslabel:B54oKgHfL2pj5wroyvgO0c1ghIMukBaL
a=ssrc:3456367315 label:B54oKgHfL2pj5wroyvgO0c1ghIMukBaLa0
a=ice-ufrag:zGQpJHvQg7bqzb2V
a=ice-pwd:5SS6gUIwcuIQy7TX
a=candidate:3772579260 1 udp 659136 PUBLIC_IP 20466 typ host generation 0
a=candidate:3772579260 2 udp 659136 PUBLIC_IP 20466 typ host generation 0
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:SvDPyP7iDr8+xmeRvVRrrqHinvGB/Ht8+VArR6Ro
a=ptime:20
a=sendrecv



On Tue, Aug 6, 2013 at 3:26 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> try prepending the bridge url with {absolute_codec_string=PCMA}
>
>
> On Tue, Aug 6, 2013 at 6:32 AM, Javier Menendez <menendez.garcia at gmail.com
> > wrote:
>
>> Hi,
>>
>> I am trying to make an outbound call to a webrtc softphone using jssip, I
>> initiate the call from an asterisk box :
>>
>> [Asterisk] -> [FS] ->[jssip]
>>
>> I always get an  INCOMPATIBLE_DESTINATION error, looking at the trace
>> logs I found out that the problem is the codec negotiation but I can not
>> make it work, AFAIK the call should use the ALAW codec as it is compatible
>> with all legs involved. this is the asterisk SDP
>>
>> 2013-08-06 13:16:50.892293 [DEBUG] sofia.c:5802 Remote SDP:
>> v=0
>> o=root 451068671 451068671 IN IP4 192.168.90.16
>> s=Asterisk PBX 1.8.16.0
>> c=IN IP4 192.168.90.16
>> t=0 0
>> m=audio 16126 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> And this is the SDP from FS wich is sent to the jssip client
>>
>> 2013-08-06 13:16:50.912323 [DEBUG] sofia_glue.c:1220 Local SDP:
>> v=0
>> o=FreeSWITCH 1375766038 1375766039 IN IP4 PUBLIC_IP
>> s=FreeSWITCH
>> c=IN IP4 PUBLIC_IP
>> t=0 0
>> a=msid-semantic: WMS 9hJmtW2hnx2uG5S2obryUBUKGF35tgps
>> m=audio 21772 RTP/SAVPF 8 0 101 13
>> a=rtpmap:101 telephone-event/8000
>> a=rtcp-mux
>> a=rtcp:21772 IN IP4 PUBLIC_IP
>> a=ssrc:3456358538 cname:oZFIMxPZUIkrD3jS
>> a=ssrc:3456358538 msid:9hJmtW2hnx2uG5S2obryUBUKGF35tgps a0
>> a=ssrc:3456358538 mslabel:9hJmtW2hnx2uG5S2obryUBUKGF35tgps
>> a=ssrc:3456358538 label:9hJmtW2hnx2uG5S2obryUBUKGF35tgpsa0
>> a=ice-ufrag:4BgP20qzvjuPqK7E
>> a=ice-pwd:ADcc0S2Lqc2T5cB0
>> a=candidate:3263618716 1 udp 659136 212.230.135.231 21772 typ host
>> generation 0
>> a=candidate:3263618716 2 udp 659136 212.230.135.231 21772 typ host
>> generation 0
>> a=crypto:1 AES_CM_128_HMAC_SHA1_80
>> inline:JbAOcaClBcoNLW6zjoucy5BU0mfS+UQqWcyYh9+7
>> a=ptime:20
>> a=sendrecv
>>
>> Shouldn't it include the PCMA codec? I hace tried to enable it in
>> configuration but it doesn't work, also tried to set it with
>>
>>   <action application="export"
>> data="nolocal:absolute_codec_string=PCMA,PCMU"/>
>>
>> before bridge, but no luck.
>>
>> Funny thing is that it is working if i innitiate the call from the jssip
>> client. [jssip] -> [FS] ->[Asterisk]
>>
>> Any clue?
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Tue Aug  6 18:22:02 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Tue, 6 Aug 2013 19:52:02 +0530
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAMvhjJtbZTd9_s37EOsem42JsOszYwc9ZEZ5V_k1=+j4CLguaQ@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<52001B43.9080901@freeswitch.org>
	<CAMKwYLCtQV1wCVzPxwhNj2d23eK__uPDYPb98p0ZiJLRqx0GKQ@mail.gmail.com>
	<CAMvhjJt8v0XVnGqBs5pMgEyNR3tDV6GuMFR5Dsb1x6XMCMO4tw@mail.gmail.com>
	<CAMvhjJtbZTd9_s37EOsem42JsOszYwc9ZEZ5V_k1=+j4CLguaQ@mail.gmail.com>
Message-ID: <CAMvhjJugTnGTj5i5pO1r7Q+XeQr658b=WsXOXOuZEnVuo004hQ@mail.gmail.com>

The mod_event_socket was loaded...
On Aug 6, 2013 1:23 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:

> How to load mod_event_socket...???
>
> Yes my freeswitch is running...
> On Aug 6, 2013 3:41 AM, "Gustavo Salazar" <guga.salazar.loor at gmail.com>
> wrote:
>
>>
>>
>> Is Freeswitch running?
>> I have seen a similar error when I tried to start the cli and Freeswitch
>> was not running .
>>
>>
>>
>> 2013/8/5 Yossi Neiman <mishehu at freeswitch.org>
>>
>>>  Do you have mod_event_socket loaded?
>>>
>>> -Yossi
>>>
>>>
>>> On 08/05/2013 04:00 PM, Ashish Mishra wrote:
>>>
>>> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i
>>> have installed freeswitch) it gives me the following error :
>>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> 
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>>
>> --
>> Gustavo Salazar
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 treitinger at as-infodienste.de  Tue Aug  6 18:29:39 2013
From: treitinger at as-infodienste.de (Melanie Treitinger)
Date: Tue, 06 Aug 2013 16:29:39 +0200
Subject: [Freeswitch-users] uuid_displace: change loudness level of audio
Message-ID: <52010853.2020603@as-infodienste.de>

Hi,

I want to use uuid_displace to play announcements while someone is 
listening to music on hold.
This works fine, except that the music on hold is too loud - it should 
be reduced when the uuid_displace takes place.
Is there a way to achieve this?




From msc at freeswitch.org  Tue Aug  6 19:38:34 2013
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Aug 2013 08:38:34 -0700
Subject: [Freeswitch-users] uuid_displace: change loudness level of audio
In-Reply-To: <52010853.2020603@as-infodienste.de>
References: <52010853.2020603@as-infodienste.de>
Message-ID: <CAKzWOxU9Bwd3eAhbTtrmv9vjwSdSC-1Fee0MYu7szqSGtU9=iA@mail.gmail.com>

Hi Melanie,

If this is a wav file and the only purpose is to be played to callers then
your best bet is to change the volume on the file itself. That way you
don't waste CPU resources each time you play the file.

Have you ever used the sox program?

-MC
On Aug 6, 2013 10:14 AM, "Melanie Treitinger" <treitinger at as-infodienste.de>
wrote:

> Hi,
>
> I want to use uuid_displace to play announcements while someone is
> listening to music on hold.
> This works fine, except that the music on hold is too loud - it should
> be reduced when the uuid_displace takes place.
> Is there a way to achieve this?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jleung at v10networks.ca  Tue Aug  6 19:44:19 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Tue, 6 Aug 2013 08:44:19 -0700
Subject: [Freeswitch-users] uuid_displace: change loudness level of audio
In-Reply-To: <CAKzWOxU9Bwd3eAhbTtrmv9vjwSdSC-1Fee0MYu7szqSGtU9=iA@mail.gmail.com>
References: <52010853.2020603@as-infodienste.de>
	<CAKzWOxU9Bwd3eAhbTtrmv9vjwSdSC-1Fee0MYu7szqSGtU9=iA@mail.gmail.com>
Message-ID: <00b901ce92bb$d1c32590$754970b0$@v10networks.ca>

You may want to turn the audio file down by -20dB or so. You'd never want to
run full levels especially if you're playing back audio into the PSTN.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, August 6, 2013 8:39 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] uuid_displace: change loudness level of
audio

 

Hi Melanie,

If this is a wav file and the only purpose is to be played to callers then
your best bet is to change the volume on the file itself. That way you don't
waste CPU resources each time you play the file. 

Have you ever used the sox program?

-MC

On Aug 6, 2013 10:14 AM, "Melanie Treitinger" <treitinger at as-infodienste.de>
wrote:

Hi,

I want to use uuid_displace to play announcements while someone is
listening to music on hold.
This works fine, except that the music on hold is too loud - it should
be reduced when the uuid_displace takes place.
Is there a way to achieve this?



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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  Tue Aug  6 20:02:08 2013
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Aug 2013 09:02:08 -0700
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAMvhjJugTnGTj5i5pO1r7Q+XeQr658b=WsXOXOuZEnVuo004hQ@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<52001B43.9080901@freeswitch.org>
	<CAMKwYLCtQV1wCVzPxwhNj2d23eK__uPDYPb98p0ZiJLRqx0GKQ@mail.gmail.com>
	<CAMvhjJt8v0XVnGqBs5pMgEyNR3tDV6GuMFR5Dsb1x6XMCMO4tw@mail.gmail.com>
	<CAMvhjJtbZTd9_s37EOsem42JsOszYwc9ZEZ5V_k1=+j4CLguaQ@mail.gmail.com>
	<CAMvhjJugTnGTj5i5pO1r7Q+XeQr658b=WsXOXOuZEnVuo004hQ@mail.gmail.com>
Message-ID: <CAKzWOxU1cX46+Tn0gchbOzrFucU2WGbzPM-B-GPRbaCrxCnjGg@mail.gmail.com>

Check the settings in conf/autoload_configs/event_socket.conf.xml and make
sure the listen IP, listen port, and password are what you think they are.
If you need to change any of them just do fs_cli -h to see the options.
-MC


On Tue, Aug 6, 2013 at 7:22 AM, Ashish Mishra <itsme.kunnu at gmail.com> wrote:

> The mod_event_socket was loaded...
> On Aug 6, 2013 1:23 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>
>> How to load mod_event_socket...???
>>
>> Yes my freeswitch is running...
>> On Aug 6, 2013 3:41 AM, "Gustavo Salazar" <guga.salazar.loor at gmail.com>
>> wrote:
>>
>>>
>>>
>>> Is Freeswitch running?
>>> I have seen a similar error when I tried to start the cli and Freeswitch
>>> was not running .
>>>
>>>
>>>
>>> 2013/8/5 Yossi Neiman <mishehu at freeswitch.org>
>>>
>>>>  Do you have mod_event_socket loaded?
>>>>
>>>> -Yossi
>>>>
>>>>
>>>> On 08/05/2013 04:00 PM, Ashish Mishra wrote:
>>>>
>>>> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which
>>>> i have installed freeswitch) it gives me the following error :
>>>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>>
>>>> 
>>>>
>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>>
>>> --
>>> Gustavo Salazar
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Tue Aug  6 20:06:24 2013
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 6 Aug 2013 09:06:24 -0700
Subject: [Freeswitch-users] Bridge outbound calls stay active even after
	hangup
In-Reply-To: <CAKbxfG_=y49pmQQLeen5GfLWA2dWQbNSNhOaFggT0Ak66j8+Lg@mail.gmail.com>
References: <1042927593.74073.1375775802329.JavaMail.apache@mail21.abv.bg>
	<CAKbxfG_=y49pmQQLeen5GfLWA2dWQbNSNhOaFggT0Ak66j8+Lg@mail.gmail.com>
Message-ID: <CAKzWOxVawLe7H+C_hygeP+O=+GOCL7hARC47XBmhQh5Ko4_jPg@mail.gmail.com>

https://wiki.freeswitch.org/wiki/Haiku#Updating


On Tue, Aug 6, 2013 at 6:30 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You may need to rebuild to eliminate build skew as a possibility.  I tried
> the same recipe on my dev box and I don't see that problem.
> There are some others with the same problem but its clearly not something
> apparent from a default build env.
> Try "make sure" to force a full rebuild and clean git tree.
>
>
>
>
>
> On Tue, Aug 6, 2013 at 2:56 AM, Georgi Stefanov <georgi_mei at abv.bg> wrote:
>
>>  Hello All,
>>
>> Today I found something, which is strange or maybe it is some kind of an
>> issue
>>
>> What I am talking about is visible here
>>
>> freeswitch at internal> uuid_exists a1524f1e-fece-4d4e-b0c6-1a81b02f7978
>> false
>> freeswitch at internal> show calls
>>
>> uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,hostname,sent_callee_name,sent_callee_num,b_uuid,b_direction,b_created,b_created_epoch,b_name,b_state,b_cid_name,b_cid_num,b_ip_addr,b_dest,b_presence_id,b_presence_data,b_callstate,b_callee_name,b_callee_num,b_callee_direction,b_sent_callee_name,b_sent_callee_num,call_created_epoch
>> a1524f1e-fece-4d4e-b0c6-1a81b02f7978,outbound,2013-08-06
>> 09:21:39,1375770099,sofia/internal/sip:LinkSysPhone at 10.102.8.101:5060,CS_EXCHANGE_MEDIA,Outbound
>> Call,LinkSysPhone,10.102.8.101,6001,LinkSysPhone at 10.102.8.58
>> ,,ACTIVE,,,SEND,a1524f1e-fece-4d4e-b0c6-1a81b02f7978,
>> mediagw2-stm1.dev.opencode.com,Outbound
>> Call,GeorgiStefanov,,,,,,,,,,,,,,,,,,,
>>
>> 1 total.
>>
>> freeswitch at internal> uuid_ki
>>
>> [ uuid_kill]
>>
>>
>> freeswitch at internal> uuid_kill a1524f1e-fece-4d4e-b0c6-1a81b02f7978
>> -ERR No such channel!
>>
>> freeswitch at internal>
>>
>>
>> 1. I have made 2 outbound calls (ex: originate user/linksysphone 6001)
>> 2. I have bridged them with uuid_bridge
>> 3. Either phone hangup
>> 4. Then I execute "show calls" "uuid_exist" and etc.
>>
>> Am I missing something ?
>> Or it is an issue ?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From sravi123 at yahoo.com  Tue Aug  6 20:42:37 2013
From: sravi123 at yahoo.com (Ravi S)
Date: Tue, 06 Aug 2013 22:12:37 +0530
Subject: [Freeswitch-users] Help with PRI Configuration - Bharti Airtel
 - Sangoma 101DE - Freeswitch
In-Reply-To: <1375720572.96851.YahooMailNeo@web160505.mail.bf1.yahoo.com>
References: <51FEBEF0.6090500@yahoo.com>	<B757A1DB-090B-479E-A18D-001A7B4C0F6D@freeswitch.org>
	<CAKzWOxUDrQ=bw5wPsVCwAkO=yZSdYHGjFq6=N=rVJMh8wAfvhw@mail.gmail.com>
	<1375720572.96851.YahooMailNeo@web160505.mail.bf1.yahoo.com>
Message-ID: <5201277D.8050709@yahoo.com>

Hello Everyone,

Thank you for your responses. For the problem below, the solution is to 
install the following:

yum -y install kernel-devel-$(uname -r) libtool* make gcc patch perl 
bison gcc-c++ ncurses-devel flex libtermcap-devel autoconf* automake* 
autoconf


However, I am still struggling to set the PRI card up. will post details 
with log shortly.

Thanks.
Ravi


On 05-08-2013 22:06, Ravi wrote:
> Thanks Ken,
>
> Previously I did not have trouble installing the Sangoma card. Now 
> when I try to go through the FS freetdm wiki, this is the error that I 
> am getting:
>
> [root at bfree-server wanpipe-7.0.5]# make freetdm
>    Error linux headers/source not found: 
> /lib/modules/2.6.32-358.14.1.el6.x86_64/build !
>
> make: *** [_checksrc] Error 1
>
>
> any reason why this is happening ??
>
> Thanks.
> Ravi
>
>
> ------------------------------------------------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Monday, August 5, 2013 7:12 AM
> *Subject:* Re: [Freeswitch-users] Help with PRI Configuration - Bharti 
> Airtel - Sangoma 101DE - Freeswitch
>
> Do they supply a "real" PRI or is it an IP connection with a device to 
> convert to PRI?
> On Aug 4, 2013 4:43 PM, "Ken Rice" <krice at freeswitch.org 
> <mailto:krice at freeswitch.org>> wrote:
>
>     you should review the instructions for setting up PRI on the fs
>     wiki... a prindoes not have a username or a password
>
>     Ken
>     Sent from my iPad
>
>     On Aug 4, 2013, at 15:52, Ravi <sravi123 at yahoo.com
>     <mailto:sravi123 at yahoo.com>> wrote:
>
>     > Hello Everyone !
>     >
>     > I am from India. I have recently taken a PRI connection from Bharti
>     > Airtel, one of the service providers. I have installed the
>     following:
>     >
>     > Cent OS
>     > Freeswitch, FreeTDM
>     > Sangoma Card
>     >
>     >
>     > I think, I have followed all the instructions. I am struggling to
>     > configure Freeswitch to start using the PRI connection. This is
>     what I
>     > have from the freeswitch cook book, to configure the gateway: we
>     need
>     > username/password, server address or IP and port.
>     >
>     > When I checked with Airtel, they are telling me that they only give
>     > username/password and IP address details for an internet
>     connection and
>     > not for PRI lines.
>     >
>     > Has anyone here in the list, tried using an indian service
>     provider and
>     > configured in Freeswitch?
>     > Has anyone done it for Airtel ?
>     > Or please help me to figure out how to configure the PRI so as
>     to make
>     > inbound and outbound calls ??
>     >
>     > Any help is much appreciated.
>     >
>     > Thanks.
>     > Ravi
>     > +91-7502029000
>     >
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     > http://www.freeswitchsolutions.com
>     >
>     > 
>     > 
>     >
>     > Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://wiki.freeswitch.org
>     > http://www.cluecon.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 <http://www.freeswitch.org/>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com/>
>
>     
>      </>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org <http://www.freeswitch.org/>
>     http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>     http://www.cluecon.com <http://www.cluecon.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 <http://www.freeswitch.org/>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
>
> 
>  </>
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org <http://www.freeswitch.org/>
> http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
> http://www.cluecon.com <http://www.cluecon.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 <http://www.freeswitch.org/>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 govoiper at gmail.com  Tue Aug  6 21:05:17 2013
From: govoiper at gmail.com (SamyGo)
Date: Tue, 6 Aug 2013 10:05:17 -0700
Subject: [Freeswitch-users] Do we have ESL-proxy?
Message-ID: <CAJUJwtgjPCgbEiJKU_iYTCbwJ26p08K7M39xF4K9TLaBZD=UTw@mail.gmail.com>

Dear Users,

I have come up with a requirement to have ESL connection made with multiple
FS Servers at the same time and send them commands(no event receiving is
required for now) The connections to FS should be pooled and monitored if
any FS goes down.

Is there any ESLproxy available to be used readily?

Thanks,
Sammy
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From menendez.garcia at gmail.com  Tue Aug  6 21:14:10 2013
From: menendez.garcia at gmail.com (Javier Menendez)
Date: Tue, 6 Aug 2013 19:14:10 +0200
Subject: [Freeswitch-users] webrtc INCOMPATIBLE_DESTINATION
In-Reply-To: <CAK59s4yjYoEDVJJyEx2CE0z=tqn_aSNvg76TXBVrFUPR3as=gQ@mail.gmail.com>
References: <CAK59s4y6YHecUr-Te-TMvJoS=ASxthL8FCuYmNYhEmkfQioFmQ@mail.gmail.com>
	<CAKbxfG96LTXzrB92TGNEkFvmTU9bZz1WX=VFK9p+a_Hra9ZDBg@mail.gmail.com>
	<CAK59s4yjYoEDVJJyEx2CE0z=tqn_aSNvg76TXBVrFUPR3as=gQ@mail.gmail.com>
Message-ID: <CAK59s4yH0-B8nxK30k7TBgDgivDwJzPfDTeE0kOXiuKKdrFvXQ@mail.gmail.com>

After trying and trying I found out it is a compatibility version with the
browser.

Using Chrome 26 I can call from jssip client but not to jssip client
Using Chrom 28 I can call both from and to jssip client
Using Firefox 23 I can't do anything!

is there anything I can do in freeswitch to improve compatibility? any
experience?


On Tue, Aug 6, 2013 at 3:41 PM, Javier Menendez
<menendez.garcia at gmail.com>wrote:

> No luck :(
>
> EXECUTE sofia/internal/asterisk at 192.168.90.16export(nolocal:absolute_codec_string=PCMA,PCMU)
> 2013-08-06 15:37:47.312306 [DEBUG] switch_channel.c:1222 EXPORT
> (export_vars) (REMOTE ONLY) [absolute_codec_string]=[PCMA,PCMU]
> EXECUTE sofia/internal/asterisk at 192.168.90.16bridge({absolute_codec_string=PCMA}sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid
> ;transport=ws;fs_nat=yes;fs_path=sip%3A4st0031l%40PUBLIC_IP%3A38333%3Btransport%3Dws)
> 2013-08-06 15:37:47.312306 [DEBUG] switch_channel.c:1176 sofia/internal/
> asterisk at 192.168.90.16 EXPORTING[export_vars]
> [absolute_codec_string]=[PCMA,PCMU] to event
> 2013-08-06 15:37:47.312306 [DEBUG] switch_ivr_originate.c:2050 Parsing
> global variables
> 2013-08-06 15:37:47.312306 [DEBUG] switch_event.c:1615 Parsing variable
> [absolute_codec_string]=[PCMA]
> 2013-08-06 15:37:47.312306 [NOTICE] switch_channel.c:1030 New Channel
> sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid[617a9f0e-fe9d-11e2-b1fd-b761e46dca72]
> 2013-08-06 15:37:47.312306 [DEBUG] mod_sofia.c:4420
> (sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid) State Change CS_NEW ->
> CS_INIT
> 2013-08-06 15:37:47.312306 [DEBUG] switch_core_session.c:1341 Send signal
> sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid [BREAK]
> 2013-08-06 15:37:47.312306 [DEBUG] switch_core_state_machine.c:416
> (sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid) Running State Change
> CS_INIT
> 2013-08-06 15:37:47.312306 [DEBUG] switch_core_state_machine.c:455
> (sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid) State INIT
> 2013-08-06 15:37:47.312306 [DEBUG] mod_sofia.c:87
> sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid SOFIA INIT
> 2013-08-06 15:37:47.312306 [DEBUG] switch_core_media.c:681 Set Local Key
> [1 AES_CM_128_HMAC_SHA1_80 inline:SvDPyP7iDr8+xmeRvVRrrqHinvGB/Ht8+VArR6Ro]
> 2013-08-06 15:37:47.312306 [DEBUG] switch_core_media.c:681 Set Local Key
> [1 AES_CM_128_HMAC_SHA1_80 inline:wUFqlClOJwLZaHzE+QlEQfVQEx979Gx4e7BCbl56]
> 2013-08-06 15:37:47.312306 [DEBUG] sofia_glue.c:1191 sip:4st0031l at PUBLIC_IP:38333;transport=ws
> Setting proxy route to sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid
> 2013-08-06 15:37:47.312306 [DEBUG] sofia_glue.c:1220 Local SDP:
> v=0
> o=FreeSWITCH 1375775801 1375775802 IN IP4 PUBLIC_IP
>
> s=FreeSWITCH
> c=IN IP4 PUBLIC_IP
> t=0 0
> a=msid-semantic: WMS B54oKgHfL2pj5wroyvgO0c1ghIMukBaL
> m=audio 20466 RTP/SAVPF 8 101 13
> a=rtpmap:101 telephone-event/8000
> a=rtcp-mux
> a=rtcp:20466 IN IP4 PUBLIC_IP
> a=ssrc:3456367315 cname:w406axcLqrVlOdzc
> a=ssrc:3456367315 msid:B54oKgHfL2pj5wroyvgO0c1ghIMukBaL a0
> a=ssrc:3456367315 mslabel:B54oKgHfL2pj5wroyvgO0c1ghIMukBaL
> a=ssrc:3456367315 label:B54oKgHfL2pj5wroyvgO0c1ghIMukBaLa0
> a=ice-ufrag:zGQpJHvQg7bqzb2V
> a=ice-pwd:5SS6gUIwcuIQy7TX
> a=candidate:3772579260 1 udp 659136 PUBLIC_IP 20466 typ host generation 0
> a=candidate:3772579260 2 udp 659136 PUBLIC_IP 20466 typ host generation 0
> a=crypto:1 AES_CM_128_HMAC_SHA1_80
> inline:SvDPyP7iDr8+xmeRvVRrrqHinvGB/Ht8+VArR6Ro
> a=ptime:20
> a=sendrecv
>
>
>
> On Tue, Aug 6, 2013 at 3:26 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> try prepending the bridge url with {absolute_codec_string=PCMA}
>>
>>
>> On Tue, Aug 6, 2013 at 6:32 AM, Javier Menendez <
>> menendez.garcia at gmail.com> wrote:
>>
>>> Hi,
>>>
>>> I am trying to make an outbound call to a webrtc softphone using jssip,
>>> I initiate the call from an asterisk box :
>>>
>>> [Asterisk] -> [FS] ->[jssip]
>>>
>>> I always get an  INCOMPATIBLE_DESTINATION error, looking at the trace
>>> logs I found out that the problem is the codec negotiation but I can not
>>> make it work, AFAIK the call should use the ALAW codec as it is compatible
>>> with all legs involved. this is the asterisk SDP
>>>
>>> 2013-08-06 13:16:50.892293 [DEBUG] sofia.c:5802 Remote SDP:
>>> v=0
>>> o=root 451068671 451068671 IN IP4 192.168.90.16
>>> s=Asterisk PBX 1.8.16.0
>>> c=IN IP4 192.168.90.16
>>> t=0 0
>>> m=audio 16126 RTP/AVP 8 101
>>> a=rtpmap:8 PCMA/8000
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-16
>>> a=ptime:20
>>>
>>> And this is the SDP from FS wich is sent to the jssip client
>>>
>>> 2013-08-06 13:16:50.912323 [DEBUG] sofia_glue.c:1220 Local SDP:
>>> v=0
>>> o=FreeSWITCH 1375766038 1375766039 IN IP4 PUBLIC_IP
>>> s=FreeSWITCH
>>> c=IN IP4 PUBLIC_IP
>>> t=0 0
>>> a=msid-semantic: WMS 9hJmtW2hnx2uG5S2obryUBUKGF35tgps
>>> m=audio 21772 RTP/SAVPF 8 0 101 13
>>> a=rtpmap:101 telephone-event/8000
>>> a=rtcp-mux
>>> a=rtcp:21772 IN IP4 PUBLIC_IP
>>> a=ssrc:3456358538 cname:oZFIMxPZUIkrD3jS
>>> a=ssrc:3456358538 msid:9hJmtW2hnx2uG5S2obryUBUKGF35tgps a0
>>> a=ssrc:3456358538 mslabel:9hJmtW2hnx2uG5S2obryUBUKGF35tgps
>>> a=ssrc:3456358538 label:9hJmtW2hnx2uG5S2obryUBUKGF35tgpsa0
>>> a=ice-ufrag:4BgP20qzvjuPqK7E
>>> a=ice-pwd:ADcc0S2Lqc2T5cB0
>>> a=candidate:3263618716 1 udp 659136 212.230.135.231 21772 typ host
>>> generation 0
>>> a=candidate:3263618716 2 udp 659136 212.230.135.231 21772 typ host
>>> generation 0
>>> a=crypto:1 AES_CM_128_HMAC_SHA1_80
>>> inline:JbAOcaClBcoNLW6zjoucy5BU0mfS+UQqWcyYh9+7
>>> a=ptime:20
>>> a=sendrecv
>>>
>>> Shouldn't it include the PCMA codec? I hace tried to enable it in
>>> configuration but it doesn't work, also tried to set it with
>>>
>>>   <action application="export"
>>> data="nolocal:absolute_codec_string=PCMA,PCMU"/>
>>>
>>> before bridge, but no luck.
>>>
>>> Funny thing is that it is working if i innitiate the call from the jssip
>>> client. [jssip] -> [FS] ->[Asterisk]
>>>
>>> Any clue?
>>>
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 nneul at mst.edu  Tue Aug  6 21:16:34 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Tue, 06 Aug 2013 12:16:34 -0500
Subject: [Freeswitch-users] Do we have ESL-proxy?
In-Reply-To: <CAJUJwtgjPCgbEiJKU_iYTCbwJ26p08K7M39xF4K9TLaBZD=UTw@mail.gmail.com>
References: <CAJUJwtgjPCgbEiJKU_iYTCbwJ26p08K7M39xF4K9TLaBZD=UTw@mail.gmail.com>
Message-ID: <52012F72.7090208@mst.edu>

Are you saying "round robin" or "mirrored" for the commands?

i.e. if you send a command over this proxy, would it be duplicated to all of the FS servers, or sent to "first available"?

-- Nathan

On 08/06/2013 12:05 PM, SamyGo wrote:
> Dear Users,
>
> I have come up with a requirement to have ESL connection made with multiple FS Servers at the same time and send them
> commands(no event receiving is required for now) The connections to FS should be pooled and monitored if any FS goes down.
>
> Is there any ESLproxy available to be used readily?
>
> Thanks,
> Sammy
>

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From govoiper at gmail.com  Tue Aug  6 21:19:48 2013
From: govoiper at gmail.com (SamyGo)
Date: Tue, 6 Aug 2013 10:19:48 -0700
Subject: [Freeswitch-users] Do we have ESL-proxy?
In-Reply-To: <52012F72.7090208@mst.edu>
References: <CAJUJwtgjPCgbEiJKU_iYTCbwJ26p08K7M39xF4K9TLaBZD=UTw@mail.gmail.com>
	<52012F72.7090208@mst.edu>
Message-ID: <CAJUJwthuOJcCus8ZOqykszY93q-3VyG33xFkAgTk5CVyPjLK5Q@mail.gmail.com>

Hi Nathan,

I can only think of Round Robin with failover. I definitely don't want to
parallel fork an ESL command.

--
Sammy



On Tue, Aug 6, 2013 at 10:16 AM, Nathan Neulinger <nneul at mst.edu> wrote:

> Are you saying "round robin" or "mirrored" for the commands?
>
> i.e. if you send a command over this proxy, would it be duplicated to all
> of the FS servers, or sent to "first available"?
>
> -- Nathan
>
>
> On 08/06/2013 12:05 PM, SamyGo wrote:
>
>> Dear Users,
>>
>> I have come up with a requirement to have ESL connection made with
>> multiple FS Servers at the same time and send them
>> commands(no event receiving is required for now) The connections to FS
>> should be pooled and monitored if any FS goes down.
>>
>> Is there any ESLproxy available to be used readily?
>>
>> Thanks,
>> Sammy
>>
>>
> --
> ------------------------------**------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
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From nneul at mst.edu  Tue Aug  6 21:22:58 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Tue, 06 Aug 2013 12:22:58 -0500
Subject: [Freeswitch-users] Do we have ESL-proxy?
In-Reply-To: <CAJUJwthuOJcCus8ZOqykszY93q-3VyG33xFkAgTk5CVyPjLK5Q@mail.gmail.com>
References: <CAJUJwtgjPCgbEiJKU_iYTCbwJ26p08K7M39xF4K9TLaBZD=UTw@mail.gmail.com>
	<52012F72.7090208@mst.edu>
	<CAJUJwthuOJcCus8ZOqykszY93q-3VyG33xFkAgTk5CVyPjLK5Q@mail.gmail.com>
Message-ID: <520130F2.20106@mst.edu>

You might look at the keepalived page on wiki, that's what I use to float an IP in the case where only one of the FS 
instances should be online.

You could also use HAProxy to define a group of systems for it to talk to if you want multiple to be online at once, but 
rotating between them.

-- Nathan

On 08/06/2013 12:19 PM, SamyGo wrote:
> Hi Nathan,
>
> I can only think of Round Robin with failover. I definitely don't want to parallel fork an ESL command.
>
> --
> Sammy
>
>
>
> On Tue, Aug 6, 2013 at 10:16 AM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu>> wrote:
>
>     Are you saying "round robin" or "mirrored" for the commands?
>
>     i.e. if you send a command over this proxy, would it be duplicated to all of the FS servers, or sent to "first
>     available"?
>
>     -- Nathan
>
>
>     On 08/06/2013 12:05 PM, SamyGo wrote:
>
>         Dear Users,
>
>         I have come up with a requirement to have ESL connection made with multiple FS Servers at the same time and send
>         them
>         commands(no event receiving is required for now) The connections to FS should be pooled and monitored if any FS
>         goes down.
>
>         Is there any ESLproxy available to be used readily?
>
>         Thanks,
>         Sammy
>
>
>     --
>     ------------------------------__------------------------------
>     Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu>
>     Missouri S&T Information Technology    (573) 612-1412
>     System Administrator - Architect
>
>

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From nneul at mst.edu  Tue Aug  6 21:40:56 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Tue, 06 Aug 2013 12:40:56 -0500
Subject: [Freeswitch-users] Do we have ESL-proxy?
In-Reply-To: <CAJUJwtj5gJNx0VWnAV+Yy+fHFoMvsn9rqnQkZpfs3MT-c5GAzg@mail.gmail.com>
References: <CAJUJwtgjPCgbEiJKU_iYTCbwJ26p08K7M39xF4K9TLaBZD=UTw@mail.gmail.com>
	<52012F72.7090208@mst.edu>
	<CAJUJwthuOJcCus8ZOqykszY93q-3VyG33xFkAgTk5CVyPjLK5Q@mail.gmail.com>
	<520130F2.20106@mst.edu>
	<CAJUJwtj5gJNx0VWnAV+Yy+fHFoMvsn9rqnQkZpfs3MT-c5GAzg@mail.gmail.com>
Message-ID: <52013528.2010709@mst.edu>

With HAProxy, you could just define a list of target servers. You'd configure your app to talk to the HA IP, and then 
each request would go to next available server in the pool you have defined.

You could also simply do a DNS round robin if you wanted. All depends on what infrastructure/etc. you have available to 
you and how resilient it needs to be.

-- Nathan

On 08/06/2013 12:38 PM, SamyGo wrote:
> Dear Nathan,
>
> I think I've missed something. I don't understand how HAproxy and Floating IP and KeepAlived will help me? I need to
> have all Active FS Servers and some mechanism to send ESL commands to them in round-robin fashion. I've an autodialer
> script so that Proxy will help me use many FS Servers at the same time !
>
> Thanks,
> Sammy
>
>
>
> On Tue, Aug 6, 2013 at 10:22 AM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu>> wrote:
>
>     You might look at the keepalived page on wiki, that's what I use to float an IP in the case where only one of the FS
>     instances should be online.
>
>     You could also use HAProxy to define a group of systems for it to talk to if you want multiple to be online at once,
>     but rotating between them.
>
>     -- Nathan
>
>
>     On 08/06/2013 12:19 PM, SamyGo wrote:
>
>         Hi Nathan,
>
>         I can only think of Round Robin with failover. I definitely don't want to parallel fork an ESL command.
>
>         --
>         Sammy
>
>
>
>         On Tue, Aug 6, 2013 at 10:16 AM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu
>         <mailto:nneul at mst.edu>>> wrote:
>
>              Are you saying "round robin" or "mirrored" for the commands?
>
>              i.e. if you send a command over this proxy, would it be duplicated to all of the FS servers, or sent to "first
>              available"?
>
>              -- Nathan
>
>
>              On 08/06/2013 12:05 PM, SamyGo wrote:
>
>                  Dear Users,
>
>                  I have come up with a requirement to have ESL connection made with multiple FS Servers at the same time
>         and send
>                  them
>                  commands(no event receiving is required for now) The connections to FS should be pooled and monitored
>         if any FS
>                  goes down.
>
>                  Is there any ESLproxy available to be used readily?
>
>                  Thanks,
>                  Sammy
>
>
>              --
>              ------------------------------____----------------------------__--
>              Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu <mailto:nneul at mst.edu>>
>
>              Missouri S&T Information Technology    (573) 612-1412
>              System Administrator - Architect
>
>
>
>     --
>     ------------------------------__------------------------------
>     Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu>
>     Missouri S&T Information Technology    (573) 612-1412
>     System Administrator - Architect
>
>

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From bdfoster at davri.com  Tue Aug  6 21:41:05 2013
From: bdfoster at davri.com (Brian Foster)
Date: Tue, 6 Aug 2013 13:41:05 -0400
Subject: [Freeswitch-users] freeswitch: Syntax error: word unexpected
 (expecting ")")
In-Reply-To: <5200AEF5.3010400@aerea.de>
References: <CAHgk4BXhFoSaEyXTopMNFsDOwd=5kxFPmq5+h=e=EPbatV-mFw@mail.gmail.com>
	<CAHgk4BVhPAfJ-rdAs2pc2mG-jz86N1BsicEZThq0Y-XCWqxAWA@mail.gmail.com>
	<CAHgk4BXaY6gPjhuG-_JR65XUL3waG34V-9DpzY0+wcUTQVNF7A@mail.gmail.com>
	<5200AEF5.3010400@aerea.de>
Message-ID: <CAHgk4BVbqiz81LubGQHtH3g0h2TtrVUyZG9HBe5ZNHYVWX_Z_A@mail.gmail.com>

Thank you for the reply.

root at bdfoster-workstation:/usr/local/freeswitch/bin# file freeswitch
freeswitch: ELF 64-bit LSB executable, x86-64, version 1 (SYSV),
dynamically linked (uses shared libs), for GNU/Linux 2.6.24,
BuildID[sha1]=0x36ddbb4c4cc574a7188b0896e02526de778a5016, not stripped


Thank you,

Brian Foster
Project Manager/Owner's Representative
Davri Investments, Incorporated
P: +1-317-787-2686
M: +1-317-600-9753
Indianapolis, Indiana


On Tue, Aug 6, 2013 at 4:08 AM, S?ren Sprenger <soeren.sprenger at aerea.de>wrote:

>  Hi,
>
> what does "file /path/to/freeswitch" say which kind binary executable file
> it is? Seams your freeswitch is not binary compatible with your running
> system...
>
>
>
> On 06.08.2013 09:54, Brian Foster wrote:
>
> Fresh build on a bare metal Dell Precision 490 running Ubuntu 12.04
> 64-bit, latest v1.2.stable (5 min ago). When I try to run freeswitch, I get:
>
> freeswitch:1 freeswitch: Syntax error: word unexpected (expecting ")")
>
> I've since deleted everything in /usr/local/freeswitch and
> /usr/local/src/freeswitch, and ran the install again. Still have the same
> outcome.
>
> The same version was built on another server, no issues. What could be the
> problem? Anything I can do to mitigate the issue?
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Representative
> Davri Investments, Incorporated
> P: +1-317-787-2686
> M: +1-317-600-9753
> Indianapolis, Indiana
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>
>
>
> --
> AereA NetworX UG (haftungsbeschr?nkt)
> Im Camisch 8
> 07768 Kahla
> Gesch?ftsf?hrerin: Franziska Sprenger
> Handelsregister: Jena B504724
> E-Mail: info at aerea.de
> Web: http://www.aerea.de
> Fon: +49 (0) 36424 760823
> Fax: +49 (0) 36651 1390009
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 govoiper at gmail.com  Tue Aug  6 21:57:48 2013
From: govoiper at gmail.com (SamyGo)
Date: Tue, 6 Aug 2013 10:57:48 -0700
Subject: [Freeswitch-users] Do we have ESL-proxy?
In-Reply-To: <52013528.2010709@mst.edu>
References: <CAJUJwtgjPCgbEiJKU_iYTCbwJ26p08K7M39xF4K9TLaBZD=UTw@mail.gmail.com>
	<52012F72.7090208@mst.edu>
	<CAJUJwthuOJcCus8ZOqykszY93q-3VyG33xFkAgTk5CVyPjLK5Q@mail.gmail.com>
	<520130F2.20106@mst.edu>
	<CAJUJwtj5gJNx0VWnAV+Yy+fHFoMvsn9rqnQkZpfs3MT-c5GAzg@mail.gmail.com>
	<52013528.2010709@mst.edu>
Message-ID: <CAJUJwtgdQb=aZXaxTiSeOvhN0w+A-HYfKZqNk5UnKtPn=G=piA@mail.gmail.com>

Dear Nathan,
Can you refer me some relevant links, this seems more complex solution. I
was thinking more like AMIproxy/ AstmanProxy thing.

BR,
Sammy



On Tue, Aug 6, 2013 at 10:40 AM, Nathan Neulinger <nneul at mst.edu> wrote:

> With HAProxy, you could just define a list of target servers. You'd
> configure your app to talk to the HA IP, and then each request would go to
> next available server in the pool you have defined.
>
> You could also simply do a DNS round robin if you wanted. All depends on
> what infrastructure/etc. you have available to you and how resilient it
> needs to be.
>
> -- Nathan
>
>
> On 08/06/2013 12:38 PM, SamyGo wrote:
>
>> Dear Nathan,
>>
>> I think I've missed something. I don't understand how HAproxy and
>> Floating IP and KeepAlived will help me? I need to
>> have all Active FS Servers and some mechanism to send ESL commands to
>> them in round-robin fashion. I've an autodialer
>> script so that Proxy will help me use many FS Servers at the same time !
>>
>> Thanks,
>> Sammy
>>
>>
>>
>> On Tue, Aug 6, 2013 at 10:22 AM, Nathan Neulinger <nneul at mst.edu <mailto:
>> nneul at mst.edu>> wrote:
>>
>>     You might look at the keepalived page on wiki, that's what I use to
>> float an IP in the case where only one of the FS
>>     instances should be online.
>>
>>     You could also use HAProxy to define a group of systems for it to
>> talk to if you want multiple to be online at once,
>>     but rotating between them.
>>
>>     -- Nathan
>>
>>
>>     On 08/06/2013 12:19 PM, SamyGo wrote:
>>
>>         Hi Nathan,
>>
>>         I can only think of Round Robin with failover. I definitely don't
>> want to parallel fork an ESL command.
>>
>>         --
>>         Sammy
>>
>>
>>
>>         On Tue, Aug 6, 2013 at 10:16 AM, Nathan Neulinger <nneul at mst.edu<mailto:
>> nneul at mst.edu> <mailto:nneul at mst.edu
>>
>>         <mailto:nneul at mst.edu>>> wrote:
>>
>>              Are you saying "round robin" or "mirrored" for the commands?
>>
>>              i.e. if you send a command over this proxy, would it be
>> duplicated to all of the FS servers, or sent to "first
>>              available"?
>>
>>              -- Nathan
>>
>>
>>              On 08/06/2013 12:05 PM, SamyGo wrote:
>>
>>                  Dear Users,
>>
>>                  I have come up with a requirement to have ESL connection
>> made with multiple FS Servers at the same time
>>         and send
>>                  them
>>                  commands(no event receiving is required for now) The
>> connections to FS should be pooled and monitored
>>         if any FS
>>                  goes down.
>>
>>                  Is there any ESLproxy available to be used readily?
>>
>>                  Thanks,
>>                  Sammy
>>
>>
>>              --
>>              ------------------------------**
>> ____--------------------------**--__--
>>              Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu>
>> <mailto:nneul at mst.edu <mailto:nneul at mst.edu>>
>>
>>
>>              Missouri S&T Information Technology    (573) 612-1412
>>              System Administrator - Architect
>>
>>
>>
>>     --
>>     ------------------------------**__----------------------------**--
>>     Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu>
>>     Missouri S&T Information Technology    (573) 612-1412
>>     System Administrator - Architect
>>
>>
>>
> --
> ------------------------------**------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
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From yehavi.bourvine at gmail.com  Tue Aug  6 21:59:38 2013
From: yehavi.bourvine at gmail.com (Yehavi Bourvine)
Date: Tue, 6 Aug 2013 20:59:38 +0300
Subject: [Freeswitch-users] A few problems with latest GIT
In-Reply-To: <CAKbxfG-jjkHY2xz1j4AM2GyenRz0nYsn1inNxnm=2LyXYoWC3A@mail.gmail.com>
References: <CAAtWKeEW+ngi=7xq=0LOuYSPg4_JUwzFjqDBMMheskDy40bFvg@mail.gmail.com>
	<BE4C37BE-3FDC-4E12-8718-CD8607A8AC81@kavun.ch>
	<CAKbxfG-jjkHY2xz1j4AM2GyenRz0nYsn1inNxnm=2LyXYoWC3A@mail.gmail.com>
Message-ID: <CAAtWKeFnvCpMcNLVMQzRgeEGGnyTM3eA03ECwArPxU0KOFb4rA@mail.gmail.com>

Hello Anthony,

  Every time I update I do a fresh git checkout. I cannot use the default
config as this is a production system.

I'll add debug information to my LUA/JS scripts to see whether one of them
gets stuck.

When I get again into the situation of sessions build-up, what do you
suggest to do then in order to collect information that can help? I cannot
repsroduce this at will...

               Thanks! __Yehavi:



2013/8/6 Anthony Minessale <anthony.minessale at gmail.com>

> If you can find a way to reproduce the left over sessions problem, its the
> only way to find the cause.  Its not something that happens by default.  It
> could be related to something specific you are doing like leaving a lua
> script in an endless loop or a bad build.
>
> Try a full clean build of a fresh git checkout against the default configs
> and see if you can still reproduce your problem and post the details to
> Jira.
>
>
>
> On Tue, Aug 6, 2013 at 3:37 AM, Emrah <lists at kavun.ch> wrote:
>
>> I confirm the last issue of your list has also happened on my latest GIT.
>> Your no audio syndrome sounds like it could be connected to my Polycom
>> bridging issue.
>> As for the sluggishnness of fs_cli I haven't noticed anything or just
>> associated it to my mobile broadband?
>> On Aug 6, 2013, at 1:02 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
>> wrote:
>>
>> Hello,
>>
>>   I am running FS version 1.0.x and trying to move to 1.2/1.4/latest-git.
>> I encounter a few problems which prevents me from using these versions, and
>> would like to know whether someone else is having it. It is quite hard to
>> create debugging information, as with full logging the behaviour is
>> slightly different than lower logging level. The problems appear only on my
>> production system, around noon (when the number of active sessins goes
>> beyond 50-60 sessions):
>>
>>
>>    - The voice path is lost in the middle of the session (Jira FS-5616).
>>    From TCPdump I see that FS receive the audio from both sides but stops
>>    sending it; there is nothing meaningfull in the logs.
>>    - FS starts to get sluggish, fs_cli is stuck, SIP messages are
>>    processed in delay (5-30 seconds). This goes away by itself after a few
>>    minutes.
>>    - sessions are left "in use" after the users hangup, and the number
>>    of active sessions starts rising. Doing uuid_kill succeeds once (next times
>>    fail with "no such uuid", but "show channels" still shows it.
>>
>> The most frustrating side is that I do not know what exact debugging
>> tools to use in order to submit meaningfull data to FS developers.
>>
>>                      Thanks! __Yehavi:
>>  _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nneul at mst.edu  Tue Aug  6 22:00:37 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Tue, 06 Aug 2013 13:00:37 -0500
Subject: [Freeswitch-users] Do we have ESL-proxy?
In-Reply-To: <CAJUJwtgdQb=aZXaxTiSeOvhN0w+A-HYfKZqNk5UnKtPn=G=piA@mail.gmail.com>
References: <CAJUJwtgjPCgbEiJKU_iYTCbwJ26p08K7M39xF4K9TLaBZD=UTw@mail.gmail.com>
	<52012F72.7090208@mst.edu>
	<CAJUJwthuOJcCus8ZOqykszY93q-3VyG33xFkAgTk5CVyPjLK5Q@mail.gmail.com>
	<520130F2.20106@mst.edu>
	<CAJUJwtj5gJNx0VWnAV+Yy+fHFoMvsn9rqnQkZpfs3MT-c5GAzg@mail.gmail.com>
	<52013528.2010709@mst.edu>
	<CAJUJwtgdQb=aZXaxTiSeOvhN0w+A-HYfKZqNk5UnKtPn=G=piA@mail.gmail.com>
Message-ID: <520139C5.1070608@mst.edu>

ESL is just a simple TCP socket connection... if all you're wanting to do is rotate between them, there are LOTs of 
different solutions. If you're looking for load balancing/calculated assignment of which one to use/etc. then you're 
going to need something more advanced.

I'm not aware of any high-level proxy services that do the sort of things described in amiproxy/astmanproxy, but someone 
else may be aware of something like it.

-- Nathan

On 08/06/2013 12:57 PM, SamyGo wrote:
> Dear Nathan,
> Can you refer me some relevant links, this seems more complex solution. I was thinking more like AMIproxy/ AstmanProxy
> thing.
>
> BR,
> Sammy
>
>
>
> On Tue, Aug 6, 2013 at 10:40 AM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu>> wrote:
>
>     With HAProxy, you could just define a list of target servers. You'd configure your app to talk to the HA IP, and
>     then each request would go to next available server in the pool you have defined.
>
>     You could also simply do a DNS round robin if you wanted. All depends on what infrastructure/etc. you have available
>     to you and how resilient it needs to be.
>
>     -- Nathan
>
>
>     On 08/06/2013 12:38 PM, SamyGo wrote:
>
>         Dear Nathan,
>
>         I think I've missed something. I don't understand how HAproxy and Floating IP and KeepAlived will help me? I need to
>         have all Active FS Servers and some mechanism to send ESL commands to them in round-robin fashion. I've an
>         autodialer
>         script so that Proxy will help me use many FS Servers at the same time !
>
>         Thanks,
>         Sammy
>
>
>
>         On Tue, Aug 6, 2013 at 10:22 AM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu
>         <mailto:nneul at mst.edu>>> wrote:
>
>              You might look at the keepalived page on wiki, that's what I use to float an IP in the case where only one
>         of the FS
>              instances should be online.
>
>              You could also use HAProxy to define a group of systems for it to talk to if you want multiple to be online
>         at once,
>              but rotating between them.
>
>              -- Nathan
>
>
>              On 08/06/2013 12:19 PM, SamyGo wrote:
>
>                  Hi Nathan,
>
>                  I can only think of Round Robin with failover. I definitely don't want to parallel fork an ESL command.
>
>                  --
>                  Sammy
>
>
>
>                  On Tue, Aug 6, 2013 at 10:16 AM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu>
>         <mailto:nneul at mst.edu <mailto:nneul at mst.edu>> <mailto:nneul at mst.edu <mailto:nneul at mst.edu>
>
>                  <mailto:nneul at mst.edu <mailto:nneul at mst.edu>>>> wrote:
>
>                       Are you saying "round robin" or "mirrored" for the commands?
>
>                       i.e. if you send a command over this proxy, would it be duplicated to all of the FS servers, or
>         sent to "first
>                       available"?
>
>                       -- Nathan
>
>
>                       On 08/06/2013 12:05 PM, SamyGo wrote:
>
>                           Dear Users,
>
>                           I have come up with a requirement to have ESL connection made with multiple FS Servers at the
>         same time
>                  and send
>                           them
>                           commands(no event receiving is required for now) The connections to FS should be pooled and
>         monitored
>                  if any FS
>                           goes down.
>
>                           Is there any ESLproxy available to be used readily?
>
>                           Thanks,
>                           Sammy
>
>
>                       --
>                       ------------------------------______--------------------------__--__--
>                       Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu
>         <mailto:nneul at mst.edu>> <mailto:nneul at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu <mailto:nneul at mst.edu>>>
>
>
>                       Missouri S&T Information Technology    (573) 612-1412
>                       System Administrator - Architect
>
>
>
>              --
>              ------------------------------____----------------------------__--
>              Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu <mailto:nneul at mst.edu>>
>              Missouri S&T Information Technology    (573) 612-1412
>              System Administrator - Architect
>
>
>
>     --
>     ------------------------------__------------------------------
>     Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu>
>     Missouri S&T Information Technology    (573) 612-1412
>     System Administrator - Architect
>
>

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From anthony.minessale at gmail.com  Tue Aug  6 22:40:40 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 6 Aug 2013 13:40:40 -0500
Subject: [Freeswitch-users] A few problems with latest GIT
In-Reply-To: <CAAtWKeFnvCpMcNLVMQzRgeEGGnyTM3eA03ECwArPxU0KOFb4rA@mail.gmail.com>
References: <CAAtWKeEW+ngi=7xq=0LOuYSPg4_JUwzFjqDBMMheskDy40bFvg@mail.gmail.com>
	<BE4C37BE-3FDC-4E12-8718-CD8607A8AC81@kavun.ch>
	<CAKbxfG-jjkHY2xz1j4AM2GyenRz0nYsn1inNxnm=2LyXYoWC3A@mail.gmail.com>
	<CAAtWKeFnvCpMcNLVMQzRgeEGGnyTM3eA03ECwArPxU0KOFb4rA@mail.gmail.com>
Message-ID: <CAKbxfG_S0YHpzrHe2T2KDH9+SswaQN9sUfQXjoz0cZ3UA_CEAQ@mail.gmail.com>

Get a core dump with gcore and do a full backrace on all threads with the
command..... thread apply all bt
On Aug 6, 2013 1:07 PM, "Yehavi Bourvine" <yehavi.bourvine at gmail.com> wrote:

> Hello Anthony,
>
>   Every time I update I do a fresh git checkout. I cannot use the default
> config as this is a production system.
>
> I'll add debug information to my LUA/JS scripts to see whether one of them
> gets stuck.
>
> When I get again into the situation of sessions build-up, what do you
> suggest to do then in order to collect information that can help? I cannot
> repsroduce this at will...
>
>                Thanks! __Yehavi:
>
>
>
> 2013/8/6 Anthony Minessale <anthony.minessale at gmail.com>
>
>> If you can find a way to reproduce the left over sessions problem, its
>> the only way to find the cause.  Its not something that happens by default.
>>  It could be related to something specific you are doing like leaving a lua
>> script in an endless loop or a bad build.
>>
>> Try a full clean build of a fresh git checkout against the default
>> configs and see if you can still reproduce your problem and post the
>> details to Jira.
>>
>>
>>
>> On Tue, Aug 6, 2013 at 3:37 AM, Emrah <lists at kavun.ch> wrote:
>>
>>> I confirm the last issue of your list has also happened on my latest GIT.
>>> Your no audio syndrome sounds like it could be connected to my Polycom
>>> bridging issue.
>>> As for the sluggishnness of fs_cli I haven't noticed anything or just
>>> associated it to my mobile broadband?
>>> On Aug 6, 2013, at 1:02 AM, Yehavi Bourvine <yehavi.bourvine at gmail.com>
>>> wrote:
>>>
>>> Hello,
>>>
>>>   I am running FS version 1.0.x and trying to move to
>>> 1.2/1.4/latest-git. I encounter a few problems which prevents me from using
>>> these versions, and would like to know whether someone else is having it.
>>> It is quite hard to create debugging information, as with full logging the
>>> behaviour is slightly different than lower logging level. The problems
>>> appear only on my production system, around noon (when the number of active
>>> sessins goes beyond 50-60 sessions):
>>>
>>>
>>>    - The voice path is lost in the middle of the session (Jira
>>>    FS-5616). From TCPdump I see that FS receive the audio from both sides but
>>>    stops sending it; there is nothing meaningfull in the logs.
>>>    - FS starts to get sluggish, fs_cli is stuck, SIP messages are
>>>    processed in delay (5-30 seconds). This goes away by itself after a few
>>>    minutes.
>>>    - sessions are left "in use" after the users hangup, and the number
>>>    of active sessions starts rising. Doing uuid_kill succeeds once (next times
>>>    fail with "no such uuid", but "show channels" still shows it.
>>>
>>> The most frustrating side is that I do not know what exact debugging
>>> tools to use in order to submit meaningfull data to FS developers.
>>>
>>>                      Thanks! __Yehavi:
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>
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From ivan at c3i.bg  Tue Aug  6 21:19:57 2013
From: ivan at c3i.bg (Ivan)
Date: Tue, 06 Aug 2013 20:19:57 +0300
Subject: [Freeswitch-users] Can not make out going call from sangoma A200
In-Reply-To: <BLU171-W488ED06FE9240BE8CD3149B55D0@phx.gbl>
References: <BLU171-W3672C8625546F122FFC013B5510@phx.gbl>,
	<CAHgk4BXX1nArMXv-T5CNzLbjh_wyn0fjzYNOUCaAPsUE14KNdA@mail.gmail.com>,
	<BLU171-W58FF5C141AECC333E5BD0FB5520@phx.gbl>,
	<CAHgk4BVdu6Dk3P=rrRBKquiydLUUGBUxSHauTv73tx=XPq=pjA@mail.gmail.com>
	<BLU171-W488ED06FE9240BE8CD3149B55D0@phx.gbl>
Message-ID: <5201303D.3070405@c3i.bg>

WRT the callerid issue, do you answer the call immediately ? If yes you 
should try to wait 2 or 3 rings before answering, since the callerid 
data is usually sent at the first ring.
There might also be a problem with your provider, did you check first 
that you get the callerid when plugging a telephone to your provider's 
line ?

Outgoing call problem: please post more debug from fs_cli (before the 
hangup).

ivan


On 08/06/2013 09:42 AM, James zhu wrote:
> hello:
> I contacted Sangoma, but I do not have any feedback yet. I think it
> might be a bug
> for analog card with freeSWITCH.  freetdm  partially works with sangoma
> A200.
> There are two issues:
> 1) Can not get callerid from IO:  only shows this:Initializing cid data!
> the callerid is empty
> even I set to CN.
> 2) Can not make outgoing calls by the port that I used to make outgoing
> calls. even I replaced a FXO module, the problem is same. I use same
> syntax as follow:
> http://blog.hiastar.com/?p=276
> Hope someone from the FreeSWITCH community to clarify that.
>
> Best regards,
> James.zhu
> website: www.hiastar.com
>
> ------------------------------------------------------------------------
> Date: Fri, 2 Aug 2013 23:03:47 -0400
> From: bdfoster at davri.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Can not make out going call from sangoma
> A200
>
> I apologize, I misread your email. Hopefully someone who knows FreeTDM
> will come around. Have you contacted Sangoma? They are the maintainers
> of FreeTDM as far as I know. You might try that route. They are very
> familiar with freeswitch im sure :).
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com <mailto:bdfoster at davri.com>
> Indianapolis, Indiana
>
> Sent from a mobile device.
>
> On Aug 2, 2013 10:39 PM, "James zhu" <zhulizhong at live.com
> <mailto:zhulizhong at live.com>> wrote:
>
>     thanks, Brian.  actually the 2:1 is idle because the port can accept
>     incoming call. after
>     I reload freetdm or restart FS, I still can not make outgoing calls.
>     If the port physically
>     failed, how come I can make incoming call use the same port.
>     confused. thanks again.
>
>     Best regards,
>     James.zhu
>     website: www.hiastar.com <http://www.hiastar.com>
>
>     ------------------------------------------------------------------------
>     Date: Fri, 2 Aug 2013 14:22:29 -0400
>     From: bdfoster at davri.com <mailto:bdfoster at davri.com>
>     To: freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     Subject: Re: [Freeswitch-users] Can not make out going call from
>     sangoma A200
>
>     That's the expected behavior if 2:1 is in use. So that means you use
>     another port to place your outbound call. Tips on how to do that,
>     however, I can't really produce. Hopefully someone with more
>     experience with FreeTDM can chime in.
>
>     Thank you,
>
>     Brian Foster
>     Project Manager/Owner's Rep.
>     Davri Investments, Inc.
>     O: 317-787-2686 x2102
>     M: 317-600-9753
>     E: bdfoster at davri.com <mailto:bdfoster at davri.com>
>     Indianapolis, Indiana
>
>     Sent from a mobile device.
>
>     On Aug 2, 2013 3:09 AM, "James zhu" <zhulizhong at live.com
>     <mailto:zhulizhong at live.com>> wrote:
>
>         hello:
>         I installed freeswitch and sangoma A200 with 2 FXOs. i set the
>         tone group to CN
>         and opermode=china. the A200 can make accept incoming calls from
>         FreeTDM/2:1 , but I use same port to make out going port, the
>         shows the port is CONGESTION.
>         --------------------------system
>         log-----------------------------------------------
>         2013-08-02 14:36:42.003531 [DEBUG] mod_freetdm.c:640 [2:1]
>         FreeTDM/2:1/13811737925 CHANNEL HANGUP EXIT
>         2013-08-02 14:36:42.003531 [DEBUG]
>         switch_core_state_machine.c:48 FreeTDM/2:1/13811737925 Standard
>         HANGUP, cause: NORMAL_CIRCUIT_CONGESTION
>         2013-08-02 14:36:42.003531 [DEBUG]
>         switch_core_state_machine.c:679 (FreeTDM/2:1/13811737925) State
>         HANGUP going to sleep
>         2013-08-02 14:36:42.003531 [DEBUG]
>         switch_core_state_machine.c:692 (FreeTDM/2:1/13811737925)
>         Callstate Change DOWN -> HANGUP
>         2013-08-02 14:36:42.003531 [DEBUG]
>         switch_core_state_machine.c:447 (FreeTDM/2:1/13811737925) State
>         Change CS_HANGUP -> CS_REPORTING
>         2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341
>         Send signal FreeTDM/2:1/13811737925 [BREAK]
>         2013-08-02 14:36:42.003531 [DEBUG]
>         switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925)
>         Running State Change CS_REPORTING
>         2013-08-02 14:36:42.003531 [DEBUG]
>         switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State
>         REPORTING
>         2013-08-02 14:36:42.003531 [DEBUG]
>         switch_core_state_machine.c:92 FreeTDM/2:1/13811737925 Standard
>         REPORTING, cause: NORMAL_CIRCUIT_CONGESTION
>         2013-08-02 14:36:42.003531 [DEBUG]
>         switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State
>         REPORTING going to sleep
>         2013-08-02 14:36:42.003531 [DEBUG]
>         switch_core_state_machine.c:441 (FreeTDM/2:1/13811737925) State
>         Change CS_REPORTING -> CS_DESTROY
>         2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341
>         Send signal FreeTDM/2:1/13811737925 [BREAK]
>         2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1549
>         Session 37 (FreeTDM/2:1/13811737925) Locked, Waiting on external
>         entities
>         2013-08-02 14:36:42.023531 [DEBUG] switch_ivr_originate.c:3632
>         Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>         2013-08-02 14:36:42.023531 [INFO] mod_dptools.c:3192 Originate
>         Failed.  Cause: NORMAL_CIRCUIT_CONGESTION
>         2013-08-02 14:36:42.023531 [NOTICE] switch_channel.c:4593 Hangup
>         sofia/internal/1008 at 192.168.0.173 <mailto:1008 at 192.168.0.173>
>         [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]
>         2013-08-02 14:36:42.023531 [DEBUG] switch_channel.c:3135 Send
>         signal sofia/internal/1008 at 192.168.0.173
>         <mailto:1008 at 192.168.0.173> [KILL]
>         2013-08-02 14:36:42.023531 [DEBUG] switch_core_session.c:1341
>         Send signal sofia/internal
>         ---------------------------------------------
>         the FS is download from git and wanpipe is latest version.
>           please give a help for that issue.
>
>         Best regards,
>         James.zhu
>         website: www.hiastar.com <http://www.hiastar.com>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.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
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
>     CudaTel Communication Server  Official
>     FreeSWITCH Sites http://www.freeswitch.org
>     http://wiki.freeswitch.org http://www.cluecon.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
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
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>     http://www.cluecon.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
>
>
> _________________________________________________________________________ Professional
> FreeSWITCH Consulting Services: consulting at freeswitch.org
> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
> CudaTel Communication Server  Official FreeSWITCH
> Sites http://www.freeswitch.org http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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>


From dujinfang at gmail.com  Tue Aug  6 22:57:48 2013
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 7 Aug 2013 02:57:48 +0800
Subject: [Freeswitch-users] Do we have ESL-proxy?
In-Reply-To: <520139C5.1070608@mst.edu>
References: <CAJUJwtgjPCgbEiJKU_iYTCbwJ26p08K7M39xF4K9TLaBZD=UTw@mail.gmail.com>
	<52012F72.7090208@mst.edu>
	<CAJUJwthuOJcCus8ZOqykszY93q-3VyG33xFkAgTk5CVyPjLK5Q@mail.gmail.com>
	<520130F2.20106@mst.edu>
	<CAJUJwtj5gJNx0VWnAV+Yy+fHFoMvsn9rqnQkZpfs3MT-c5GAzg@mail.gmail.com>
	<52013528.2010709@mst.edu>
	<CAJUJwtgdQb=aZXaxTiSeOvhN0w+A-HYfKZqNk5UnKtPn=G=piA@mail.gmail.com>
	<520139C5.1070608@mst.edu>
Message-ID: <C1A145048BD047698A20C905C5FC194C@gmail.com>

I was actually thinking how to do that. Raw idea would be run a daemon that connects to multiple FS instances and routes ESL requests from a single client accordingly.

FS has core-uuid in events so it can tell which FS it comes from, and api and sendmsg might need to be extended to add the core-uuid so the proxy nows which FS it needs to route to.

Is this what are we talking about? I might do this in Erlang.

-- 
Seven Du
http://www.freeswitch.org.cn
http://about.me/dujinfang
http://www.dujinfang.com


Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Wednesday, August 7, 2013 at 2:00 AM, Nathan Neulinger wrote:

> ESL is just a simple TCP socket connection... if all you're wanting to do is rotate between them, there are LOTs of 
> different solutions. If you're looking for load balancing/calculated assignment of which one to use/etc. then you're 
> going to need something more advanced.
> 
> I'm not aware of any high-level proxy services that do the sort of things described in amiproxy/astmanproxy, but someone 
> else may be aware of something like it.
> 
> -- Nathan
> 
> On 08/06/2013 12:57 PM, SamyGo wrote:
> > Dear Nathan,
> > Can you refer me some relevant links, this seems more complex solution. I was thinking more like AMIproxy/ AstmanProxy
> > thing.
> > 
> > BR,
> > Sammy
> > 
> > 
> > 
> > On Tue, Aug 6, 2013 at 10:40 AM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu>> wrote:
> > 
> > With HAProxy, you could just define a list of target servers. You'd configure your app to talk to the HA IP, and
> > then each request would go to next available server in the pool you have defined.
> > 
> > You could also simply do a DNS round robin if you wanted. All depends on what infrastructure/etc. you have available
> > to you and how resilient it needs to be.
> > 
> > -- Nathan
> > 
> > 
> > On 08/06/2013 12:38 PM, SamyGo wrote:
> > 
> > Dear Nathan,
> > 
> > I think I've missed something. I don't understand how HAproxy and Floating IP and KeepAlived will help me? I need to
> > have all Active FS Servers and some mechanism to send ESL commands to them in round-robin fashion. I've an
> > autodialer
> > script so that Proxy will help me use many FS Servers at the same time !
> > 
> > Thanks,
> > Sammy
> > 
> > 
> > 
> > On Tue, Aug 6, 2013 at 10:22 AM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu
> > <mailto:nneul at mst.edu>>> wrote:
> > 
> > You might look at the keepalived page on wiki, that's what I use to float an IP in the case where only one
> > of the FS
> > instances should be online.
> > 
> > You could also use HAProxy to define a group of systems for it to talk to if you want multiple to be online
> > at once,
> > but rotating between them.
> > 
> > -- Nathan
> > 
> > 
> > On 08/06/2013 12:19 PM, SamyGo wrote:
> > 
> > Hi Nathan,
> > 
> > I can only think of Round Robin with failover. I definitely don't want to parallel fork an ESL command.
> > 
> > --
> > Sammy
> > 
> > 
> > 
> > On Tue, Aug 6, 2013 at 10:16 AM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu>
> > <mailto:nneul at mst.edu <mailto:nneul at mst.edu>> <mailto:nneul at mst.edu <mailto:nneul at mst.edu>
> > 
> > <mailto:nneul at mst.edu <mailto:nneul at mst.edu>>>> wrote:
> > 
> > Are you saying "round robin" or "mirrored" for the commands?
> > 
> > i.e. if you send a command over this proxy, would it be duplicated to all of the FS servers, or
> > sent to "first
> > available"?
> > 
> > -- Nathan
> > 
> > 
> > On 08/06/2013 12:05 PM, SamyGo wrote:
> > 
> > Dear Users,
> > 
> > I have come up with a requirement to have ESL connection made with multiple FS Servers at the
> > same time
> > and send
> > them
> > commands(no event receiving is required for now) The connections to FS should be pooled and
> > monitored
> > if any FS
> > goes down.
> > 
> > Is there any ESLproxy available to be used readily?
> > 
> > Thanks,
> > Sammy
> > 
> > 
> > --
> > ------------------------------______--------------------------__--__--
> > Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu
> > <mailto:nneul at mst.edu>> <mailto:nneul at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu <mailto:nneul at mst.edu>>>
> > 
> > 
> > Missouri S&T Information Technology (573) 612-1412
> > System Administrator - Architect
> > 
> > 
> > 
> > --
> > ------------------------------____----------------------------__--
> > Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu <mailto:nneul at mst.edu>>
> > Missouri S&T Information Technology (573) 612-1412
> > System Administrator - Architect
> > 
> > 
> > 
> > --
> > ------------------------------__------------------------------
> > Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu>
> > Missouri S&T Information Technology (573) 612-1412
> > System Administrator - Architect
> > 
> 
> 
> -- 
> ------------------------------------------------------------
> Nathan Neulinger nneul at mst.edu (mailto:nneul at mst.edu)
> Missouri S&T Information Technology (573) 612-1412
> System Administrator - Architect
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 


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From ccachor at gmail.com  Tue Aug  6 23:09:15 2013
From: ccachor at gmail.com (Chris Cachor)
Date: Tue, 6 Aug 2013 14:09:15 -0500
Subject: [Freeswitch-users] Do we have ESL-proxy?
In-Reply-To: <mailman.21987.1375815503.576.freeswitch-users@lists.freeswitch.org>
References: <mailman.21987.1375815503.576.freeswitch-users@lists.freeswitch.org>
Message-ID: <FA9F9ADD-567C-47DA-94E2-A2C6269C27E2@gmail.com>

I'm working on a similar setup except that I'm having Freeswitch make a Curl request to a Node.js script with the IP to connect to, and the Node.js script will listen to the incoming request then open a ESL inbound socket connection to the box that requested it. The Node.js script will handle the multiple connection instances (and won't try to connect if it's connected to the machine already). Note: The Node script runs as a daemon. It's a solution that scales with minimal config changes as well.

- Chris

On Aug 6, 2013, at 1:58 PM, freeswitch-users-request at lists.freeswitch.org wrote:

> Send FreeSWITCH-users mailing list submissions to
> 	freeswitch-users at lists.freeswitch.org
> 
> To subscribe or unsubscribe via the World Wide Web, visit
> 	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
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> 
> You can reach the person managing the list at
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> 
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
> Today's Topics:
> 
>   1. Re: Can not make out going call from sangoma A200 (Ivan)
>   2. Re: Do we have ESL-proxy? (Seven Du)
> 
> From: Ivan <ivan at c3i.bg>
> Subject: Re: [Freeswitch-users] Can not make out going call from sangoma A200
> Date: August 6, 2013 12:19:57 PM CDT
> To: freeswitch-users at lists.freeswitch.org
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> WRT the callerid issue, do you answer the call immediately ? If yes you should try to wait 2 or 3 rings before answering, since the callerid data is usually sent at the first ring.
> There might also be a problem with your provider, did you check first that you get the callerid when plugging a telephone to your provider's line ?
> 
> Outgoing call problem: please post more debug from fs_cli (before the hangup).
> 
> ivan
> 
> 
> On 08/06/2013 09:42 AM, James zhu wrote:
>> hello:
>> I contacted Sangoma, but I do not have any feedback yet. I think it
>> might be a bug
>> for analog card with freeSWITCH.  freetdm  partially works with sangoma
>> A200.
>> There are two issues:
>> 1) Can not get callerid from IO:  only shows this:Initializing cid data!
>> the callerid is empty
>> even I set to CN.
>> 2) Can not make outgoing calls by the port that I used to make outgoing
>> calls. even I replaced a FXO module, the problem is same. I use same
>> syntax as follow:
>> http://blog.hiastar.com/?p=276
>> Hope someone from the FreeSWITCH community to clarify that.
>> 
>> Best regards,
>> James.zhu
>> website: www.hiastar.com
>> 
>> ------------------------------------------------------------------------
>> Date: Fri, 2 Aug 2013 23:03:47 -0400
>> From: bdfoster at davri.com
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Can not make out going call from sangoma
>> A200
>> 
>> I apologize, I misread your email. Hopefully someone who knows FreeTDM
>> will come around. Have you contacted Sangoma? They are the maintainers
>> of FreeTDM as far as I know. You might try that route. They are very
>> familiar with freeswitch im sure :).
>> 
>> Thank you,
>> 
>> Brian Foster
>> Project Manager/Owner's Rep.
>> Davri Investments, Inc.
>> O: 317-787-2686 x2102
>> M: 317-600-9753
>> E: bdfoster at davri.com <mailto:bdfoster at davri.com>
>> Indianapolis, Indiana
>> 
>> Sent from a mobile device.
>> 
>> On Aug 2, 2013 10:39 PM, "James zhu" <zhulizhong at live.com
>> <mailto:zhulizhong at live.com>> wrote:
>> 
>>    thanks, Brian.  actually the 2:1 is idle because the port can accept
>>    incoming call. after
>>    I reload freetdm or restart FS, I still can not make outgoing calls.
>>    If the port physically
>>    failed, how come I can make incoming call use the same port.
>>    confused. thanks again.
>> 
>>    Best regards,
>>    James.zhu
>>    website: www.hiastar.com <http://www.hiastar.com>
>> 
>>    ------------------------------------------------------------------------
>>    Date: Fri, 2 Aug 2013 14:22:29 -0400
>>    From: bdfoster at davri.com <mailto:bdfoster at davri.com>
>>    To: freeswitch-users at lists.freeswitch.org
>>    <mailto:freeswitch-users at lists.freeswitch.org>
>>    Subject: Re: [Freeswitch-users] Can not make out going call from
>>    sangoma A200
>> 
>>    That's the expected behavior if 2:1 is in use. So that means you use
>>    another port to place your outbound call. Tips on how to do that,
>>    however, I can't really produce. Hopefully someone with more
>>    experience with FreeTDM can chime in.
>> 
>>    Thank you,
>> 
>>    Brian Foster
>>    Project Manager/Owner's Rep.
>>    Davri Investments, Inc.
>>    O: 317-787-2686 x2102
>>    M: 317-600-9753
>>    E: bdfoster at davri.com <mailto:bdfoster at davri.com>
>>    Indianapolis, Indiana
>> 
>>    Sent from a mobile device.
>> 
>>    On Aug 2, 2013 3:09 AM, "James zhu" <zhulizhong at live.com
>>    <mailto:zhulizhong at live.com>> wrote:
>> 
>>        hello:
>>        I installed freeswitch and sangoma A200 with 2 FXOs. i set the
>>        tone group to CN
>>        and opermode=china. the A200 can make accept incoming calls from
>>        FreeTDM/2:1 , but I use same port to make out going port, the
>>        shows the port is CONGESTION.
>>        --------------------------system
>>        log-----------------------------------------------
>>        2013-08-02 14:36:42.003531 [DEBUG] mod_freetdm.c:640 [2:1]
>>        FreeTDM/2:1/13811737925 CHANNEL HANGUP EXIT
>>        2013-08-02 14:36:42.003531 [DEBUG]
>>        switch_core_state_machine.c:48 FreeTDM/2:1/13811737925 Standard
>>        HANGUP, cause: NORMAL_CIRCUIT_CONGESTION
>>        2013-08-02 14:36:42.003531 [DEBUG]
>>        switch_core_state_machine.c:679 (FreeTDM/2:1/13811737925) State
>>        HANGUP going to sleep
>>        2013-08-02 14:36:42.003531 [DEBUG]
>>        switch_core_state_machine.c:692 (FreeTDM/2:1/13811737925)
>>        Callstate Change DOWN -> HANGUP
>>        2013-08-02 14:36:42.003531 [DEBUG]
>>        switch_core_state_machine.c:447 (FreeTDM/2:1/13811737925) State
>>        Change CS_HANGUP -> CS_REPORTING
>>        2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341
>>        Send signal FreeTDM/2:1/13811737925 [BREAK]
>>        2013-08-02 14:36:42.003531 [DEBUG]
>>        switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925)
>>        Running State Change CS_REPORTING
>>        2013-08-02 14:36:42.003531 [DEBUG]
>>        switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State
>>        REPORTING
>>        2013-08-02 14:36:42.003531 [DEBUG]
>>        switch_core_state_machine.c:92 FreeTDM/2:1/13811737925 Standard
>>        REPORTING, cause: NORMAL_CIRCUIT_CONGESTION
>>        2013-08-02 14:36:42.003531 [DEBUG]
>>        switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State
>>        REPORTING going to sleep
>>        2013-08-02 14:36:42.003531 [DEBUG]
>>        switch_core_state_machine.c:441 (FreeTDM/2:1/13811737925) State
>>        Change CS_REPORTING -> CS_DESTROY
>>        2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341
>>        Send signal FreeTDM/2:1/13811737925 [BREAK]
>>        2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1549
>>        Session 37 (FreeTDM/2:1/13811737925) Locked, Waiting on external
>>        entities
>>        2013-08-02 14:36:42.023531 [DEBUG] switch_ivr_originate.c:3632
>>        Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>>        2013-08-02 14:36:42.023531 [INFO] mod_dptools.c:3192 Originate
>>        Failed.  Cause: NORMAL_CIRCUIT_CONGESTION
>>        2013-08-02 14:36:42.023531 [NOTICE] switch_channel.c:4593 Hangup
>>        sofia/internal/1008 at 192.168.0.173 <mailto:1008 at 192.168.0.173>
>>        [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]
>>        2013-08-02 14:36:42.023531 [DEBUG] switch_channel.c:3135 Send
>>        signal sofia/internal/1008 at 192.168.0.173
>>        <mailto:1008 at 192.168.0.173> [KILL]
>>        2013-08-02 14:36:42.023531 [DEBUG] switch_core_session.c:1341
>>        Send signal sofia/internal
>>        ---------------------------------------------
>>        the FS is download from git and wanpipe is latest version.
>>          please give a help for that issue.
>> 
>>        Best regards,
>>        James.zhu
>>        website: www.hiastar.com <http://www.hiastar.com>
>> 
>>        _________________________________________________________________________
>>        Professional FreeSWITCH Consulting Services:
>>        consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>        http://www.freeswitchsolutions.com
>> 
>>        
>>        
>> 
>>        Official FreeSWITCH Sites
>>        http://www.freeswitch.org
>>        http://wiki.freeswitch.org
>>        http://www.cluecon.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
>> 
>> 
>>    _________________________________________________________________________
>>    Professional FreeSWITCH Consulting Services:
>>    consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>    http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
>>    CudaTel Communication Server  Official
>>    FreeSWITCH Sites http://www.freeswitch.org
>>    http://wiki.freeswitch.org http://www.cluecon.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
>> 
>>    _________________________________________________________________________
>>    Professional FreeSWITCH Consulting Services:
>>    consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>    http://www.freeswitchsolutions.com
>> 
>>    
>>    
>> 
>>    Official FreeSWITCH Sites
>>    http://www.freeswitch.org
>>    http://wiki.freeswitch.org
>>    http://www.cluecon.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
>> 
>> 
>> _________________________________________________________________________ Professional
>> FreeSWITCH Consulting Services: consulting at freeswitch.org
>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
>> CudaTel Communication Server  Official FreeSWITCH
>> Sites http://www.freeswitch.org http://wiki.freeswitch.org
>> http://www.cluecon.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
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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: Seven Du <dujinfang at gmail.com>
> Subject: Re: [Freeswitch-users] Do we have ESL-proxy?
> Date: August 6, 2013 1:57:48 PM CDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> I was actually thinking how to do that. Raw idea would be run a daemon that connects to multiple FS instances and routes ESL requests from a single client accordingly.
> 
> FS has core-uuid in events so it can tell which FS it comes from, and api and sendmsg might need to be extended to add the core-uuid so the proxy nows which FS it needs to route to.
> 
> Is this what are we talking about? I might do this in Erlang.
> 
> -- 
> Seven Du
> http://www.freeswitch.org.cn
> http://about.me/dujinfang
> http://www.dujinfang.com
> 
> Sent with Sparrow
> 
> On Wednesday, August 7, 2013 at 2:00 AM, Nathan Neulinger wrote:
> 
>> ESL is just a simple TCP socket connection... if all you're wanting to do is rotate between them, there are LOTs of
>> different solutions. If you're looking for load balancing/calculated assignment of which one to use/etc. then you're
>> going to need something more advanced.
>> 
>> I'm not aware of any high-level proxy services that do the sort of things described in amiproxy/astmanproxy, but someone
>> else may be aware of something like it.
>> 
>> -- Nathan
>> 
>> On 08/06/2013 12:57 PM, SamyGo wrote:
>>> Dear Nathan,
>>> Can you refer me some relevant links, this seems more complex solution. I was thinking more like AMIproxy/ AstmanProxy
>>> thing.
>>> 
>>> BR,
>>> Sammy
>>> 
>>> 
>>> 
>>> On Tue, Aug 6, 2013 at 10:40 AM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu>> wrote:
>>> 
>>> With HAProxy, you could just define a list of target servers. You'd configure your app to talk to the HA IP, and
>>> then each request would go to next available server in the pool you have defined.
>>> 
>>> You could also simply do a DNS round robin if you wanted. All depends on what infrastructure/etc. you have available
>>> to you and how resilient it needs to be.
>>> 
>>> -- Nathan
>>> 
>>> 
>>> On 08/06/2013 12:38 PM, SamyGo wrote:
>>> 
>>> Dear Nathan,
>>> 
>>> I think I've missed something. I don't understand how HAproxy and Floating IP and KeepAlived will help me? I need to
>>> have all Active FS Servers and some mechanism to send ESL commands to them in round-robin fashion. I've an
>>> autodialer
>>> script so that Proxy will help me use many FS Servers at the same time !
>>> 
>>> Thanks,
>>> Sammy
>>> 
>>> 
>>> 
>>> On Tue, Aug 6, 2013 at 10:22 AM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu
>>> <mailto:nneul at mst.edu>>> wrote:
>>> 
>>> You might look at the keepalived page on wiki, that's what I use to float an IP in the case where only one
>>> of the FS
>>> instances should be online.
>>> 
>>> You could also use HAProxy to define a group of systems for it to talk to if you want multiple to be online
>>> at once,
>>> but rotating between them.
>>> 
>>> -- Nathan
>>> 
>>> 
>>> On 08/06/2013 12:19 PM, SamyGo wrote:
>>> 
>>> Hi Nathan,
>>> 
>>> I can only think of Round Robin with failover. I definitely don't want to parallel fork an ESL command.
>>> 
>>> --
>>> Sammy
>>> 
>>> 
>>> 
>>> On Tue, Aug 6, 2013 at 10:16 AM, Nathan Neulinger <nneul at mst.edu <mailto:nneul at mst.edu>
>>> <mailto:nneul at mst.edu <mailto:nneul at mst.edu>> <mailto:nneul at mst.edu <mailto:nneul at mst.edu>
>>> 
>>> <mailto:nneul at mst.edu <mailto:nneul at mst.edu>>>> wrote:
>>> 
>>> Are you saying "round robin" or "mirrored" for the commands?
>>> 
>>> i.e. if you send a command over this proxy, would it be duplicated to all of the FS servers, or
>>> sent to "first
>>> available"?
>>> 
>>> -- Nathan
>>> 
>>> 
>>> On 08/06/2013 12:05 PM, SamyGo wrote:
>>> 
>>> Dear Users,
>>> 
>>> I have come up with a requirement to have ESL connection made with multiple FS Servers at the
>>> same time
>>> and send
>>> them
>>> commands(no event receiving is required for now) The connections to FS should be pooled and
>>> monitored
>>> if any FS
>>> goes down.
>>> 
>>> Is there any ESLproxy available to be used readily?
>>> 
>>> Thanks,
>>> Sammy
>>> 
>>> 
>>> --
>>> ------------------------------______--------------------------__--__--
>>> Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu
>>> <mailto:nneul at mst.edu>> <mailto:nneul at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu <mailto:nneul at mst.edu>>>
>>> 
>>> 
>>> Missouri S&T Information Technology (573) 612-1412
>>> System Administrator - Architect
>>> 
>>> 
>>> 
>>> --
>>> ------------------------------____----------------------------__--
>>> Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu> <mailto:nneul at mst.edu <mailto:nneul at mst.edu>>
>>> Missouri S&T Information Technology (573) 612-1412
>>> System Administrator - Architect
>>> 
>>> 
>>> 
>>> --
>>> ------------------------------__------------------------------
>>> Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu>
>>> Missouri S&T Information Technology (573) 612-1412
>>> System Administrator - Architect
>> 
>> --
>> ------------------------------------------------------------
>> Nathan Neulinger nneul at mst.edu
>> Missouri S&T Information Technology (573) 612-1412
>> System Administrator - Architect
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 bdfoster at davri.com  Wed Aug  7 01:25:57 2013
From: bdfoster at davri.com (Brian Foster)
Date: Tue, 6 Aug 2013 17:25:57 -0400
Subject: [Freeswitch-users] freeswitch: Syntax error: word unexpected
 (expecting ")")
In-Reply-To: <CAHgk4BVbqiz81LubGQHtH3g0h2TtrVUyZG9HBe5ZNHYVWX_Z_A@mail.gmail.com>
References: <CAHgk4BXhFoSaEyXTopMNFsDOwd=5kxFPmq5+h=e=EPbatV-mFw@mail.gmail.com>
	<CAHgk4BVhPAfJ-rdAs2pc2mG-jz86N1BsicEZThq0Y-XCWqxAWA@mail.gmail.com>
	<CAHgk4BXaY6gPjhuG-_JR65XUL3waG34V-9DpzY0+wcUTQVNF7A@mail.gmail.com>
	<5200AEF5.3010400@aerea.de>
	<CAHgk4BVbqiz81LubGQHtH3g0h2TtrVUyZG9HBe5ZNHYVWX_Z_A@mail.gmail.com>
Message-ID: <CAHgk4BWivCsGYOWc8QEQFHiV+LrZZWrAXCaLc=Bmux9Xg1X9UA@mail.gmail.com>

Yea....I'm just dumb. Lack of sleep.

./freeswitch works just fine.



Thank you,

Brian Foster
Project Manager/Owner's Representative
Davri Investments, Incorporated
P: +1-317-787-2686
M: +1-317-600-9753
Indianapolis, Indiana


On Tue, Aug 6, 2013 at 1:41 PM, Brian Foster <bdfoster at davri.com> wrote:

> Thank you for the reply.
>
> root at bdfoster-workstation:/usr/local/freeswitch/bin# file freeswitch
> freeswitch: ELF 64-bit LSB executable, x86-64, version 1 (SYSV),
> dynamically linked (uses shared libs), for GNU/Linux 2.6.24,
> BuildID[sha1]=0x36ddbb4c4cc574a7188b0896e02526de778a5016, not stripped
>
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Representative
> Davri Investments, Incorporated
> P: +1-317-787-2686
> M: +1-317-600-9753
> Indianapolis, Indiana
>
>
> On Tue, Aug 6, 2013 at 4:08 AM, S?ren Sprenger <soeren.sprenger at aerea.de>wrote:
>
>>  Hi,
>>
>> what does "file /path/to/freeswitch" say which kind binary executable
>> file it is? Seams your freeswitch is not binary compatible with your
>> running system...
>>
>>
>>
>> On 06.08.2013 09:54, Brian Foster wrote:
>>
>> Fresh build on a bare metal Dell Precision 490 running Ubuntu 12.04
>> 64-bit, latest v1.2.stable (5 min ago). When I try to run freeswitch, I get:
>>
>> freeswitch:1 freeswitch: Syntax error: word unexpected (expecting ")")
>>
>> I've since deleted everything in /usr/local/freeswitch and
>> /usr/local/src/freeswitch, and ran the install again. Still have the same
>> outcome.
>>
>> The same version was built on another server, no issues. What could be
>> the problem? Anything I can do to mitigate the issue?
>>
>> Thank you,
>>
>> Brian Foster
>> Project Manager/Owner's Representative
>> Davri Investments, Incorporated
>> P: +1-317-787-2686
>> M: +1-317-600-9753
>> Indianapolis, Indiana
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>>
>>
>>
>> --
>> AereA NetworX UG (haftungsbeschr?nkt)
>> Im Camisch 8
>> 07768 Kahla
>> Gesch?ftsf?hrerin: Franziska Sprenger
>> Handelsregister: Jena B504724
>> E-Mail: info at aerea.de
>> Web: http://www.aerea.de
>> Fon: +49 (0) 36424 760823
>> Fax: +49 (0) 36651 1390009
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 nandy1925 at gmail.com  Wed Aug  7 02:47:10 2013
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Wed, 7 Aug 2013 06:47:10 +0800
Subject: [Freeswitch-users] Freeswitch installation
In-Reply-To: <CAMvhjJtd_PJUN2K3eGY-f0Um_3hiPPm9UREck5b5rf8=Qb9bEQ@mail.gmail.com>
References: <CAMvhjJvS+_PA_Ou2WKxPnscp31y+7U8Fx5K0ZC3rgqLi6PnwiQ@mail.gmail.com>
	<004d01ce90e1$1b78b560$526a2020$@v10networks.ca>
	<CAMvhjJuAkxgmo=VS8YNayZvCR8gK1F1BW_JZOXJ4wwSrR7Xx9A@mail.gmail.com>
	<005801ce90e5$6f12b2d0$4d381870$@v10networks.ca>
	<CAMvhjJv9u5mcUDPxUch0mNsEZ_o1POuP1quM9uUPMM7K2hgtWA@mail.gmail.com>
	<CALtno0-EZ9kKjAUo2qOXgWcTH7ADK750FXK1go0L9i8wjg9UGg@mail.gmail.com>
	<CAMvhjJuq=xY9ySsmNDFAxZAvk6mci4yZ4UeC4-bU7D1GTUgGMQ@mail.gmail.com>
	<CAMvhjJtd_PJUN2K3eGY-f0Um_3hiPPm9UREck5b5rf8=Qb9bEQ@mail.gmail.com>
Message-ID: <CAHKqMb43wWXvXazh6-5zUQnPp38HH6EnfiNA1TGtF-B9aDErrQ@mail.gmail.com>

The Wiki can help newbies a lot to start with FreeSWITCH. Here's the one
for GSM:
http://wiki.freeswitch.org/wiki/GSMopen

/Nandy
On Sun, Aug 4, 2013 at 10:30 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Thank you sir...problem solved...but can you help me on how to enable
> freeswitch to use gsm modem
> On Aug 4, 2013 7:48 PM, "Rafal Gwizdala" <rafal.gwizdala at gmail.com> wrote:
>
> Run the console in administrator mode (right click on command line/run as
> administrator). Then start freeswitch.console.exe
> or just configure FS as a windows service under local system account
>
>
> On Sun, Aug 4, 2013 at 11:43 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Sir i tried running it from cmd prompt and it showed me
>> Cannot open pid file C:/Program Files/FreeSWITCH/run/freeswitch.pid
>> On Aug 4, 2013 1:13 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:
>>
>>> Well, run it under a command prompt window and see what the console is
>>> telling you. I.e. do not start it up from Windows Explorer.****
>>>
>>> ** **
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish
>>> Mishra
>>> *Sent:* Sunday, August 4, 2013 12:13 AM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] Freeswitch installation****
>>>
>>> ** **
>>>
>>> Sir i have already installed VC2010 EXPRESS...****
>>>
>>> On Aug 4, 2013 12:40 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:****
>>>
>>> Install the VC2012 and the VC2010 Runtime before you install the
>>> FreeSWITCH windows binaries****
>>>
>>>  ****
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish
>>> Mishra
>>> *Sent:* Sunday, August 4, 2013 12:03 AM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* [Freeswitch-users] Freeswitch installation****
>>>
>>>  ****
>>>
>>> Sir i have downloaded the msi file for windows of freeswitch and also
>>> ran the setup file to install freeswitch...but when i double click on the
>>> freeswitchconsole.exe file from the windows explorer it automatically
>>> closes...kindly help with this...
>>> Ashish mishra****
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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****
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 adahary at gmail.com  Wed Aug  7 02:55:51 2013
From: adahary at gmail.com (adahary)
Date: Tue, 6 Aug 2013 15:55:51 -0700 (PDT)
Subject: [Freeswitch-users] fs_ivrd 'Could not open input file'
In-Reply-To: <1375788086795-7593663.post@n2.nabble.com>
References: <1375788086795-7593663.post@n2.nabble.com>
Message-ID: <1375829751550-7593696.post@n2.nabble.com>

I've managed to run the outbound script with xinetd (give up on fs_ivrd with
php).

I can connect and answer and read back the response.

Now I'm trying to read some digits from the user so I execute:
$cmd = "call-command: execute\n";
$cmd .= "execute-app-name: play_and_get_digits\n";
$cmd .= "execute-app-arg: 4 10
/usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav
digits 10000 #\n\n";
echo $cmd ;

and get back imediatly only the system reply 'Content-Type' with no user's
digits (the digits are being displayed on the fs_cli console).
I tries to loop on the fgets() while I'm entering digits on the phone - but
still no digits event was sent back.

I did also:
$cmd = "call-command: myevents\n\n";
$cmd = "call-command: linger\n\n";
but still couldn't read back the user's digits.

How to read the digits? what I'm missing?

thanks
assaf



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/fs-ivrd-Could-not-open-input-file-tp7593663p7593696.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From adahary at gmail.com  Wed Aug  7 02:59:20 2013
From: adahary at gmail.com (adahary)
Date: Tue, 6 Aug 2013 15:59:20 -0700 (PDT)
Subject: [Freeswitch-users] fs_ivrd 'Could not open input file'
In-Reply-To: <1375829751550-7593696.post@n2.nabble.com>
References: <1375788086795-7593663.post@n2.nabble.com>
	<1375829751550-7593696.post@n2.nabble.com>
Message-ID: <1375829960982-7593697.post@n2.nabble.com>

correction - the command is 'read' and not 'play_and_get_digits' as I wrote
before.
$cmd = "call-command: execute\n"; 
$cmd .= "execute-app-name: read\n"; 
$cmd .= "execute-app-arg: 4 10
/usr/local/freeswitch/sounds/en/us/callie/conference/8000/conf-pin.wav
digits 10000 #\n\n"; 
echo $cmd ; 



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/fs-ivrd-Could-not-open-input-file-tp7593663p7593697.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From eidevm5 at gmail.com  Wed Aug  7 03:14:55 2013
From: eidevm5 at gmail.com (Peter)
Date: Wed, 7 Aug 2013 09:14:55 +1000
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAMvhjJt8v0XVnGqBs5pMgEyNR3tDV6GuMFR5Dsb1x6XMCMO4tw@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<52001B43.9080901@freeswitch.org>
	<CAMKwYLCtQV1wCVzPxwhNj2d23eK__uPDYPb98p0ZiJLRqx0GKQ@mail.gmail.com>
	<CAMvhjJt8v0XVnGqBs5pMgEyNR3tDV6GuMFR5Dsb1x6XMCMO4tw@mail.gmail.com>
Message-ID: <CABkS=xptY82XhGFxfOh5ED1P04mZtxKV3Fiy2KmwT+x=8oGmAA@mail.gmail.com>

Do you have a firewall running?

What does

iptables -L

display?


On Tue, Aug 6, 2013 at 5:51 PM, Ashish Mishra <itsme.kunnu at gmail.com> wrote:

> Yes my freeswitch is running...
> On Aug 6, 2013 3:41 AM, "Gustavo Salazar" <guga.salazar.loor at gmail.com>
> wrote:
>
>>
>>
>> Is Freeswitch running?
>> I have seen a similar error when I tried to start the cli and Freeswitch
>> was not running .
>>
>>
>>
>> 2013/8/5 Yossi Neiman <mishehu at freeswitch.org>
>>
>>>  Do you have mod_event_socket loaded?
>>>
>>> -Yossi
>>>
>>>
>>> On 08/05/2013 04:00 PM, Ashish Mishra wrote:
>>>
>>> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i
>>> have installed freeswitch) it gives me the following error :
>>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> 
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>>
>> --
>> Gustavo Salazar
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 stuart.mills3 at btopenworld.com  Wed Aug  7 04:15:53 2013
From: stuart.mills3 at btopenworld.com (Stuart Mills)
Date: Wed, 7 Aug 2013 01:15:53 +0100
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <F496A3CCE6994A408714AE2DCE31BAC6@PBPC>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC><000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com><1138329210A14BE4AEE53D7419427A15@PBPC><CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com><CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com><3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC><CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com><B782F8192B3E46608CE6666897F86BEB@PBPC><CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com><CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com><CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com>
	<F496A3CCE6994A408714AE2DCE31BAC6@PBPC>
Message-ID: <0BB9E38C82674637982EDAF416B7B552@PBPC>

Hi,

Has anyone managed to examine these pastebin cdr?s? 

I?m still wondering if it is a bug or if I need to manage this situation myself, any help/pointers would be fantastic.

Thanks again,

Stuart

From: Stuart Mills 
Sent: Monday, August 05, 2013 10:04 AM
To: FreeSWITCH Users Help 
Subject: Re: [Freeswitch-users] Call at once and XML-CDR

LOL, I guess I should have visited pastebin before asking that question 

I?ve created an account and pasted 2 cdrs - 

http://pastebin.com/wsLnYxhz ? this is the inbound CDR for the call that originates 2 outbound calls using dial at once.

http://pastebin.com/YXkuqWsT ? this is the winner of the 2 outbound calls.

You will see in the inbound paste, the app_log stops at the point of the 2 calls, then it continues in the outbound winners cdr record.

If I just originate one call, this doesn?t happen, all of the app_log data is contained in the inbound cdr record.

Regards,

Stuart

From: Brian Foster 
Sent: Sunday, August 04, 2013 2:32 AM
To: FreeSWITCH Users Help 
Subject: Re: [Freeswitch-users] Call at once and XML-CDR

It's an "if you fail this simple test, part of your problem might be attention to detail" sort of test.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.

On Aug 3, 2013 9:30 PM, "Raymond Chandler" <intralanman at freeswitch.org> wrote:


  On Aug 3, 2013 9:19 PM, "Michael Collins" <msc at freeswitch.org> wrote:
  >
  >  "Speak friend and enter" 

  +1

  Love the LOTR reference :-)

  -Ray 


  _________________________________________________________________________
  Professional FreeSWITCH Consulting Services:
  consulting at freeswitch.org
  http://www.freeswitchsolutions.com

  
  

  Official FreeSWITCH Sites
  http://www.freeswitch.org
  http://wiki.freeswitch.org
  http://www.cluecon.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




--------------------------------------------------------------------------------
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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



--------------------------------------------------------------------------------
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 zhulizhong at live.com  Wed Aug  7 05:52:15 2013
From: zhulizhong at live.com (James zhu)
Date: Wed, 7 Aug 2013 01:52:15 +0000
Subject: [Freeswitch-users] Can not make out going call from sangoma A200
In-Reply-To: <BLU171-W488ED06FE9240BE8CD3149B55D0@phx.gbl>
References: <BLU171-W3672C8625546F122FFC013B5510@phx.gbl>, ,
	<CAHgk4BXX1nArMXv-T5CNzLbjh_wyn0fjzYNOUCaAPsUE14KNdA@mail.gmail.com>,
	, <BLU171-W58FF5C141AECC333E5BD0FB5520@phx.gbl>, ,
	<CAHgk4BVdu6Dk3P=rrRBKquiydLUUGBUxSHauTv73tx=XPq=pjA@mail.gmail.com>,
	<BLU171-W488ED06FE9240BE8CD3149B55D0@phx.gbl>
Message-ID: <BLU171-W755EFA24D8B38183EE150AB55E0@phx.gbl>

thanks for the support. this is log:013-08-02 14:36:36.943647 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/1008 at 192.168.0.173) State ROUTING going to sleepfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/1008 at 192.168.0.173) Running State Change CS_EXECUTEfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/1008 at 192.168.0.173) State EXECUTEfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] mod_sofia.c:230 sofia/internal/1008 at 192.168.0.173 SOFIA EXECUTEfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] switch_core_state_machine.c:209 sofia/internal/1008 at 192.168.0.173 Standard EXECUTEfreeswitch at internal> 
EXECUTE sofia/internal/1008 at 192.168.0.173 set(open=true)freeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] mod_dptools.c:1393 sofia/internal/1008 at 192.168.0.173 SET [open]=[true]freeswitch at internal> 
EXECUTE sofia/internal/1008 at 192.168.0.173 bridge(freetdm/2/1/13811737925)freeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] switch_ivr_originate.c:2060 Parsing global variablesfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] mod_freetdm.c:403 Set codec PCMA 20msfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] mod_freetdm.c:1242 Connect outbound channel FreeTDM/2:1/13811737925freeswitch at internal> 
2013-08-02 14:36:36.943647 [NOTICE] switch_channel.c:1030 New Channel FreeTDM/2:1/13811737925 [db274fb7-a395-4f29-8ff1-5a6634f01c56]freeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] mod_freetdm.c:1251 (FreeTDM/2:1/13811737925) State Change CS_NEW -> CS_INITfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]freeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] mod_freetdm.c:1270 Attached session db274fb7-a395-4f29-8ff1-5a6634f01c56 to channel 2:1freeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] ftmod_analog.c:62 [s2c1][1:3] Changed state from DOWN to DIALINGfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] ftmod_analog.c:437 [s2c1][1:3] ANALOG CHANNEL thread starting.freeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] ftdm_io.c:3381 [s2c1][1:3] Enabled software DTMF detectorfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] ftmod_analog.c:457 [s2c1][1:3] Initialized DTMF detectionfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] ftmod_analog.c:638 [s2c1][1:3] Completed state change from DOWN to DIALING in 1 msfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] ftmod_analog.c:644 [s2c1][1:3] Executing state handler on 2:1 for DIALINGfreeswitch at internal> 
2013-08-02 14:36:36.963647 [DEBUG] ftmod_wanpipe.c:1002 [s2c1][1:3] First packet read stats: Rx queue len: 0, Rx queue size: 10freeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925) Running State Change CS_INITfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:455 (FreeTDM/2:1/13811737925) State INITfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] mod_freetdm.c:430 (FreeTDM/2:1/13811737925) State Change CS_INIT -> CS_ROUTINGfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]freeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:455 (FreeTDM/2:1/13811737925) State INIT going to sleepfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925) Running State Change CS_ROUTINGfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:471 (FreeTDM/2:1/13811737925) State ROUTINGfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] mod_freetdm.c:453 FreeTDM/2:1/13811737925 CHANNEL ROUTINGfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_ivr_originate.c:67 (FreeTDM/2:1/13811737925) State Change CS_ROUTING -> CS_CONSUME_MEDIAfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]freeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:471 (FreeTDM/2:1/13811737925) State ROUTING going to sleepfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925) Running State Change CS_CONSUME_MEDIAfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:490 (FreeTDM/2:1/13811737925) State CONSUME_MEDIAfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:490 (FreeTDM/2:1/13811737925) State CONSUME_MEDIA going to sleepfreeswitch at internal> 
2013-08-02 14:36:41.963531 [DEBUG] ftmod_analog.c:503 [s2c1][1:3] Changed state from DIALING to BUSYfreeswitch at internal> 
2013-08-02 14:36:41.983901 [DEBUG] ftmod_analog.c:638 [s2c1][1:3] Completed state change from DIALING to BUSY in 20 msfreeswitch at internal> 
2013-08-02 14:36:41.983901 [DEBUG] ftmod_analog.c:644 [s2c1][1:3] Executing state handler on 2:1 for BUSYfreeswitch at internal> 
2013-08-02 14:36:41.983901 [DEBUG] ftmod_analog.c:803 [s2c1][1:3] Changed state from BUSY to DOWNfreeswitch at internal> 
2013-08-02 14:36:42.003531 [DEBUG] ftmod_analog.c:638 [s2c1][1:3] Completed state change from BUSY to DOWN in 20 msfreeswitch at internal> 
2013-08-02 14:36:42.003531 [DEBUG] ftmod_analog.c:644 [s2c1][1:3] Executing state handler on 2:1 for DOWNfreeswitch at internal> 
2013-08-02 14:36:42.003531 [DEBUG] mod_freetdm.c:2253 got FXO sig 2:1 [STOP]freeswitch at internal> 
2013-08-02 14:36:42.003531 [NOTICE] mod_freetdm.c:2273 Hangup FreeTDM/2:1/13811737925 [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION]freeswitch at internal> 
2013-08-02 14:36:42.003531 [DEBUG] switch_channel.c:3135 Send signal FreeTDM/2:1/13811737925 [KILL]freeswitch at internal> 
2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]freeswitch at internal> 
freeswitch at internal> -----------------------------------------------------------------------the port is working for inbound calls. 

Best regards,

James.zhu

website: www.hiastar.com

From: zhulizhong at live.com
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 6 Aug 2013 06:42:14 +0000
Subject: Re: [Freeswitch-users] Can not make out going call from sangoma A200




hello:I contacted Sangoma, but I do not have any feedback yet. I think it might be a bugfor analog card with freeSWITCH.  freetdm  partially works with sangoma A200. There are two issues:1) Can not get callerid from IO:  only shows this:Initializing cid data! the callerid is emptyeven I set to CN.2) Can not make outgoing calls by the port that I used to make outgoing calls. even I replaced a FXO module, the problem is same. I use same syntax as follow:http://blog.hiastar.com/?p=276 Hope someone from the FreeSWITCH community to clarify that. 
Best regards,

James.zhu

website: www.hiastar.com

Date: Fri, 2 Aug 2013 23:03:47 -0400
From: bdfoster at davri.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can not make out going call from sangoma A200

I apologize, I misread your email. Hopefully someone who knows FreeTDM will come around. Have you contacted Sangoma? They are the maintainers of FreeTDM as far as I know. You might try that route. They are very familiar with freeswitch im sure :).

Thank you,
Brian Foster

Project Manager/Owner's Rep.

Davri Investments, Inc.

O: 317-787-2686 x2102

M: 317-600-9753

E: bdfoster at davri.com

Indianapolis, Indiana
Sent from a mobile device.
On Aug 2, 2013 10:39 PM, "James zhu" <zhulizhong at live.com> wrote:




thanks, Brian.  actually the 2:1 is idle because the port can accept incoming call. afterI reload freetdm or restart FS, I still can not make outgoing calls. If the port physicallyfailed, how come I can make incoming call use the same port. confused. thanks again.

Best regards,

James.zhu

website: www.hiastar.com

Date: Fri, 2 Aug 2013 14:22:29 -0400
From: bdfoster at davri.com
To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] Can not make out going call from sangoma A200

That's the expected behavior if 2:1 is in use. So that means you use another port to place your outbound call. Tips on how to do that, however, I can't really produce. Hopefully someone with more experience with FreeTDM can chime in.


Thank you,
Brian Foster

Project Manager/Owner's Rep.

Davri Investments, Inc.

O: 317-787-2686 x2102

M: 317-600-9753

E: bdfoster at davri.com

Indianapolis, Indiana
Sent from a mobile device.
On Aug 2, 2013 3:09 AM, "James zhu" <zhulizhong at live.com> wrote:




hello:I installed freeswitch and sangoma A200 with 2 FXOs. i set the tone group to CNand opermode=china. the A200 can make accept incoming calls from FreeTDM/2:1 , but I use same port to make out going port, the shows the port is CONGESTION.

--------------------------system log-----------------------------------------------2013-08-02 14:36:42.003531 [DEBUG] mod_freetdm.c:640 [2:1] FreeTDM/2:1/13811737925 CHANNEL HANGUP EXIT

2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:48 FreeTDM/2:1/13811737925 Standard HANGUP, cause: NORMAL_CIRCUIT_CONGESTION2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:679 (FreeTDM/2:1/13811737925) State HANGUP going to sleep

2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:692 (FreeTDM/2:1/13811737925) Callstate Change DOWN -> HANGUP2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:447 (FreeTDM/2:1/13811737925) State Change CS_HANGUP -> CS_REPORTING

2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925) Running State Change CS_REPORTING

2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State REPORTING2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:92 FreeTDM/2:1/13811737925 Standard REPORTING, cause: NORMAL_CIRCUIT_CONGESTION

2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State REPORTING going to sleep2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:441 (FreeTDM/2:1/13811737925) State Change CS_REPORTING -> CS_DESTROY

2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1549 Session 37 (FreeTDM/2:1/13811737925) Locked, Waiting on external entities

2013-08-02 14:36:42.023531 [DEBUG] switch_ivr_originate.c:3632 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]2013-08-02 14:36:42.023531 [INFO] mod_dptools.c:3192 Originate Failed.  Cause: NORMAL_CIRCUIT_CONGESTION

2013-08-02 14:36:42.023531 [NOTICE] switch_channel.c:4593 Hangup sofia/internal/1008 at 192.168.0.173 [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]2013-08-02 14:36:42.023531 [DEBUG] switch_channel.c:3135 Send signal sofia/internal/1008 at 192.168.0.173 [KILL]

2013-08-02 14:36:42.023531 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal---------------------------------------------the FS is download from git and wanpipe is latest version.  please give a help for that issue.


Best regards,

James.zhu

website: www.hiastar.com 		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.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




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.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




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 		 	   		  

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 lloyd.aloysius at gmail.com  Wed Aug  7 07:25:48 2013
From: lloyd.aloysius at gmail.com (lloyd.aloysius at gmail.com)
Date: Wed, 7 Aug 2013 03:25:48 +0000
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <0BB9E38C82674637982EDAF416B7B552@PBPC>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>
	<1138329210A14BE4AEE53D7419427A15@PBPC>
	<CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
	<CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
	<3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
	<CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>
	<B782F8192B3E46608CE6666897F86BEB@PBPC>
	<CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com>
	<CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com>
	<CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com>
	<F496A3CCE6994A408714AE2DCE31BAC6@PBPC>
	<0BB9E38C82674637982EDAF416B7B552@PBPC>
Message-ID: <1599944384-1375845943-cardhu_decombobulator_blackberry.rim.net-1994125046-@b14.c5.bise6.blackberry>

Freeswitch pastebin link below

http://pastebin.freeswitch.org/

Lloyd
Sent from my BlackBerry device on the Rogers Wireless Network

-----Original Message-----
From: "Stuart Mills" <stuart.mills3 at btopenworld.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Wed, 7 Aug 2013 01:15:53 
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] Call at once and XML-CDR

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 pasha at prosperity4ever.com  Wed Aug  7 08:44:31 2013
From: pasha at prosperity4ever.com (Paul)
Date: Wed, 07 Aug 2013 04:37:31 -0007
Subject: [Freeswitch-users] FreeSwitch and Yealink
Message-ID: <20130807044439.9A5F8F4002@mail.mydcs.ca>

Hi guys,

Has anyone had any issues using FreeSwitch with Yealink phones? My 
phones connect to FS via openvpn tunnel. All incoming calls work no 
problem, call comes through phones ring everyone can hear each other no 
issues, having a very strange issue though on the outgoing calls. As 
soon as the destination party picks up (this external calls) the call 
hangs up.

short FS LOG:

switch_ivr_bridge.c:475 Channel [sofia/internal/105 at 10.0.0.34] has been 
answered
sofia.c:6528 Hangup sofia/internal/105 at 10.0.0.34 [CS_EXECUTE] 
[ORIGINATOR_CANCEL]
switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 
[CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]?

So FS thinks the phone sent a BYE packet (which I can see with 
siptrace) but the phone's timer keeps going as if it thinks the call is 
supposed to keep going.

Internal extension to extension works fine (even if the extensions are 
at a different physical location and subnet).

I setup a second account to one of my asterisk servers and 
outgoing/incoming work just fine, so it seems this strange combination 
of FS and Yealink ... does it on 2 models T32G and T38G (only phones I 
have).

I have updated firwmare to their latest version (which in the comments 
say freeswitch ready)

Wondering if anyone else had any experience with these, or has some 
thoughts?

Thanks

Paul
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From karl at xtronics.com  Wed Aug  7 08:50:51 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Tue, 06 Aug 2013 23:50:51 -0500
Subject: [Freeswitch-users] freeDTM  detect if line is in use
In-Reply-To: <1599944384-1375845943-cardhu_decombobulator_blackberry.rim.net-1994125046-@b14.c5.bise6.blackberry>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>
	<1138329210A14BE4AEE53D7419427A15@PBPC>
	<CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
	<CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
	<3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
	<CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>
	<B782F8192B3E46608CE6666897F86BEB@PBPC>
	<CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com>
	<CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com>
	<CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com>
	<F496A3CCE6994A408714AE2DCE31BAC6@PBPC>
	<0BB9E38C82674637982EDAF416B7B552@PBPC>
	<1599944384-1375845943-cardhu_decombobulator_blackberry.rim.net-1994125046-@b14.c5.bise6.blackberry>
Message-ID: <5201D22B.4080104@xtronics.com>

I want to use a freeDTM fxs to make local calls on a POTS line IF it isn't busy - not seeing an 
example of how to test. ( There are some ftdm commands for the fs_cli - but I'm not seeing how to 
grab this information for the dialplan.

Has someone else set up a backup line like this?




--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

With great power comes - some complexity.
I can deal with a bit of complexity - that's
why I run Debian Linux.
--------------------------------------------------------------------------------


From ivan at c3i.bg  Wed Aug  7 09:48:02 2013
From: ivan at c3i.bg (Ivan)
Date: Wed, 07 Aug 2013 08:48:02 +0300
Subject: [Freeswitch-users] FreeSwitch and Yealink
In-Reply-To: <20130807044439.9A5F8F4002@mail.mydcs.ca>
References: <20130807044439.9A5F8F4002@mail.mydcs.ca>
Message-ID: <5201DF92.9010800@c3i.bg>

Hangups with the cause you describe can be caused by a failed codec 
negotiation. Are you sure it's not the case ? Does the remote phone send 
a sip "not acceptable here" ?

If needed you can force the selection of the "outgoing" codec:

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

ivan


On 08/07/2013 07:44 AM, Paul wrote:
> Hi guys,
>
> Has anyone had any issues using FreeSwitch with Yealink phones? My
> phones connect to FS via openvpn tunnel. All incoming calls work no
> problem, call comes through phones ring everyone can hear each other no
> issues, having a very strange issue though on the outgoing calls. As
> soon as the destination party picks up (this external calls) the call
> hangs up.
>
> short FS LOG:
>
> switch_ivr_bridge.c:475 Channel [sofia/internal/105 at 10.0.0.34] has been
> answered
> sofia.c:6528 Hangup sofia/internal/105 at 10.0.0.34 [CS_EXECUTE]
> [ORIGINATOR_CANCEL]
> switch_ivr_bridge.c:721 Hangup sofia/external/2503004900
> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>
> So FS thinks the phone sent a BYE packet (which I can see with siptrace)
> but the phone's timer keeps going as if it thinks the call is supposed
> to keep going.
>
> Internal extension to extension works fine (even if the extensions are
> at a different physical location and subnet).
>
> I setup a second account to one of my asterisk servers and
> outgoing/incoming work just fine, so it seems this strange combination
> of FS and Yealink ... does it on 2 models T32G and T38G (only phones I
> have).
>
> I have updated firwmare to their latest version (which in the comments
> say freeswitch ready)
>
> Wondering if anyone else had any experience with these, or has some
> thoughts?
>
> Thanks
>
> Paul
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
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>
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> http://www.freeswitch.org
>


From pasha at prosperity4ever.com  Wed Aug  7 09:59:58 2013
From: pasha at prosperity4ever.com (Paul)
Date: Wed, 07 Aug 2013 05:52:58 -0007
Subject: [Freeswitch-users] FreeSwitch and Yealink
In-Reply-To: <5201DF92.9010800@c3i.bg>
References: <20130807044439.9A5F8F4002@mail.mydcs.ca> <5201DF92.9010800@c3i.bg>
Message-ID: <20130807060000.E564FF4002@mail.mydcs.ca>

Ivan thanks for your reply.

I'm willing to try anything :)

Where would I set those variables? In the dialplan out to the sip trunk?

Thanks

Paul

On Tue, 6 Aug, 2013 at 10:48 PM, Ivan <ivan at c3i.bg> wrote:
> Hangups with the cause you describe can be caused by a failed codec 
> negotiation. Are you sure it's not the case ? Does the remote phone 
> send 
> a sip "not acceptable here" ?
> 
> If needed you can force the selection of the "outgoing" codec:
> 
> http://wiki.freeswitch.org/wiki/Channel_Variables#absolute_codec_string
> 
> ivan
> 
> 
> On 08/07/2013 07:44 AM, Paul wrote:
>>  Hi guys,
>> 
>>  Has anyone had any issues using FreeSwitch with Yealink phones? My
>>  phones connect to FS via openvpn tunnel. All incoming calls work no
>>  problem, call comes through phones ring everyone can hear each 
>> other no
>>  issues, having a very strange issue though on the outgoing calls. As
>>  soon as the destination party picks up (this external calls) the 
>> call
>>  hangs up.
>> 
>>  short FS LOG:
>> 
>>  switch_ivr_bridge.c:475 Channel [sofia/internal/105 at 10.0.0.34] has 
>> been
>>  answered
>>  sofia.c:6528 Hangup sofia/internal/105 at 10.0.0.34 [CS_EXECUTE]
>>  [ORIGINATOR_CANCEL]
>>  switch_ivr_bridge.c:721 Hangup sofia/external/2503004900
>>  [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>> 
>>  So FS thinks the phone sent a BYE packet (which I can see with 
>> siptrace)
>>  but the phone's timer keeps going as if it thinks the call is 
>> supposed
>>  to keep going.
>> 
>>  Internal extension to extension works fine (even if the extensions 
>> are
>>  at a different physical location and subnet).
>> 
>>  I setup a second account to one of my asterisk servers and
>>  outgoing/incoming work just fine, so it seems this strange 
>> combination
>>  of FS and Yealink ... does it on 2 models T32G and T38G (only 
>> phones I
>>  have).
>> 
>>  I have updated firwmare to their latest version (which in the 
>> comments
>>  say freeswitch ready)
>> 
>>  Wondering if anyone else had any experience with these, or has some
>>  thoughts?
>> 
>>  Thanks
>> 
>>  Paul
>> 
>> 
>>  
>> _________________________________________________________________________
>>  Professional FreeSWITCH Consulting Services:
>>  consulting at freeswitch.org
>>  http://www.freeswitchsolutions.com
>> 
>>  
>>  
>> 
>>  Official FreeSWITCH Sites
>>  http://www.freeswitch.org
>>  http://wiki.freeswitch.org
>>  http://www.cluecon.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
>> 
>> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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> 
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From govoiper at gmail.com  Wed Aug  7 10:53:02 2013
From: govoiper at gmail.com (SamyGo)
Date: Tue, 6 Aug 2013 23:53:02 -0700
Subject: [Freeswitch-users] Do we have ESL-proxy?
In-Reply-To: <FA9F9ADD-567C-47DA-94E2-A2C6269C27E2@gmail.com>
References: <mailman.21987.1375815503.576.freeswitch-users@lists.freeswitch.org>
	<FA9F9ADD-567C-47DA-94E2-A2C6269C27E2@gmail.com>
Message-ID: <CAJUJwthBKVJZD0Z_4hvqf9B6O8Atg1k=NLP_VsYp_DSpVf8Fsg@mail.gmail.com>

Seven yes that's somewhat I was thinking, Chris has somewhat opposite of
the required.

I've multiple clients connecting to multiple FS-ESL sockets. Clients send
just "originate" command and they don't care which FS received the command
and what happened with the request.

All I think about what the ESL-Proxy should do is either send messages to a
mentioned FS-Server if user/client specifies else do a round-robin to
active FS-ESL sockets.

If anyone has any partial script doing anything like this I'll be happy to
modify it according to the requirements and put up for others to use.

Regards,
Sammy




On Tue, Aug 6, 2013 at 12:09 PM, Chris Cachor <ccachor at gmail.com> wrote:

> I'm working on a similar setup except that I'm having Freeswitch make a
> Curl request to a Node.js script with the IP to connect to, and the Node.js
> script will listen to the incoming request then open a ESL inbound socket
> connection to the box that requested it. The Node.js script will handle the
> multiple connection instances (and won't try to connect if it's connected
> to the machine already). Note: The Node script runs as a daemon. It's a
> solution that scales with minimal config changes as well.
>
> - Chris
>
> On Aug 6, 2013, at 1:58 PM, freeswitch-users-request at lists.freeswitch.orgwrote:
>
> 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: Can not make out going call from sangoma A200 (Ivan)
>   2. Re: Do we have ESL-proxy? (Seven Du)
>
> *From: *Ivan <ivan at c3i.bg>
> *Subject: **Re: [Freeswitch-users] Can not make out going call from
> sangoma A200*
> *Date: *August 6, 2013 12:19:57 PM CDT
> *To: *freeswitch-users at lists.freeswitch.org
> *Reply-To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
> WRT the callerid issue, do you answer the call immediately ? If yes you
> should try to wait 2 or 3 rings before answering, since the callerid data
> is usually sent at the first ring.
> There might also be a problem with your provider, did you check first that
> you get the callerid when plugging a telephone to your provider's line ?
>
> Outgoing call problem: please post more debug from fs_cli (before the
> hangup).
>
> ivan
>
>
> On 08/06/2013 09:42 AM, James zhu wrote:
>
> hello:
> I contacted Sangoma, but I do not have any feedback yet. I think it
> might be a bug
> for analog card with freeSWITCH.  freetdm  partially works with sangoma
> A200.
> There are two issues:
> 1) Can not get callerid from IO:  only shows this:Initializing cid data!
> the callerid is empty
> even I set to CN.
> 2) Can not make outgoing calls by the port that I used to make outgoing
> calls. even I replaced a FXO module, the problem is same. I use same
> syntax as follow:
> http://blog.hiastar.com/?p=276
> Hope someone from the FreeSWITCH community to clarify that.
>
> Best regards,
> James.zhu
> website: www.hiastar.com
>
> ------------------------------------------------------------------------
> Date: Fri, 2 Aug 2013 23:03:47 -0400
> From: bdfoster at davri.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Can not make out going call from sangoma
> A200
>
> I apologize, I misread your email. Hopefully someone who knows FreeTDM
> will come around. Have you contacted Sangoma? They are the maintainers
> of FreeTDM as far as I know. You might try that route. They are very
> familiar with freeswitch im sure :).
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com <mailto:bdfoster at davri.com>
> Indianapolis, Indiana
>
> Sent from a mobile device.
>
> On Aug 2, 2013 10:39 PM, "James zhu" <zhulizhong at live.com
> <mailto:zhulizhong at live.com>> wrote:
>
>    thanks, Brian.  actually the 2:1 is idle because the port can accept
>    incoming call. after
>    I reload freetdm or restart FS, I still can not make outgoing calls.
>    If the port physically
>    failed, how come I can make incoming call use the same port.
>    confused. thanks again.
>
>    Best regards,
>    James.zhu
>    website: www.hiastar.com <http://www.hiastar.com>
>
>    ------------------------------------------------------------------------
>    Date: Fri, 2 Aug 2013 14:22:29 -0400
>    From: bdfoster at davri.com <mailto:bdfoster at davri.com>
>    To: freeswitch-users at lists.freeswitch.org
>    <mailto:freeswitch-users at lists.freeswitch.org>
>    Subject: Re: [Freeswitch-users] Can not make out going call from
>    sangoma A200
>
>    That's the expected behavior if 2:1 is in use. So that means you use
>    another port to place your outbound call. Tips on how to do that,
>    however, I can't really produce. Hopefully someone with more
>    experience with FreeTDM can chime in.
>
>    Thank you,
>
>    Brian Foster
>    Project Manager/Owner's Rep.
>    Davri Investments, Inc.
>    O: 317-787-2686 x2102
>    M: 317-600-9753
>    E: bdfoster at davri.com <mailto:bdfoster at davri.com>
>    Indianapolis, Indiana
>
>    Sent from a mobile device.
>
>    On Aug 2, 2013 3:09 AM, "James zhu" <zhulizhong at live.com
>    <mailto:zhulizhong at live.com>> wrote:
>
>        hello:
>        I installed freeswitch and sangoma A200 with 2 FXOs. i set the
>        tone group to CN
>        and opermode=china. the A200 can make accept incoming calls from
>        FreeTDM/2:1 , but I use same port to make out going port, the
>        shows the port is CONGESTION.
>        --------------------------system
>        log-----------------------------------------------
>        2013-08-02 14:36:42.003531 [DEBUG] mod_freetdm.c:640 [2:1]
>        FreeTDM/2:1/13811737925 CHANNEL HANGUP EXIT
>        2013-08-02 14:36:42.003531 [DEBUG]
>        switch_core_state_machine.c:48 FreeTDM/2:1/13811737925 Standard
>        HANGUP, cause: NORMAL_CIRCUIT_CONGESTION
>        2013-08-02 14:36:42.003531 [DEBUG]
>        switch_core_state_machine.c:679 (FreeTDM/2:1/13811737925) State
>        HANGUP going to sleep
>        2013-08-02 14:36:42.003531 [DEBUG]
>        switch_core_state_machine.c:692 (FreeTDM/2:1/13811737925)
>        Callstate Change DOWN -> HANGUP
>        2013-08-02 14:36:42.003531 [DEBUG]
>        switch_core_state_machine.c:447 (FreeTDM/2:1/13811737925) State
>        Change CS_HANGUP -> CS_REPORTING
>        2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341
>        Send signal FreeTDM/2:1/13811737925 [BREAK]
>        2013-08-02 14:36:42.003531 [DEBUG]
>        switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925)
>        Running State Change CS_REPORTING
>        2013-08-02 14:36:42.003531 [DEBUG]
>        switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State
>        REPORTING
>        2013-08-02 14:36:42.003531 [DEBUG]
>        switch_core_state_machine.c:92 FreeTDM/2:1/13811737925 Standard
>        REPORTING, cause: NORMAL_CIRCUIT_CONGESTION
>        2013-08-02 14:36:42.003531 [DEBUG]
>        switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State
>        REPORTING going to sleep
>        2013-08-02 14:36:42.003531 [DEBUG]
>        switch_core_state_machine.c:441 (FreeTDM/2:1/13811737925) State
>        Change CS_REPORTING -> CS_DESTROY
>        2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341
>        Send signal FreeTDM/2:1/13811737925 [BREAK]
>        2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1549
>        Session 37 (FreeTDM/2:1/13811737925) Locked, Waiting on external
>        entities
>        2013-08-02 14:36:42.023531 [DEBUG] switch_ivr_originate.c:3632
>        Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]
>        2013-08-02 14:36:42.023531 [INFO] mod_dptools.c:3192 Originate
>        Failed.  Cause: NORMAL_CIRCUIT_CONGESTION
>        2013-08-02 14:36:42.023531 [NOTICE] switch_channel.c:4593 Hangup
>        sofia/internal/1008 at 192.168.0.173 <mailto:1008 at 192.168.0.173>
>        [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]
>        2013-08-02 14:36:42.023531 [DEBUG] switch_channel.c:3135 Send
>        signal sofia/internal/1008 at 192.168.0.173
>        <mailto:1008 at 192.168.0.173> [KILL]
>        2013-08-02 14:36:42.023531 [DEBUG] switch_core_session.c:1341
>        Send signal sofia/internal
>        ---------------------------------------------
>        the FS is download from git and wanpipe is latest version.
>          please give a help for that issue.
>
>        Best regards,
>        James.zhu
>        website: www.hiastar.com <http://www.hiastar.com>
>
>
>
>        _________________________________________________________________________
>        Professional FreeSWITCH Consulting Services:
>        consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>
>        http://www.freeswitchsolutions.com
>
>        
>        
>
>        Official FreeSWITCH Sites
>        http://www.freeswitch.org
>        http://wiki.freeswitch.org
>        http://www.cluecon.com
>
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>
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>
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>
>    
>    
>
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>
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> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
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>
> *From: *Seven Du <dujinfang at gmail.com>
> *Subject: **Re: [Freeswitch-users] Do we have ESL-proxy?*
> *Date: *August 6, 2013 1:57:48 PM CDT
> *To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Reply-To: *FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>
>
>  I was actually thinking how to do that. Raw idea would be run a daemon
> that connects to multiple FS instances and routes ESL requests from a
> single client accordingly.
>
> FS has core-uuid in events so it can tell which FS it comes from, and api
> and sendmsg might need to be extended to add the core-uuid so the proxy
> nows which FS it needs to route to.
>
> Is this what are we talking about? I might do this in Erlang.
>
> --
> Seven Du
> http://www.freeswitch.org.cn
> http://about.me/dujinfang
> http://www.dujinfang.com
>
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Wednesday, August 7, 2013 at 2:00 AM, Nathan Neulinger wrote:
>
> ESL is just a simple TCP socket connection... if all you're wanting to do
> is rotate between them, there are LOTs of
> different solutions. If you're looking for load balancing/calculated
> assignment of which one to use/etc. then you're
> going to need something more advanced.
>
> I'm not aware of any high-level proxy services that do the sort of things
> described in amiproxy/astmanproxy, but someone
> else may be aware of something like it.
>
> -- Nathan
>
> On 08/06/2013 12:57 PM, SamyGo wrote:
>
> Dear Nathan,
> Can you refer me some relevant links, this seems more complex solution. I
> was thinking more like AMIproxy/ AstmanProxy
> thing.
>
> BR,
> Sammy
>
>
>
> On Tue, Aug 6, 2013 at 10:40 AM, Nathan Neulinger <nneul at mst.edu <
> mailto:nneul at mst.edu <nneul at mst.edu>>> wrote:
>
> With HAProxy, you could just define a list of target servers. You'd
> configure your app to talk to the HA IP, and
> then each request would go to next available server in the pool you have
> defined.
>
> You could also simply do a DNS round robin if you wanted. All depends on
> what infrastructure/etc. you have available
> to you and how resilient it needs to be.
>
> -- Nathan
>
>
> On 08/06/2013 12:38 PM, SamyGo wrote:
>
> Dear Nathan,
>
> I think I've missed something. I don't understand how HAproxy and Floating
> IP and KeepAlived will help me? I need to
> have all Active FS Servers and some mechanism to send ESL commands to them
> in round-robin fashion. I've an
> autodialer
> script so that Proxy will help me use many FS Servers at the same time !
>
> Thanks,
> Sammy
>
>
>
> On Tue, Aug 6, 2013 at 10:22 AM, Nathan Neulinger <nneul at mst.edu <
> mailto:nneul at mst.edu <nneul at mst.edu>> <mailto:nneul at mst.edu<nneul at mst.edu>
> <mailto:nneul at mst.edu <nneul at mst.edu>>>> wrote:
>
> You might look at the keepalived page on wiki, that's what I use to float
> an IP in the case where only one
> of the FS
> instances should be online.
>
> You could also use HAProxy to define a group of systems for it to talk to
> if you want multiple to be online
> at once,
> but rotating between them.
>
> -- Nathan
>
>
> On 08/06/2013 12:19 PM, SamyGo wrote:
>
> Hi Nathan,
>
> I can only think of Round Robin with failover. I definitely don't want to
> parallel fork an ESL command.
>
> --
> Sammy
>
>
>
> On Tue, Aug 6, 2013 at 10:16 AM, Nathan Neulinger <nneul at mst.edu <
> mailto:nneul at mst.edu <nneul at mst.edu>>
> <mailto:nneul at mst.edu <nneul at mst.edu> <mailto:nneul at mst.edu<nneul at mst.edu>>>
> <mailto:nneul at mst.edu <nneul at mst.edu> <mailto:nneul at mst.edu<nneul at mst.edu>
> >
>
> <mailto:nneul at mst.edu <nneul at mst.edu> <mailto:nneul at mst.edu<nneul at mst.edu>>>>>
> wrote:
>
> Are you saying "round robin" or "mirrored" for the commands?
>
> i.e. if you send a command over this proxy, would it be duplicated to all
> of the FS servers, or
> sent to "first
> available"?
>
> -- Nathan
>
>
> On 08/06/2013 12:05 PM, SamyGo wrote:
>
> Dear Users,
>
> I have come up with a requirement to have ESL connection made with
> multiple FS Servers at the
> same time
> and send
> them
> commands(no event receiving is required for now) The connections to FS
> should be pooled and
> monitored
> if any FS
> goes down.
>
> Is there any ESLproxy available to be used readily?
>
> Thanks,
> Sammy
>
>
> --
> ------------------------------______--------------------------__--__--
> Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu <nneul at mst.edu>> <
> mailto:nneul at mst.edu <nneul at mst.edu>
> <mailto:nneul at mst.edu <nneul at mst.edu>>> <mailto:nneul at mst.edu<nneul at mst.edu><
> mailto:nneul at mst.edu <nneul at mst.edu>> <mailto:nneul at mst.edu<nneul at mst.edu><
> mailto:nneul at mst.edu <nneul at mst.edu>>>>
>
>
> Missouri S&T Information Technology (573) 612-1412
> System Administrator - Architect
>
>
>
> --
> ------------------------------____----------------------------__--
> Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu <nneul at mst.edu>> <
> mailto:nneul at mst.edu <nneul at mst.edu> <mailto:nneul at mst.edu <nneul at mst.edu>
> >>
> Missouri S&T Information Technology (573) 612-1412
> System Administrator - Architect
>
>
>
> --
> ------------------------------__------------------------------
> Nathan Neulinger nneul at mst.edu <mailto:nneul at mst.edu <nneul at mst.edu>>
> Missouri S&T Information Technology (573) 612-1412
> System Administrator - Architect
>
>
> --
> ------------------------------------------------------------
> Nathan Neulinger nneul at mst.edu
> Missouri S&T Information Technology (573) 612-1412
> System Administrator - Architect
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Wed Aug  7 11:12:41 2013
From: eidevm5 at gmail.com (Peter)
Date: Wed, 7 Aug 2013 17:12:41 +1000
Subject: [Freeswitch-users] One way audio to CME
In-Reply-To: <9C8EC1A4-7DAC-4226-BDD6-F63A9AFD107F@gmail.com>
References: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>
	<89239AD2-AE02-4E1C-8692-6E77894D8384@gmail.com>
	<CABkS=xo7NQoTNpeR0dMrLkeBndypfPmTbPEEbUonMu_NJ7wfAA@mail.gmail.com>
	<9C8EC1A4-7DAC-4226-BDD6-F63A9AFD107F@gmail.com>
Message-ID: <CABkS=xpxRiZ_+SGsjO-CASNSnV66zUxhn_-GmaU+S0_HRS=A1Q@mail.gmail.com>

Hi Anthony.

Yes, the SIP profiles are the same for calls going to Kamailio and to
CME/CUBE.

Note that CME only has one interface, so binding the source interface
doesn't really make much sense.

Note that I've simplified my set up a little and the phone that was
registered to CUCM is now registered to CME.  However, the result is still
the same, ie:  one way audio to the Cisco phone.

You can see the SIP debug from CME at:

http://pastebin.freeswitch.org/21274

The call is coming from 1001 at 10.1.1.204 to 3000 at 10.10.10.203

where

10.1.1.204  - Freeswitch where SIP clients register to
10.1.1.206  - External side of Freeswitch SBC
10.10.10.206 - Internal side of Freeswitch SBC
10.10.10.203 - CME

Peter



On Tue, Aug 6, 2013 at 5:13 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:

> Hi Peter,
>
> Because the calls are fine when using Kamailio I'm assuming your sip
> profiles are fine and you FS SBC config is fine. Are you using the same
> profiles?
> Yes you are correct. Have you added the commands? Add them as a first step.
> Send on a 'debug ccsip messages'
>
> Anthony
>
>
>
> On 6 Aug 2013, at 05:35, Peter <eidevm5 at gmail.com> wrote:
>
> Thanks for replying Anthony.
>
> Keep in mind that I have very little experience with Cisco products, so I
> may be missing something fundamental.
>
> As far as I can see
>
> voice-class sip bind media source-interface ....
>
> is just used to bind the SIP or media stream to the appropriate interface
> on the CUBE.
>
> My issue is that the CUBE is trying to initiate the return RTP stream to
> the external interface (instead of the internal interface) on the
> Freeswitch SBC.
>
> Is my understanding of the sip bind media command correct?
>
> Thanks
>
> Peter
>
>
> On Mon, Aug 5, 2013 at 5:23 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>
>> On cube make sure you specify the source address on your dial-peers
>> voice-class sip bind media|control
>> to the correct side. I have seen one way audio when not set.
>>
>> On 5 Aug 2013, at 06:29, Peter <eidevm5 at gmail.com> wrote:
>>
>> >
>> >
>> > I currently have successful two way calls (signalling and media) in the
>> following setup
>> >
>> >
>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  ->
>> Kamailio --> Internal Linphone
>> >
>> > However, when I try to call a Cisco handset that is registered to CUCM9
>> via CME in the following config:
>> >
>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  ->
>> CME ->  CUCM9 -->  Cisco handset
>> >
>> > The call signalling appears to be working fine and I can successfully
>>  initiate a call from each end, but the only RTP stream that is working is
>> from the external Linphone client to the Cisco handset.
>> >
>> > Note that CME is being used as a CUBE device, so all SIP and RTP goes
>> via it.
>> >
>> > Looking at the RTP debugs on CME I can see the problem is that the
>> "Media Dest Addr" is getting set to the external side of the FS SBC rather
>> than the internal IP address.
>> >
>> >
>> > I tried setting adding:
>> >
>> >             <action application="set"
>> data="disable_rtp_auto_adjust="true" />
>> >
>> > to the dialplan on the SBC, but it made no difference.
>> >
>> >
>> > Any suggestions as to what to check next?
>> >
>> > Thanks
>> >
>> > Peter
>> >
>> >
>> _________________________________________________________________________
>>
>>
>> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Wed Aug  7 11:15:11 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Wed, 7 Aug 2013 12:45:11 +0530
Subject: [Freeswitch-users] how to check that my freeswitch is correctly
	installed
Message-ID: <CAMvhjJtLrebLehm85S_oV78oWYAu55w_-VTz0nybWa8=BBhoTA@mail.gmail.com>

hii,
I have installed the freeswitch 1.2 on my ubuntu 12.04 machine. Now how
will make sure that freeswitch has been installed correctly on my ubuntu
machine. Do i need to connect it with a LAN (local) and c install a
softphone to check or there is some other method as well ?
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From itsme.kunnu at gmail.com  Wed Aug  7 11:21:24 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Wed, 7 Aug 2013 12:51:24 +0530
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CABkS=xptY82XhGFxfOh5ED1P04mZtxKV3Fiy2KmwT+x=8oGmAA@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<52001B43.9080901@freeswitch.org>
	<CAMKwYLCtQV1wCVzPxwhNj2d23eK__uPDYPb98p0ZiJLRqx0GKQ@mail.gmail.com>
	<CAMvhjJt8v0XVnGqBs5pMgEyNR3tDV6GuMFR5Dsb1x6XMCMO4tw@mail.gmail.com>
	<CABkS=xptY82XhGFxfOh5ED1P04mZtxKV3Fiy2KmwT+x=8oGmAA@mail.gmail.com>
Message-ID: <CAMvhjJuY3NSinMpu7N=wzh7Y+6zUxw4UfsGFhM6j-ECzY4tXjw@mail.gmail.com>

Sir i would like to give you some information about how am i launching
fs_cli...I have installed freeswitch in my ubuntu machine now on the same
machine i am trying to launch fs_cli but i am getting an error which i have
already mentioned...Also as far as the output of iptables -L is concerned i
am getiing:

Chain INPUT (policy ACCEPT)
target        prot opt source        destination

Chain FORWARD (policy ACCEPT)
target        prot opt source        destination

Chain OUTPUT (policy ACCEPT)
target        prot opt source        destination


On Wed, Aug 7, 2013 at 4:44 AM, Peter <eidevm5 at gmail.com> wrote:

> Do you have a firewall running?
>
> What does
>
> iptables -L
>
> display?
>
>
> On Tue, Aug 6, 2013 at 5:51 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Yes my freeswitch is running...
>> On Aug 6, 2013 3:41 AM, "Gustavo Salazar" <guga.salazar.loor at gmail.com>
>> wrote:
>>
>>>
>>>
>>> Is Freeswitch running?
>>> I have seen a similar error when I tried to start the cli and Freeswitch
>>> was not running .
>>>
>>>
>>>
>>> 2013/8/5 Yossi Neiman <mishehu at freeswitch.org>
>>>
>>>>  Do you have mod_event_socket loaded?
>>>>
>>>> -Yossi
>>>>
>>>>
>>>> On 08/05/2013 04:00 PM, Ashish Mishra wrote:
>>>>
>>>> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which
>>>> i have installed freeswitch) it gives me the following error :
>>>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>>
>>>> 
>>>>
>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>>
>>> --
>>> Gustavo Salazar
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 agtmcgarry at gmail.com  Wed Aug  7 12:31:40 2013
From: agtmcgarry at gmail.com (Anthony McGarry)
Date: Wed, 7 Aug 2013 09:31:40 +0100
Subject: [Freeswitch-users] One way audio to CME
In-Reply-To: <CABkS=xpxRiZ_+SGsjO-CASNSnV66zUxhn_-GmaU+S0_HRS=A1Q@mail.gmail.com>
References: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>
	<89239AD2-AE02-4E1C-8692-6E77894D8384@gmail.com>
	<CABkS=xo7NQoTNpeR0dMrLkeBndypfPmTbPEEbUonMu_NJ7wfAA@mail.gmail.com>
	<9C8EC1A4-7DAC-4226-BDD6-F63A9AFD107F@gmail.com>
	<CABkS=xpxRiZ_+SGsjO-CASNSnV66zUxhn_-GmaU+S0_HRS=A1Q@mail.gmail.com>
Message-ID: <1393ACC3-56D8-49B4-86E7-2F5C80435799@gmail.com>

Hi Peter,

Your debug shows the invite with via/from/contact/rpid all coming from 10.1.1.206, your external side. 
Check your bridge statement, is it using the correct sip profile? 
Check your sip profile SBC internal params rtp-ip & sip-ip, make sure they are set correctly to 10.10.10.206
Paste up your logs from the sbc including sip trace.

Anthony


On 7 Aug 2013, at 08:12, Peter <eidevm5 at gmail.com> wrote:

> Hi Anthony.
> 
> Yes, the SIP profiles are the same for calls going to Kamailio and to CME/CUBE.
> 
> Note that CME only has one interface, so binding the source interface doesn't really make much sense.
> 
> Note that I've simplified my set up a little and the phone that was registered to CUCM is now registered to CME.  However, the result is still the same, ie:  one way audio to the Cisco phone.
> 
> You can see the SIP debug from CME at:
> 
> http://pastebin.freeswitch.org/21274
> 
> The call is coming from 1001 at 10.1.1.204 to 3000 at 10.10.10.203
> 
> where
> 
> 10.1.1.204  - Freeswitch where SIP clients register to
> 10.1.1.206  - External side of Freeswitch SBC
> 10.10.10.206 - Internal side of Freeswitch SBC
> 10.10.10.203 - CME
> 
> Peter
> 
> 
> 
> On Tue, Aug 6, 2013 at 5:13 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
> Hi Peter,
> 
> Because the calls are fine when using Kamailio I'm assuming your sip profiles are fine and you FS SBC config is fine. Are you using the same profiles?
> Yes you are correct. Have you added the commands? Add them as a first step.
> Send on a 'debug ccsip messages' 
> 
> Anthony 
> 
> 
> 
> On 6 Aug 2013, at 05:35, Peter <eidevm5 at gmail.com> wrote:
> 
>> Thanks for replying Anthony.
>> 
>> Keep in mind that I have very little experience with Cisco products, so I may be missing something fundamental.
>> 
>> As far as I can see
>> 
>> voice-class sip bind media source-interface ....
>> 
>> is just used to bind the SIP or media stream to the appropriate interface on the CUBE.
>> 
>> My issue is that the CUBE is trying to initiate the return RTP stream to the external interface (instead of the internal interface) on the Freeswitch SBC.
>> 
>> Is my understanding of the sip bind media command correct?
>> 
>> Thanks
>> 
>> Peter
>> 
>> 
>> On Mon, Aug 5, 2013 at 5:23 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
>> On cube make sure you specify the source address on your dial-peers
>> voice-class sip bind media|control
>> to the correct side. I have seen one way audio when not set.
>> 
>> On 5 Aug 2013, at 06:29, Peter <eidevm5 at gmail.com> wrote:
>> 
>> >
>> >
>> > I currently have successful two way calls (signalling and media) in the following setup
>> >
>> >
>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> Kamailio --> Internal Linphone
>> >
>> > However, when I try to call a Cisco handset that is registered to CUCM9 via CME in the following config:
>> >
>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> CME ->  CUCM9 -->  Cisco handset
>> >
>> > The call signalling appears to be working fine and I can successfully  initiate a call from each end, but the only RTP stream that is working is from the external Linphone client to the Cisco handset.
>> >
>> > Note that CME is being used as a CUBE device, so all SIP and RTP goes via it.
>> >
>> > Looking at the RTP debugs on CME I can see the problem is that the "Media Dest Addr" is getting set to the external side of the FS SBC rather than the internal IP address.
>> >
>> >
>> > I tried setting adding:
>> >
>> >             <action application="set" data="disable_rtp_auto_adjust="true" />
>> >
>> > to the dialplan on the SBC, but it made no difference.
>> >
>> >
>> > Any suggestions as to what to check next?
>> >
>> > Thanks
>> >
>> > Peter
>> >
>> > _________________________________________________________________________
>> 
>> 
> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 zhulizhong at live.com  Wed Aug  7 13:19:14 2013
From: zhulizhong at live.com (james.zhu)
Date: Wed, 7 Aug 2013 17:19:14 +0800
Subject: [Freeswitch-users] Help with PRI Configuration - Bharti Airtel
	- Sangoma 101DE - Freeswitch
References: <51FEBEF0.6090500@yahoo.com>	<B757A1DB-090B-479E-A18D-001A7B4C0F6D@freeswitch.org><CAKzWOxUDrQ=bw5wPsVCwAkO=yZSdYHGjFq6=N=rVJMh8wAfvhw@mail.gmail.com><1375720572.96851.YahooMailNeo@web160505.mail.bf1.yahoo.com>
	<5201277D.8050709@yahoo.com>
Message-ID: <BLU0-SMTP118FFC42052BFDACF49FE36B55E0@phx.gbl>

hello:
i think you have to check kernel of the the version, and make sure to match
exactly. 
Best regards!
Connect With Sangoma
www.hiastar.com
  ----- Original Message ----- 
  From: Ravi S 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Wednesday, August 07, 2013 12:42 AM
  Subject: Re: [Freeswitch-users] Help with PRI Configuration - Bharti Airtel - Sangoma 101DE - Freeswitch


  Hello Everyone,

  Thank you for your responses. For the problem below, the solution is to install the following:

  yum -y install kernel-devel-$(uname -r) libtool* make gcc patch perl bison gcc-c++ ncurses-devel flex libtermcap-devel autoconf* automake* autoconf


  However, I am still struggling to set the PRI card up. will post details with log shortly. 

  Thanks.
  Ravi


  On 05-08-2013 22:06, Ravi wrote:

    Thanks Ken,


    Previously I did not have trouble installing the Sangoma card. Now when I try to go through the FS freetdm wiki, this is the error that I am getting:

    [root at bfree-server wanpipe-7.0.5]# make freetdm
       Error linux headers/source not found: /lib/modules/2.6.32-358.14.1.el6.x86_64/build !

    make: *** [_checksrc] Error 1


    any reason why this is happening ??

    Thanks.
    Ravi






----------------------------------------------------------------------------
    From: Michael Collins <msc at freeswitch.org>
    To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
    Sent: Monday, August 5, 2013 7:12 AM
    Subject: Re: [Freeswitch-users] Help with PRI Configuration - Bharti Airtel - Sangoma 101DE - Freeswitch



    Do they supply a "real" PRI or is it an IP connection with a device to convert to PRI?
    On Aug 4, 2013 4:43 PM, "Ken Rice" <krice at freeswitch.org> wrote:

      you should review the instructions for setting up PRI on the fs wiki... a prindoes not have a username or a password

      Ken
      Sent from my iPad

      On Aug 4, 2013, at 15:52, Ravi <sravi123 at yahoo.com> wrote:

      > Hello Everyone !
      >
      > I am from India. I have recently taken a PRI connection from Bharti
      > Airtel, one of the service providers. I have installed the following:
      >
      > Cent OS
      > Freeswitch, FreeTDM
      > Sangoma Card
      >
      >
      > I think, I have followed all the instructions. I am struggling to
      > configure Freeswitch to start using the PRI connection. This is what I
      > have from the freeswitch cook book, to configure the gateway: we need
      > username/password, server address or IP and port.
      >
      > When I checked with Airtel, they are telling me that they only give
      > username/password and IP address details for an internet connection and
      > not for PRI lines.
      >
      > Has anyone here in the list, tried using an indian service provider and
      > configured in Freeswitch?
      > Has anyone done it for Airtel ?
      > Or please help me to figure out how to configure the PRI so as to make
      > inbound and outbound calls ??
      >
      > Any help is much appreciated.
      >
      > Thanks.
      > Ravi
      > +91-7502029000
      >
      >
      > _________________________________________________________________________
      > Professional FreeSWITCH Consulting Services:
      > consulting at freeswitch.org
      > http://www.freeswitchsolutions.com
      >
      > 
      > 
      >
      > Official FreeSWITCH Sites
      > http://www.freeswitch.org
      > http://wiki.freeswitch.org
      > http://www.cluecon.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

      _________________________________________________________________________
      Professional FreeSWITCH Consulting Services:
      consulting at freeswitch.org
      http://www.freeswitchsolutions.com

      
      

      Official FreeSWITCH Sites
      http://www.freeswitch.org
      http://wiki.freeswitch.org
      http://www.cluecon.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


    _________________________________________________________________________
    Professional FreeSWITCH Consulting Services:
    consulting at freeswitch.org
    http://www.freeswitchsolutions.com

    
    

    Official FreeSWITCH Sites
    http://www.freeswitch.org
    http://wiki.freeswitch.org
    http://www.cluecon.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




     

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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




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


  _________________________________________________________________________
  Professional FreeSWITCH Consulting Services:
  consulting at freeswitch.org
  http://www.freeswitchsolutions.com

  
  

  Official FreeSWITCH Sites
  http://www.freeswitch.org
  http://wiki.freeswitch.org
  http://www.cluecon.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 GB at cm.nl  Wed Aug  7 14:12:02 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Wed, 7 Aug 2013 12:12:02 +0200
Subject: [Freeswitch-users] Correlate Call-ID
Message-ID: <FB7D97A214987F458242ACBDF876140738C388854F@clubvirtual40.ClubMessage.local>

Hello,

I was wondering about the below scenario and if it's possible to do in FreeSwitch:

The setup is like this: Carrier --- FreeSwitch --- Kamailio --- UAS

The UAS sends a SIP INVITE to Kamailio. Kamailio relays the INVITE to FreeSwitch. FreeSwitch creates a new call leg towards the Carrier (new INVITE). So far pretty basic.
When the carrier responds back with a  200 OK, I need to append the Call-ID of the 200 OK to the 200 OK of the initial INVITE as a custom header. Is this possible?

Regards,

Grant
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From stuart.mills3 at btopenworld.com  Wed Aug  7 14:15:58 2013
From: stuart.mills3 at btopenworld.com (Stuart Mills)
Date: Wed, 7 Aug 2013 11:15:58 +0100
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <1599944384-1375845943-cardhu_decombobulator_blackberry.rim.net-1994125046-@b14.c5.bise6.blackberry>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC><000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com><1138329210A14BE4AEE53D7419427A15@PBPC><CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com><CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com><3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC><CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com><B782F8192B3E46608CE6666897F86BEB@PBPC><CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com><CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com><CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com><F496A3CCE6994A408714AE2DCE31BAC6@PBPC><0BB9E38C82674637982EDAF416B7B552@PBPC>
	<1599944384-1375845943-cardhu_decombobulator_blackberry.rim.net-1994125046-@b14.c5.bise6.blackberry>
Message-ID: <30DA71723BF24A35A0EA0F3798BB3C3A@PBPC>

Thankyou Lloyd.

Pastes are as follows -

inbound CDR  -  http://pastebin.freeswitch.org/21275 - shows the app_log 
stop when the calls are originated.

Outbound winner CDR - http://pastebin.freeswitch.org/21276 - shows the 
app_log carry on from where the inbound seemed to stop.

Regards,

Stuart



-----Original Message----- 
From: lloyd.aloysius at gmail.com
Sent: Wednesday, August 07, 2013 4:25 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call at once and XML-CDR

Freeswitch pastebin link below

http://pastebin.freeswitch.org/

Lloyd
Sent from my BlackBerry device on the Rogers Wireless Network

-----Original Message-----
From: "Stuart Mills" <stuart.mills3 at btopenworld.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Wed, 7 Aug 2013 01:15:53
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] Call at once and XML-CDR

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 fs.user at fordior.net  Wed Aug  7 15:01:04 2013
From: fs.user at fordior.net (EL)
Date: Wed, 7 Aug 2013 13:01:04 +0200
Subject: [Freeswitch-users] FreeSwitch and Yealink
In-Reply-To: <20130807044439.9A5F8F4002@mail.mydcs.ca>
References: <20130807044439.9A5F8F4002@mail.mydcs.ca>
Message-ID: <20130807110104.GA4058@0rdior.com>

Hi Paul,
 
> So FS thinks the phone sent a BYE packet (which I can see with
> siptrace) but the phone's timer keeps going as if it thinks the call
> is supposed to keep going.

Get a real capture of the traffic with for example ngrep, dumpcap or
tcpdump.

# ngrep -d tun0  -O /directory/file/save.here.cap

or

# dumpcap -i tun0 -w /directory/file/save.here.cap

Afterwards, analyse the capture with wireshark. It will give you all
the information you need to analyse your issue.
 
-- 
EL


From fs.user at fordior.net  Wed Aug  7 15:07:09 2013
From: fs.user at fordior.net (EL)
Date: Wed, 7 Aug 2013 13:07:09 +0200
Subject: [Freeswitch-users] how to check that my freeswitch is correctly
 installed
In-Reply-To: <CAMvhjJtLrebLehm85S_oV78oWYAu55w_-VTz0nybWa8=BBhoTA@mail.gmail.com>
References: <CAMvhjJtLrebLehm85S_oV78oWYAu55w_-VTz0nybWa8=BBhoTA@mail.gmail.com>
Message-ID: <20130807110709.GB4058@0rdior.com>

Ashish,

I think you're better off starting to read here:

https://wiki.freeswitch.org/wiki/Main_Page#New_Users_-_Start_Here

It explains everything you need to know. 

And yes, trusted users (you) connect to the internal profile. For an
installation on your local machine this could be 127.0.0.1:5060.
-- 
EL


From randhawaay at gmail.com  Wed Aug  7 15:20:01 2013
From: randhawaay at gmail.com (Shan Randhawa)
Date: Wed, 7 Aug 2013 04:20:01 -0700
Subject: [Freeswitch-users] PROBLEM: Making Outbound Calls through voip
Message-ID: <CAECKtz+J5A44NQDp9tg8cHXodfkz25BH+gb+oX70rhMFNq9BSA@mail.gmail.com>

Hello everyone,

I have successfully make calls and send sms through the software stack of
freeswitch and openbts.Now i want to make outbound calls through VOIP by
providing internet backhaul.

If Any one have done that,if possible can send me the configurations used.
And what service he had acquired to make calls like i m considering NEXEMO
and IPTEL.ORG.


Does any one has any feedback on how to do this,

Thanks in Advance.

regards,


Shan Randhawa
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From a.afzali2003 at gmail.com  Wed Aug  7 15:21:38 2013
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Wed, 7 Aug 2013 15:51:38 +0430
Subject: [Freeswitch-users] Problem with separating inline dialplan apps
Message-ID: <CABK0fC6iVUPx_QWNJSadO_Xixf_XB-hCo2Aw+6JA9O0sVK-fuQ@mail.gmail.com>

Hi Guys,

I use this command
  originate loopback/answer,park/context_1/inline &transfer(2001)
over a inbound event socket. Problem is that server takes just
loopback/answer part for new channel:
[NOTICE] switch_channel.c:1030 New Channel loopback/answer-a
[0eb8f17c-ff50-11e2-97ec-d950156f6aef]
and then try park for new channel!
[ERR] switch_core_session.c:496 Could not locate channel type park
When I submit same command over CLI (just added a \ before comma) it works
fine.

appreciate all,
afshin
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From steveayre at gmail.com  Wed Aug  7 17:42:17 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 7 Aug 2013 14:42:17 +0100
Subject: [Freeswitch-users] Problem with separating inline dialplan apps
In-Reply-To: <CABK0fC6iVUPx_QWNJSadO_Xixf_XB-hCo2Aw+6JA9O0sVK-fuQ@mail.gmail.com>
References: <CABK0fC6iVUPx_QWNJSadO_Xixf_XB-hCo2Aw+6JA9O0sVK-fuQ@mail.gmail.com>
Message-ID: <CAFiqYukbaptA3oNWeSzVRkJC-B7k2ob_nQ3yLsvndAwagjKmdA@mail.gmail.com>

You need to escape the comma - see example 4 on
http://wiki.freeswitch.org/wiki/Loopback_endpoint

As it stands now you are doing a forked dial via both the 'loopback'
endpoint and the 'park' endpoint - except there is no such endpoint.

-Steve




On 7 August 2013 12:21, afshin afzali <a.afzali2003 at gmail.com> wrote:

> Hi Guys,
>
> I use this command
>   originate loopback/answer,park/context_1/inline &transfer(2001)
> over a inbound event socket. Problem is that server takes just
> loopback/answer part for new channel:
> [NOTICE] switch_channel.c:1030 New Channel loopback/answer-a
> [0eb8f17c-ff50-11e2-97ec-d950156f6aef]
> and then try park for new channel!
> [ERR] switch_core_session.c:496 Could not locate channel type park
> When I submit same command over CLI (just added a \ before comma) it works
> fine.
>
> appreciate all,
> afshin
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 babak.freeswitch at gmail.com  Wed Aug  7 18:10:52 2013
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Wed, 7 Aug 2013 18:40:52 +0430
Subject: [Freeswitch-users] problem when using playAndGetDigits after bridge
 is completed where bypass_media is true
Message-ID: <CAEVfcaZnRX5hMF==0GBGe7nKgV1FmAmaDNBxYdFsOKNBC-=Qpg@mail.gmail.com>

Hi
I'm using this code in a loop:

session:execute("set","bypass_media=true");
session:execute("bridge","${lcr_auto_route}");
if session:getVariable("last_bridge_hangup_cause") == "USER_BUSY" then
session:sayPhrase("voip_destination_busy",gender);
end

tryIn =
session:playAndGetDigits(1,1,3,10000,"#","phrase:voip_try_call_again:"..gender,"phrase:voip_no_input_entered:"..gender,"\\d+",10000);

but after the bridge ends, playAndGetDigists is not able to playback the
phrase and so returns immediately.

thanks
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From steveayre at gmail.com  Wed Aug  7 18:11:26 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 7 Aug 2013 15:11:26 +0100
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAMvhjJuY3NSinMpu7N=wzh7Y+6zUxw4UfsGFhM6j-ECzY4tXjw@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<52001B43.9080901@freeswitch.org>
	<CAMKwYLCtQV1wCVzPxwhNj2d23eK__uPDYPb98p0ZiJLRqx0GKQ@mail.gmail.com>
	<CAMvhjJt8v0XVnGqBs5pMgEyNR3tDV6GuMFR5Dsb1x6XMCMO4tw@mail.gmail.com>
	<CABkS=xptY82XhGFxfOh5ED1P04mZtxKV3Fiy2KmwT+x=8oGmAA@mail.gmail.com>
	<CAMvhjJuY3NSinMpu7N=wzh7Y+6zUxw4UfsGFhM6j-ECzY4tXjw@mail.gmail.com>
Message-ID: <CAFiqYumS-4QvsgcH48TugtUC3__2PYQSk7Sg-QZru8qDikoFHw@mail.gmail.com>

1) Check mod_event_socket is enabled to load on start in
autoload_configs/modules.conf.xml

2) Check freeswitch is running:
$ ps -C freeswitch -f
UID        PID  PPID  C STIME TTY          TIME CMD
999      12993     1  1 14:44 ?        00:00:23 /usr/bin/freeswitch -u
freeswitch -g freeswitch -nc -rp -nonat

3) Check freeswitch is listening on the ESL socket:
$ netstat -anp | grep :8021
tcp        0      0 127.0.0.1:8021          0.0.0.0:*               LISTEN
     12993/freeswitch

4) Check the freeswitch.log file to check mod_event_socket tried to load,
and for any errors while loading. Or for startup errors if freeswitch isn't
running at all.

Note if it's listening on 127.0.0.1:8021 then you won't be able to connect
from remote machines. If you need that set listen-ip to 0.0.0.0 and make
sure you change the default password and you should also consider using
ACLs or a firewall to restrict who can connect.

-Steve



On 7 August 2013 08:21, Ashish Mishra <itsme.kunnu at gmail.com> wrote:

> Sir i would like to give you some information about how am i launching
> fs_cli...I have installed freeswitch in my ubuntu machine now on the same
> machine i am trying to launch fs_cli but i am getting an error which i have
> already mentioned...Also as far as the output of iptables -L is concerned i
> am getiing:
>
> Chain INPUT (policy ACCEPT)
> target        prot opt source        destination
>
> Chain FORWARD (policy ACCEPT)
> target        prot opt source        destination
>
> Chain OUTPUT (policy ACCEPT)
> target        prot opt source        destination
>
>
> On Wed, Aug 7, 2013 at 4:44 AM, Peter <eidevm5 at gmail.com> wrote:
>
>> Do you have a firewall running?
>>
>> What does
>>
>> iptables -L
>>
>> display?
>>
>>
>> On Tue, Aug 6, 2013 at 5:51 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> Yes my freeswitch is running...
>>> On Aug 6, 2013 3:41 AM, "Gustavo Salazar" <guga.salazar.loor at gmail.com>
>>> wrote:
>>>
>>>>
>>>>
>>>> Is Freeswitch running?
>>>> I have seen a similar error when I tried to start the cli and
>>>> Freeswitch was not running .
>>>>
>>>>
>>>>
>>>> 2013/8/5 Yossi Neiman <mishehu at freeswitch.org>
>>>>
>>>>>  Do you have mod_event_socket loaded?
>>>>>
>>>>> -Yossi
>>>>>
>>>>>
>>>>> On 08/05/2013 04:00 PM, Ashish Mishra wrote:
>>>>>
>>>>> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which
>>>>> i have installed freeswitch) it gives me the following error :
>>>>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Gustavo Salazar
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lloyd.aloysius at sunteltech.ca  Wed Aug  7 19:50:17 2013
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Wed, 7 Aug 2013 11:50:17 -0400
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAFiqYumS-4QvsgcH48TugtUC3__2PYQSk7Sg-QZru8qDikoFHw@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<52001B43.9080901@freeswitch.org>
	<CAMKwYLCtQV1wCVzPxwhNj2d23eK__uPDYPb98p0ZiJLRqx0GKQ@mail.gmail.com>
	<CAMvhjJt8v0XVnGqBs5pMgEyNR3tDV6GuMFR5Dsb1x6XMCMO4tw@mail.gmail.com>
	<CABkS=xptY82XhGFxfOh5ED1P04mZtxKV3Fiy2KmwT+x=8oGmAA@mail.gmail.com>
	<CAMvhjJuY3NSinMpu7N=wzh7Y+6zUxw4UfsGFhM6j-ECzY4tXjw@mail.gmail.com>
	<CAFiqYumS-4QvsgcH48TugtUC3__2PYQSk7Sg-QZru8qDikoFHw@mail.gmail.com>
Message-ID: <CAGvuWYbACF=vR=wMUBG5u1qY4HoGEEqApT0t-4FEa6AzSecMHw@mail.gmail.com>

If you are accessing fs_cli locally , no need to worry about your firewall.
By default fs_cli should work. Check freeswitch process status.

You can try the following method to test freeswitch , go to your
freeswitch/bin folder and run ./freeswitch manually and see the switch is
loading without any issue.

Lloyd


*
*
*  <http://www.sunteltech.ca/blog/>*


On Wed, Aug 7, 2013 at 10:11 AM, Steven Ayre <steveayre at gmail.com> wrote:

> 1) Check mod_event_socket is enabled to load on start in
> autoload_configs/modules.conf.xml
>
> 2) Check freeswitch is running:
> $ ps -C freeswitch -f
> UID        PID  PPID  C STIME TTY          TIME CMD
> 999      12993     1  1 14:44 ?        00:00:23 /usr/bin/freeswitch -u
> freeswitch -g freeswitch -nc -rp -nonat
>
> 3) Check freeswitch is listening on the ESL socket:
> $ netstat -anp | grep :8021
> tcp        0      0 127.0.0.1:8021          0.0.0.0:*
> LISTEN      12993/freeswitch
>
> 4) Check the freeswitch.log file to check mod_event_socket tried to load,
> and for any errors while loading. Or for startup errors if freeswitch isn't
> running at all.
>
> Note if it's listening on 127.0.0.1:8021 then you won't be able to
> connect from remote machines. If you need that set listen-ip to 0.0.0.0 and
> make sure you change the default password and you should also consider
> using ACLs or a firewall to restrict who can connect.
>
> -Steve
>
>
>
> On 7 August 2013 08:21, Ashish Mishra <itsme.kunnu at gmail.com> wrote:
>
>> Sir i would like to give you some information about how am i launching
>> fs_cli...I have installed freeswitch in my ubuntu machine now on the same
>> machine i am trying to launch fs_cli but i am getting an error which i have
>> already mentioned...Also as far as the output of iptables -L is concerned i
>> am getiing:
>>
>> Chain INPUT (policy ACCEPT)
>> target        prot opt source        destination
>>
>> Chain FORWARD (policy ACCEPT)
>> target        prot opt source        destination
>>
>> Chain OUTPUT (policy ACCEPT)
>> target        prot opt source        destination
>>
>>
>> On Wed, Aug 7, 2013 at 4:44 AM, Peter <eidevm5 at gmail.com> wrote:
>>
>>> Do you have a firewall running?
>>>
>>> What does
>>>
>>> iptables -L
>>>
>>> display?
>>>
>>>
>>> On Tue, Aug 6, 2013 at 5:51 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> Yes my freeswitch is running...
>>>> On Aug 6, 2013 3:41 AM, "Gustavo Salazar" <guga.salazar.loor at gmail.com>
>>>> wrote:
>>>>
>>>>>
>>>>>
>>>>> Is Freeswitch running?
>>>>> I have seen a similar error when I tried to start the cli and
>>>>> Freeswitch was not running .
>>>>>
>>>>>
>>>>>
>>>>> 2013/8/5 Yossi Neiman <mishehu at freeswitch.org>
>>>>>
>>>>>>  Do you have mod_event_socket loaded?
>>>>>>
>>>>>> -Yossi
>>>>>>
>>>>>>
>>>>>> On 08/05/2013 04:00 PM, Ashish Mishra wrote:
>>>>>>
>>>>>> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on
>>>>>> which i have installed freeswitch) it gives me the following error :
>>>>>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Gustavo Salazar
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lloyd.aloysius at sunteltech.ca  Wed Aug  7 20:15:18 2013
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Wed, 7 Aug 2013 12:15:18 -0400
Subject: [Freeswitch-users] problem when using playAndGetDigits after
 bridge is completed where bypass_media is true
In-Reply-To: <CAEVfcaZnRX5hMF==0GBGe7nKgV1FmAmaDNBxYdFsOKNBC-=Qpg@mail.gmail.com>
References: <CAEVfcaZnRX5hMF==0GBGe7nKgV1FmAmaDNBxYdFsOKNBC-=Qpg@mail.gmail.com>
Message-ID: <CAGvuWYarf8pNC=GjDXwEnu=euqysmKkrJ13cYEtZjO2VLYHK3g@mail.gmail.com>

before bridge , set the following and give a try

session:execute("set","continue_on_fail=true");


Lloyd
*
*
*  <http://www.sunteltech.ca/blog/>*


On Wed, Aug 7, 2013 at 10:10 AM, Babak Yakhchali <babak.freeswitch at gmail.com
> wrote:

> Hi
> I'm using this code in a loop:
>
> session:execute("set","bypass_media=true");
> session:execute("bridge","${lcr_auto_route}");
>  if session:getVariable("last_bridge_hangup_cause") == "USER_BUSY" then
> session:sayPhrase("voip_destination_busy",gender);
> end
>
> tryIn =
> session:playAndGetDigits(1,1,3,10000,"#","phrase:voip_try_call_again:"..gender,"phrase:voip_no_input_entered:"..gender,"\\d+",10000);
>
> but after the bridge ends, playAndGetDigists is not able to playback the
> phrase and so returns immediately.
>
> thanks
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 moises.silva at gmail.com  Wed Aug  7 20:18:15 2013
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 7 Aug 2013 11:18:15 -0500
Subject: [Freeswitch-users] freeDTM detect if line is in use
In-Reply-To: <5201D22B.4080104@xtronics.com>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>
	<1138329210A14BE4AEE53D7419427A15@PBPC>
	<CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
	<CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
	<3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
	<CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>
	<B782F8192B3E46608CE6666897F86BEB@PBPC>
	<CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com>
	<CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com>
	<CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com>
	<F496A3CCE6994A408714AE2DCE31BAC6@PBPC>
	<0BB9E38C82674637982EDAF416B7B552@PBPC>
	<1599944384-1375845943-cardhu_decombobulator_blackberry.rim.net-1994125046-@b14.c5.bise6.blackberry>
	<5201D22B.4080104@xtronics.com>
Message-ID: <CAA4nhyA4y2xKcP=7MgCv_oQ8zKy9ufE8dipVQQsFvnTWDkbQYQ@mail.gmail.com>

On Tue, Aug 6, 2013 at 11:50 PM, Karl Schmidt <karl at xtronics.com> wrote:

> I want to use a freeDTM fxs to make local calls on a POTS line IF it isn't
> busy - not seeing an
> example of how to test. ( There are some ftdm commands for the fs_cli -
> but I'm not seeing how to
> grab this information for the dialplan.
>
> Has someone else set up a backup line like this?
>

Hello Karl,

You can execute CLI commands (which in FreeSWITCH lingo are called APIs)
from the dialplan using dollar-sign syntax such as:

<condition field="${ftdm_usage(1 1)}" expression="^0$">
</condition>

That checks if span 1 channel 1 has 0 calls. Note there is an inherent
small race between the time you check and the time you try to use the
channel, but that ought to be good enough I think.

*Moises Silva
**Manager, Software Engineering***

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



**<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>

Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
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 | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`|
| YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
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From lloyd.aloysius at sunteltech.ca  Wed Aug  7 20:20:51 2013
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Wed, 7 Aug 2013 12:20:51 -0400
Subject: [Freeswitch-users] PROBLEM: Making Outbound Calls through voip
In-Reply-To: <CAECKtz+J5A44NQDp9tg8cHXodfkz25BH+gb+oX70rhMFNq9BSA@mail.gmail.com>
References: <CAECKtz+J5A44NQDp9tg8cHXodfkz25BH+gb+oX70rhMFNq9BSA@mail.gmail.com>
Message-ID: <CAGvuWYZbDW8PF=DgJ6_M8oK+ywEi6CPokePy8GcMQY-DOV-ifw@mail.gmail.com>

You can find the service provider examples below

https://wiki.freeswitch.org/wiki/SIP_Provider_Examples


Lloyd
*  <http://www.sunteltech.ca/blog/>*


On Wed, Aug 7, 2013 at 7:20 AM, Shan Randhawa <randhawaay at gmail.com> wrote:

> Hello everyone,
>
> I have successfully make calls and send sms through the software stack of
> freeswitch and openbts.Now i want to make outbound calls through VOIP by
> providing internet backhaul.
>
> If Any one have done that,if possible can send me the configurations used.
> And what service he had acquired to make calls like i m considering NEXEMO
> and IPTEL.ORG.
>
>
> Does any one has any feedback on how to do this,
>
> Thanks in Advance.
>
> regards,
>
>
> Shan Randhawa
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 stuart.mills3 at btopenworld.com  Wed Aug  7 21:00:16 2013
From: stuart.mills3 at btopenworld.com (Stuart Mills)
Date: Wed, 7 Aug 2013 18:00:16 +0100
Subject: [Freeswitch-users] att_xfer origination_cancel_key
In-Reply-To: <CAKzWOxXk1Vrc21CNA7SU1HzdJaAVMhG8hzb2dw0JgdHPZeY-Ow@mail.gmail.com>
References: <7EA4E9B1D66C4202A60862B65D251EAB@PBPC><8B109374-6241-4C55-800A-6BCE4E71E618@freeswitch.org><B82AE16B-71B3-4860-8DC3-72D7EB1C2A8A@btopenworld.com>
	<CAKzWOxXk1Vrc21CNA7SU1HzdJaAVMhG8hzb2dw0JgdHPZeY-Ow@mail.gmail.com>
Message-ID: <AF5EA3E6FCE94115B1E704C21C73A06B@PBPC>

Hi Michael,

Thanks again for your tip on this.

I?ve located the code in mod_dptools.c and made the modification to allow custom digits bound to the existing functions of att_xfer. This compiles and appears to work fine but when I choose the ?conference digit? all 3 calls go into limbo. When I press the ?complete transfer? key to hang up the B party and connect A + C, all 3 disconnect. I can see in the debug ?threeway(UUID)? where the UUID looks like a real UUID, so it?s trying to do it just failing for some reason.

I then reverted back (in case I?d made a mistake), compiled and installed again but the problem persists, so I?m wondering if I have some dial plan issue somewhere stopping att_xfer from working properly.

Here is my feature extension, bound to a digit so the B party can initiate att_xfer ? is there anything obviously wrong with what I am doing here?

<extension name="start_att_xfer">
   <condition field="destination_number" expression="^START_ATT_XFER$">
    <action application="set" data="continue_on_fail=true"/>
    <action application="set" data="origination_cancel_key=#"/>
    <action application="set" data="att_xfer_conference_key=5"/>
    <action application="set" data="att_xfer_txcomplete_key=1"/>
    <action application="log" data="INFO inside att_xfer"/>          
    <action application="read" data="4 12 'tone_stream://%(10000,0,350,440)' divertnumber 5000 #"/>
    <action application="log" data="INFO att_xfer to ${divertnumber}"/>          
    <action application="att_xfer" data="sofia/gateway/sip_gateway/44${divertnumber:1}"/>
  </condition>
</extension>

Regards,

Stuart

From: Michael Collins 
Sent: Sunday, August 04, 2013 12:57 AM
To: FreeSWITCH Users Help 
Subject: Re: [Freeswitch-users] att_xfer origination_cancel_key

You may also research and see how you might add this yourself. The place to look is in mod_dialplan.c at approx line 2230 inside static switch_status_t xfer_on_dtmf(...) :

            if (dtmf->digit == '0') {
                switch_caller_extension_t *extension = NULL;
                const char *app = "three_way";

<snip>


In any case, JM is right that it is relatively easy to change that to check for a channel variable. If you peruse the code you'll find lots of examples of that sort of thing. If it's not your cup of tea then by all means open a jira as a feature request.


-MC




On Sat, Aug 3, 2013 at 12:57 AM, Stuart Mills <stuart.mills3 at btopenworld.com> wrote:

  That is great, thanks for your reply.

  I haven't opened a Jira before, so will read up on the wiki and put a request in.

  Regards

  Stuart

  Sent from my iPhone

  On 3 Aug 2013, at 04:31, Jo?o Mesquita <jmesquita at freeswitch.org> wrote:


    I am not looking at the code right now but if I recall correctly no you cannot. It is trivial to make single key configurable. Open a Jira and I can make a patch for it to configure using channel variables. If you want multiple keys like bda, forget it. A lot more complicated...

    Sent from my iPhone

    On Aug 2, 2013, at 6:17 PM, "Stuart Mills" <stuart.mills3 at btopenworld.com> wrote:


      Hi,

      I have noticed that att_xfer has a configurable cancel key to stop the transfer mid dial, is there an option to change keys for the other transfer features? 

      0 is conference all 3 parties at the moment, but I'd like to designate a 4 or some other key, is this possible?

      Many Thanks,

      Stuart Mills
      _________________________________________________________________________
      Professional FreeSWITCH Consulting Services:
      consulting at freeswitch.org
      http://www.freeswitchsolutions.com

      
      

      Official FreeSWITCH Sites
      http://www.freeswitch.org
      http://wiki.freeswitch.org
      http://www.cluecon.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

    _________________________________________________________________________
    Professional FreeSWITCH Consulting Services:
    consulting at freeswitch.org
    http://www.freeswitchsolutions.com

    
    

    Official FreeSWITCH Sites
    http://www.freeswitch.org
    http://wiki.freeswitch.org
    http://www.cluecon.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


  _________________________________________________________________________
  Professional FreeSWITCH Consulting Services:
  consulting at freeswitch.org
  http://www.freeswitchsolutions.com

  
  

  Official FreeSWITCH Sites
  http://www.freeswitch.org
  http://wiki.freeswitch.org
  http://www.cluecon.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





-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org




--------------------------------------------------------------------------------
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 itsme.kunnu at gmail.com  Wed Aug  7 22:39:43 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Thu, 8 Aug 2013 00:09:43 +0530
Subject: [Freeswitch-users] error connecting softphone
Message-ID: <CAMvhjJszS-_L+2kFEZ6cVafTCm2PepkGmKQZu64Ukgk7r5DK9A@mail.gmail.com>

hi
i have installed x-lite softphone on my windows 8 pc (on which freeswitch
is not installed) through LAN i have connected a ubuntu machine on which is
installed when i am trying to add a user on x-lite softphone with domain as
the ip of the ubuntu machine and and user id as 1005 and password as 1234,
as i click ok button the softphone gives me an error saying that cannot
enable user...

thank you
Ashish Mishra
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From fs.user at fordior.net  Wed Aug  7 23:32:34 2013
From: fs.user at fordior.net (EL)
Date: Wed, 7 Aug 2013 21:32:34 +0200
Subject: [Freeswitch-users] error connecting softphone
In-Reply-To: <CAMvhjJszS-_L+2kFEZ6cVafTCm2PepkGmKQZu64Ukgk7r5DK9A@mail.gmail.com>
References: <CAMvhjJszS-_L+2kFEZ6cVafTCm2PepkGmKQZu64Ukgk7r5DK9A@mail.gmail.com>
Message-ID: <20130807193234.GC4058@0rdior.com>

And what did the fs_cli say at the moment x-lite was trying to connect? 

Did you read all the documentation of the URL I sended you earlier?

-- 
EL


From nickolayr at gmail.com  Thu Aug  8 00:47:59 2013
From: nickolayr at gmail.com (Nikolay Rogoshchenkov)
Date: Wed, 7 Aug 2013 16:47:59 -0400
Subject: [Freeswitch-users] [CRIT] switch_core_session.c:1705 Thread Failure!
Message-ID: <CAAQCs32D4Gmzft8mtuxnFFgEkU1149C2K6sovW3CziX0HNy7JQ@mail.gmail.com>

Could you please explain what  this mean in FreeSwitch log???

----------------------------------------------------------------------
*[...]*
*2013-08-07 23:40:48.011687 [CRIT] switch_core_session.c:1705 Thread
Failure!*
*2013-08-07 23:40:48.011687 [CRIT] switch_core_session.c:1665 LUKE: I'm
hit, but not bad.*
*2013-08-07 23:40:48.011687 [CRIT] switch_core_session.c:1666 LUKE'S VOICE:
Artoo, see what you can do with it. Hang on back there....*
*Green laserfire moves past the beeping little robot as his head turns.
 After a few beeps and a twist of his mechanical arm,*
*Artoo reduces the max sessions to 101 thus, saving the switch from certain
doom.*
*2013-08-07 23:40:48.031685 [CRIT] switch_core_session.c:1705 Thread
Failure!*
*2013-08-07 23:40:48.031685 [CRIT] switch_core_session.c:1665 LUKE: I'm
hit, but not bad.*
*2013-08-07 23:40:48.031685 [CRIT] switch_core_session.c:1666 LUKE'S VOICE:
Artoo, see what you can do with it. Hang on back there....*
*Green laserfire moves past the beeping little robot as his head turns.
 After a few beeps and a twist of his mechanical arm,*
*Artoo reduces the max sessions to 101 thus, saving the switch from certain
doom.*
*2013-08-07 23:40:48.041688 [CRIT] switch_core_session.c:1705 Thread
Failure!*
*[...]*
*2013-08-07 23:55:38.141571 [CRIT] switch_core_session.c:1705 Thread
Failure!*
*2013-08-07 23:55:38.151578 [CRIT] switch_core_session.c:1705 Thread
Failure!*
*2013-08-07 23:55:38.171584 [CRIT] switch_core_session.c:1705 Thread
Failure!*
*[..]*
----------------------------------------------------------------------

It is not funny, at that moment freeswitch consumes 98% of CPU time, and
did not response at all.


PS: (Version 1.5.5b git 8d005a4 2013-07-22 14:41:01Z) @FreeBSD 8.2-RELEASE
(x64)

--
Nikolay
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From vipkilla at gmail.com  Thu Aug  8 00:56:46 2013
From: vipkilla at gmail.com (Vik Killa)
Date: Wed, 7 Aug 2013 16:56:46 -0400
Subject: [Freeswitch-users] [CRIT] switch_core_session.c:1705 Thread
	Failure!
In-Reply-To: <CAAQCs32D4Gmzft8mtuxnFFgEkU1149C2K6sovW3CziX0HNy7JQ@mail.gmail.com>
References: <CAAQCs32D4Gmzft8mtuxnFFgEkU1149C2K6sovW3CziX0HNy7JQ@mail.gmail.com>
Message-ID: <CAC-LwPPJNycg47+zOSpLx3Z=HmxXt4S+udhoEczG25j-EByvqA@mail.gmail.com>

it is kind of funny lol
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From anthony.minessale at gmail.com  Thu Aug  8 00:59:04 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 7 Aug 2013 15:59:04 -0500
Subject: [Freeswitch-users] [CRIT] switch_core_session.c:1705 Thread
	Failure!
In-Reply-To: <CAAQCs32D4Gmzft8mtuxnFFgEkU1149C2K6sovW3CziX0HNy7JQ@mail.gmail.com>
References: <CAAQCs32D4Gmzft8mtuxnFFgEkU1149C2K6sovW3CziX0HNy7JQ@mail.gmail.com>
Message-ID: <CAKbxfG_e4C=SBZteLGtb_9SN1CW94nd9cP5C-+unVpbr4FUtYA@mail.gmail.com>

The 98% cpu usage is the catalyst for the error, not the other way around.
You have consumed all your space for threads and the system cannot create
any more for some reason.


FreeBSD has chosen to implement their own threading lib a few times and it
seems to cause us trouble.
Did you set the thread stack size to 244 before launching?





On Wed, Aug 7, 2013 at 3:47 PM, Nikolay Rogoshchenkov
<nickolayr at gmail.com>wrote:

> Could you please explain what  this mean in FreeSwitch log???
>
> ----------------------------------------------------------------------
> *[...]*
> *2013-08-07 23:40:48.011687 [CRIT] switch_core_session.c:1705 Thread
> Failure!*
> *2013-08-07 23:40:48.011687 [CRIT] switch_core_session.c:1665 LUKE: I'm
> hit, but not bad.*
> *2013-08-07 23:40:48.011687 [CRIT] switch_core_session.c:1666 LUKE'S
> VOICE: Artoo, see what you can do with it. Hang on back there....*
> *Green laserfire moves past the beeping little robot as his head turns.
>  After a few beeps and a twist of his mechanical arm,*
> *Artoo reduces the max sessions to 101 thus, saving the switch from
> certain doom.*
> *2013-08-07 23:40:48.031685 [CRIT] switch_core_session.c:1705 Thread
> Failure!*
> *2013-08-07 23:40:48.031685 [CRIT] switch_core_session.c:1665 LUKE: I'm
> hit, but not bad.*
> *2013-08-07 23:40:48.031685 [CRIT] switch_core_session.c:1666 LUKE'S
> VOICE: Artoo, see what you can do with it. Hang on back there....*
> *Green laserfire moves past the beeping little robot as his head turns.
>  After a few beeps and a twist of his mechanical arm,*
> *Artoo reduces the max sessions to 101 thus, saving the switch from
> certain doom.*
> *2013-08-07 23:40:48.041688 [CRIT] switch_core_session.c:1705 Thread
> Failure!*
> *[...]*
> *2013-08-07 23:55:38.141571 [CRIT] switch_core_session.c:1705 Thread
> Failure!*
> *2013-08-07 23:55:38.151578 [CRIT] switch_core_session.c:1705 Thread
> Failure!*
> *2013-08-07 23:55:38.171584 [CRIT] switch_core_session.c:1705 Thread
> Failure!*
> *[..]*
> ----------------------------------------------------------------------
>
> It is not funny, at that moment freeswitch consumes 98% of CPU time, and
> did not response at all.
>
>
> PS: (Version 1.5.5b git 8d005a4 2013-07-22 14:41:01Z) @FreeBSD 8.2-RELEASE
> (x64)
>
> --
> Nikolay
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From karl at xtronics.com  Thu Aug  8 01:48:55 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Wed, 07 Aug 2013 16:48:55 -0500
Subject: [Freeswitch-users] freeDTM detect if line is in use
In-Reply-To: <CAA4nhyA4y2xKcP=7MgCv_oQ8zKy9ufE8dipVQQsFvnTWDkbQYQ@mail.gmail.com>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>
	<1138329210A14BE4AEE53D7419427A15@PBPC>
	<CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
	<CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
	<3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
	<CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>
	<B782F8192B3E46608CE6666897F86BEB@PBPC>
	<CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com>
	<CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com>
	<CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com>
	<F496A3CCE6994A408714AE2DCE31BAC6@PBPC>
	<0BB9E38C82674637982EDAF416B7B552@PBPC>
	<1599944384-1375845943-cardhu_decombobulator_blackberry.rim.net-1994125046-@b14.c5.bise6.blackberry>
	<5201D22B.4080104@xtronics.com>
	<CAA4nhyA4y2xKcP=7MgCv_oQ8zKy9ufE8dipVQQsFvnTWDkbQYQ@mail.gmail.com>
Message-ID: <5202C0C7.7080502@xtronics.com>

On 08/07/2013 11:18 AM, Moises Silva wrote:
> You can execute CLI commands (which in FreeSWITCH lingo are called APIs) from the dialplan using
> dollar-sign syntax such as:
>
> <condition field="${ftdm_usage(1 1)}" expression="^0$">
> </condition>


Testing from the cli:


freeswitch at internal> ftdm_usage 1 1
0
freeswitch at internal> eval ${ftdm_usage (1 1)}
-ERR invalid span

freeswitch at internal> eval ${ftdm_usage 1 1}
0

I think we need to drop the parenthesis?

Added this to the wiki for the next guy..



Also, is there a command to list all of the available commands from the cli? Something like ls 
/usr/bin would do for us in bash.

Or even a place in the source code that would list all the ftdi commands?

I'm getting things going - finding the learning curve steep in spite of having read all 3 books, but 
I'm getting old..

One thing that would be very useful would be some dialplan log command that would dump all the 
available variables to the cli - so you can at least know what you have to work with.


--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

They can print money; they cannot print wealth.
Wealth only comes from productive behavior.
Printing money only destroys the ability of wealth
producers to produce more wealth. KPS
--------------------------------------------------------------------------------


From nickolayr at gmail.com  Thu Aug  8 01:50:39 2013
From: nickolayr at gmail.com (Nikolay Rogoshchenkov)
Date: Wed, 7 Aug 2013 17:50:39 -0400
Subject: [Freeswitch-users] [CRIT] switch_core_session.c:1705 Thread
	Failure!
In-Reply-To: <CAKbxfG_e4C=SBZteLGtb_9SN1CW94nd9cP5C-+unVpbr4FUtYA@mail.gmail.com>
References: <CAAQCs32D4Gmzft8mtuxnFFgEkU1149C2K6sovW3CziX0HNy7JQ@mail.gmail.com>
	<CAKbxfG_e4C=SBZteLGtb_9SN1CW94nd9cP5C-+unVpbr4FUtYA@mail.gmail.com>
Message-ID: <CAAQCs31tvMHBnf8QVJm+FS7nk+9-PLr7NATUALU8+AKhjDFzxA@mail.gmail.com>

Thank you for your answer Anthony,
No I did not set thread stack size.

Here is my system limits:
*fsnode1# limits*
*Resource limits (current):*
*  cputime              infinity secs*
*  filesize             infinity kB*
*  datasize             33554432 kB*
*  stacksize              524288 kB*
*  coredumpsize         infinity kB*
*  memoryuse            infinity kB*
*  memorylocked         infinity kB*
*  maxprocesses             5547*
*  openfiles               11095*
*  sbsize               infinity bytes*
*  vmemoryuse           infinity kB*
*  pseudo-terminals     infinity*
*  swapuse              infinity kB*
*
*
Also for each call I've used MySQL base (via lua script), may be this is
the reason.

My odbcinst.ini looks like
*
*
*[MySQL]*
*Description=ODBC for MySQL*
*Driver=/usr/local/lib/libmyodbc3.so*
*UsageCount=20003*

and now I have changed to:
*[MySQL]*
*Description=ODBC for MySQL*
*Driver=/usr/local/lib/libmyodbc3_r.so*
*UsageCount=20003*
*Threading=0*
*
*
*as suggested here<http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core#1._Install_unixODBC>
.*


*fsnode1# ldd /usr/local/freeswitch/bin/freeswitch*
*/usr/local/freeswitch/bin/freeswitch:*
*        libodbc.so.2 => /usr/local/lib/libodbc.so.2 (0x80064b000)*
*        libm.so.5 => /lib/libm.so.5 (0x8007bb000)*
*        libfreeswitch.so.1 => /usr/local/freeswitch/lib/libfreeswitch.so.1
(0x8008db000)*
*        libiconv.so.3 => /usr/local/lib/libiconv.so.3 (0x800c56000)*
*        libz.so.5 => /lib/libz.so.5 (0x800e38000)*
*        librt.so.1 => /usr/lib/librt.so.1 (0x800f4d000)*
*        libcrypt.so.5 => /lib/libcrypt.so.5 (0x801052000)*
*        libssl.so.6 => /usr/lib/libssl.so.6 (0x80116b000)*
*        libcrypto.so.6 => /lib/libcrypto.so.6 (0x8012be000)*
*        libncurses.so.8 => /lib/libncurses.so.8 (0x80155e000)*
*        libjpeg.so.11 => /usr/local/lib/libjpeg.so.11 (0x8016ab000)*
*        libthr.so.3 => /lib/libthr.so.3 (0x8017e2000)*
*        libc.so.7 => /lib/libc.so.7 (0x8018fb000)*
*        libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x801b3d000)*
*        libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x801d4d000)*

--
Rogoshchenkov Nikolay


On Wed, Aug 7, 2013 at 4:59 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> The 98% cpu usage is the catalyst for the error, not the other way around.
> You have consumed all your space for threads and the system cannot create
> any more for some reason.
>
>
> FreeBSD has chosen to implement their own threading lib a few times and it
> seems to cause us trouble.
> Did you set the thread stack size to 244 before launching?
>
>
>
>
>
> On Wed, Aug 7, 2013 at 3:47 PM, Nikolay Rogoshchenkov <nickolayr at gmail.com
> > wrote:
>
>> Could you please explain what  this mean in FreeSwitch log???
>>
>> ----------------------------------------------------------------------
>> *[...]*
>> *2013-08-07 23:40:48.011687 [CRIT] switch_core_session.c:1705 Thread
>> Failure!*
>> *2013-08-07 23:40:48.011687 [CRIT] switch_core_session.c:1665 LUKE: I'm
>> hit, but not bad.*
>> *2013-08-07 23:40:48.011687 [CRIT] switch_core_session.c:1666 LUKE'S
>> VOICE: Artoo, see what you can do with it. Hang on back there....*
>> *Green laserfire moves past the beeping little robot as his head turns.
>>  After a few beeps and a twist of his mechanical arm,*
>> *Artoo reduces the max sessions to 101 thus, saving the switch from
>> certain doom.*
>> *2013-08-07 23:40:48.031685 [CRIT] switch_core_session.c:1705 Thread
>> Failure!*
>> *2013-08-07 23:40:48.031685 [CRIT] switch_core_session.c:1665 LUKE: I'm
>> hit, but not bad.*
>> *2013-08-07 23:40:48.031685 [CRIT] switch_core_session.c:1666 LUKE'S
>> VOICE: Artoo, see what you can do with it. Hang on back there....*
>> *Green laserfire moves past the beeping little robot as his head turns.
>>  After a few beeps and a twist of his mechanical arm,*
>> *Artoo reduces the max sessions to 101 thus, saving the switch from
>> certain doom.*
>> *2013-08-07 23:40:48.041688 [CRIT] switch_core_session.c:1705 Thread
>> Failure!*
>> *[...]*
>> *2013-08-07 23:55:38.141571 [CRIT] switch_core_session.c:1705 Thread
>> Failure!*
>> *2013-08-07 23:55:38.151578 [CRIT] switch_core_session.c:1705 Thread
>> Failure!*
>> *2013-08-07 23:55:38.171584 [CRIT] switch_core_session.c:1705 Thread
>> Failure!*
>> *[..]*
>> ----------------------------------------------------------------------
>>
>> It is not funny, at that moment freeswitch consumes 98% of CPU time, and
>> did not response at all.
>>
>>
>> PS: (Version 1.5.5b git 8d005a4 2013-07-22 14:41:01Z) @FreeBSD
>> 8.2-RELEASE (x64)
>>
>> --
>> Nikolay
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 william.king at quentustech.com  Thu Aug  8 02:24:13 2013
From: william.king at quentustech.com (William King)
Date: Wed, 07 Aug 2013 17:24:13 -0500
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
Message-ID: <5202C90D.3020308@quentustech.com>

Is anyone on this list interested in FreeSWITCH support for SMPP for SMS
messages?

For more information about the SMPP protocol checkout:
http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer

If so feel free to contact me on or off this list.
-- 
William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com


From dvl36.ripe.nick at gmail.com  Thu Aug  8 02:46:21 2013
From: dvl36.ripe.nick at gmail.com (Dmitry Lysenko)
Date: Thu, 8 Aug 2013 01:46:21 +0300
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <5202C90D.3020308@quentustech.com>
References: <5202C90D.3020308@quentustech.com>
Message-ID: <CAMjaK3C=eYeNZsYDvjHg57zobNPOGiAVhkVSRyLHNg6mACO9gg@mail.gmail.com>

Yes. I am interested.

Dmitry.


2013/8/8 William King <william.king at quentustech.com>

> Is anyone on this list interested in FreeSWITCH support for SMPP for SMS
> messages?
>
> For more information about the SMPP protocol checkout:
> http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
>
> If so feel free to contact me on or off this list.
> --
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 magnus.kelly at gmail.com  Thu Aug  8 02:47:42 2013
From: magnus.kelly at gmail.com (Magnus Kelly)
Date: Wed, 07 Aug 2013 23:47:42 +0100
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <5202C90D.3020308@quentustech.com>
Message-ID: <CE288D00.66380%magnus.kelly@gmail.com>

I very interested - been awaiting this for some time for voicemail sms
alerting

On 07/08/2013 23:24, "William King" <william.king at quentustech.com> wrote:

>Is anyone on this list interested in FreeSWITCH support for SMPP for SMS
>messages?
>
>For more information about the SMPP protocol checkout:
>http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
>
>If so feel free to contact me on or off this list.
>-- 
>William King
>Senior Engineer
>Quentus Technologies, INC
>1037 NE 65th St Suite 273
>Seattle, WA 98115
>Main:   (877) 211-9337
>Office: (206) 388-4772
>Cell:   (253) 686-5518
>william.king at quentustech.com
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 drk at drkngs.net  Thu Aug  8 02:48:55 2013
From: drk at drkngs.net (Dave R. Kompel)
Date: Wed, 07 Aug 2013 15:48:55 -0700
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <5202C90D.3020308@quentustech.com>
Message-ID: <20130807224855.4fef4f12@mail.tritonwest.net>

William, remember I told you a few months back I needed to talk to you about the message routing? Well that's what I was going to talk to you about. If we could add an action to a chatplan to invoke a managed module, just like in the dialplan, I have a good SMPP library that I could hook up to. I most likely could have it working in like 2 to 3 days.  
   
--Dave
      _____  

  From: William King [mailto:william.king at quentustech.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Wed, 07 Aug 2013 15:24:13 -0700
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP

Is anyone on this list interested in FreeSWITCH support for SMPP for SMS
messages?

For more information about the SMPP protocol checkout:
http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer

If so feel free to contact me on or off this list.
-- 
William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main: (877) 211-9337
Office: (206) 388-4772
Cell: (253) 686-5518
william.king at quentustech.com

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 bdfoster at davri.com  Thu Aug  8 05:00:14 2013
From: bdfoster at davri.com (Brian Foster)
Date: Wed, 7 Aug 2013 21:00:14 -0400
Subject: [Freeswitch-users] freeDTM detect if line is in use
In-Reply-To: <5202C0C7.7080502@xtronics.com>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>
	<1138329210A14BE4AEE53D7419427A15@PBPC>
	<CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
	<CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
	<3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
	<CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>
	<B782F8192B3E46608CE6666897F86BEB@PBPC>
	<CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com>
	<CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com>
	<CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com>
	<F496A3CCE6994A408714AE2DCE31BAC6@PBPC>
	<0BB9E38C82674637982EDAF416B7B552@PBPC>
	<1599944384-1375845943-cardhu_decombobulator_blackberry.rim.net-1994125046-@b14.c5.bise6.blackberry>
	<5201D22B.4080104@xtronics.com>
	<CAA4nhyA4y2xKcP=7MgCv_oQ8zKy9ufE8dipVQQsFvnTWDkbQYQ@mail.gmail.com>
	<5202C0C7.7080502@xtronics.com>
Message-ID: <CAHgk4BWSRFGW7mckLTZOtSTuv4Dm8NJ-YHfM9sz6L0sv5aS2Qw@mail.gmail.com>

Press tab in fs_cli will get you started.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 7, 2013 5:57 PM, "Karl Schmidt" <karl at xtronics.com> wrote:

> On 08/07/2013 11:18 AM, Moises Silva wrote:
> > You can execute CLI commands (which in FreeSWITCH lingo are called APIs)
> from the dialplan using
> > dollar-sign syntax such as:
> >
> > <condition field="${ftdm_usage(1 1)}" expression="^0$">
> > </condition>
>
>
> Testing from the cli:
>
>
> freeswitch at internal> ftdm_usage 1 1
> 0
> freeswitch at internal> eval ${ftdm_usage (1 1)}
> -ERR invalid span
>
> freeswitch at internal> eval ${ftdm_usage 1 1}
> 0
>
> I think we need to drop the parenthesis?
>
> Added this to the wiki for the next guy..
>
>
>
> Also, is there a command to list all of the available commands from the
> cli? Something like ls
> /usr/bin would do for us in bash.
>
> Or even a place in the source code that would list all the ftdi commands?
>
> I'm getting things going - finding the learning curve steep in spite of
> having read all 3 books, but
> I'm getting old..
>
> One thing that would be very useful would be some dialplan log command
> that would dump all the
> available variables to the cli - so you can at least know what you have to
> work with.
>
>
>
> --------------------------------------------------------------------------------
> Karl Schmidt                                  EMail Karl at xtronics.com
> Transtronics, Inc.                              WEB
> http://secure.transtronics.com
> 3209 West 9th Street                             Ph (785) 841-3089
> Lawrence, KS 66049                              FAX (785) 841-0434
>
> They can print money; they cannot print wealth.
> Wealth only comes from productive behavior.
> Printing money only destroys the ability of wealth
> producers to produce more wealth. KPS
>
> --------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Thu Aug  8 05:30:13 2013
From: eidevm5 at gmail.com (Peter)
Date: Thu, 8 Aug 2013 11:30:13 +1000
Subject: [Freeswitch-users] One way audio to CME
In-Reply-To: <1393ACC3-56D8-49B4-86E7-2F5C80435799@gmail.com>
References: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>
	<89239AD2-AE02-4E1C-8692-6E77894D8384@gmail.com>
	<CABkS=xo7NQoTNpeR0dMrLkeBndypfPmTbPEEbUonMu_NJ7wfAA@mail.gmail.com>
	<9C8EC1A4-7DAC-4226-BDD6-F63A9AFD107F@gmail.com>
	<CABkS=xpxRiZ_+SGsjO-CASNSnV66zUxhn_-GmaU+S0_HRS=A1Q@mail.gmail.com>
	<1393ACC3-56D8-49B4-86E7-2F5C80435799@gmail.com>
Message-ID: <CABkS=xrnJFayPeh8du-nScJKKROmHmSOnFiwV_Vug5t9tvh-Vw@mail.gmail.com>

Hi Anthony

Really appreciate the taking your time to look at this.  It's starting to
drive me nuts.

My dialplan for calls to CME is:

  <extension name="incoming-cme">
        <condition field="destination_number" expression="^(3\d\d\d)$">
            <action application="bridge" data="sofia/internal/${
destination_number}@10.10.10.203"/>
        </condition>
  </extension>


My internal sip profile has:

    <param name="rtp-ip" value="10.10.10.206"/>
    <param name="sip-ip" value="10.10.10.206"/>
    <param name="ext-rtp-ip" value="10.1.1.206"/>
    <param name="ext-sip-ip" value="10.1.1.206"/>

The output from:

sofia status

is:

internal    profile                 sip:mod_sofia at 10.1.1.206:5060
RUNNING (0)
external    profile                 sip:mod_sofia at 10.1.1.206:5060
RUNNING (0)

Should internal show 10.10.10.206??

The output from:

sofia status profile internal

shows:

Name                 internal
Domain Name          N/A
Auto-NAT             false
DBName               sofia_reg_internal
Pres Hosts           10.10.10.206,10.1.1.206
Dialplan             XML
Context              public
Challenge Realm      auto_from
RTP-IP               10.10.10.206
Ext-RTP-IP           10.1.1.206
SIP-IP               10.10.10.206
Ext-SIP-IP           10.1.1.206
URL                  sip:mod_sofia at 10.1.1.206:5060
BIND-URL             sip:mod_sofia at 10.1.1.206:5060;maddr=10.10.10.206
HOLD-MUSIC           N/A
OUTBOUND-PROXY       N/A
CODECS IN            iLBC at 30i,PCMU,PCMA,GSM
CODECS OUT           iLBC at 30i,PCMU,PCMA,GSM
TEL-EVENT            101
DTMF-MODE            rfc2833
CNG                  13
SESSION-TO           0
MAX-DIALOG           0
NOMEDIA              false
LATE-NEG             false
PROXY-MEDIA          false
ZRTP-PASSTHRU        false
AGGRESSIVENAT        false
STUN-ENABLED         true
STUN-AUTO-DISABLE    false


The SIP trace from the Freeswitch SBC is at:

http://pastebin.freeswitch.org/21279

I've been playing around with all sorts of different combinations of
SIP/RTP IP settings, but still no closer.

Hope you have some insight.

Thanks

Peter

On Wed, Aug 7, 2013 at 6:31 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:

> Hi Peter,
>
> Your debug shows the invite with via/from/contact/rpid all coming from
> 10.1.1.206, your external side.
> Check your bridge statement, is it using the correct sip profile?
> Check your sip profile SBC internal params rtp-ip & sip-ip, make sure they
> are set correctly to 10.10.10.206
> Paste up your logs from the sbc including sip trace.
>
> Anthony
>
>
> On 7 Aug 2013, at 08:12, Peter <eidevm5 at gmail.com> wrote:
>
> Hi Anthony.
>
> Yes, the SIP profiles are the same for calls going to Kamailio and to
> CME/CUBE.
>
> Note that CME only has one interface, so binding the source interface
> doesn't really make much sense.
>
> Note that I've simplified my set up a little and the phone that was
> registered to CUCM is now registered to CME.  However, the result is still
> the same, ie:  one way audio to the Cisco phone.
>
> You can see the SIP debug from CME at:
>
> http://pastebin.freeswitch.org/21274
>
> The call is coming from 1001 at 10.1.1.204 to 3000 at 10.10.10.203
>
> where
>
> 10.1.1.204  - Freeswitch where SIP clients register to
> 10.1.1.206  - External side of Freeswitch SBC
> 10.10.10.206 - Internal side of Freeswitch SBC
> 10.10.10.203 - CME
>
> Peter
>
>
>
> On Tue, Aug 6, 2013 at 5:13 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>
>> Hi Peter,
>>
>> Because the calls are fine when using Kamailio I'm assuming your sip
>> profiles are fine and you FS SBC config is fine. Are you using the same
>> profiles?
>> Yes you are correct. Have you added the commands? Add them as a first
>> step.
>> Send on a 'debug ccsip messages'
>>
>> Anthony
>>
>>
>>
>> On 6 Aug 2013, at 05:35, Peter <eidevm5 at gmail.com> wrote:
>>
>> Thanks for replying Anthony.
>>
>> Keep in mind that I have very little experience with Cisco products, so I
>> may be missing something fundamental.
>>
>> As far as I can see
>>
>> voice-class sip bind media source-interface ....
>>
>> is just used to bind the SIP or media stream to the appropriate interface
>> on the CUBE.
>>
>> My issue is that the CUBE is trying to initiate the return RTP stream to
>> the external interface (instead of the internal interface) on the
>> Freeswitch SBC.
>>
>> Is my understanding of the sip bind media command correct?
>>
>> Thanks
>>
>> Peter
>>
>>
>> On Mon, Aug 5, 2013 at 5:23 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>>
>>> On cube make sure you specify the source address on your dial-peers
>>> voice-class sip bind media|control
>>> to the correct side. I have seen one way audio when not set.
>>>
>>> On 5 Aug 2013, at 06:29, Peter <eidevm5 at gmail.com> wrote:
>>>
>>> >
>>> >
>>> > I currently have successful two way calls (signalling and media) in
>>> the following setup
>>> >
>>> >
>>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  ->
>>> Kamailio --> Internal Linphone
>>> >
>>> > However, when I try to call a Cisco handset that is registered to
>>> CUCM9 via CME in the following config:
>>> >
>>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  ->
>>> CME ->  CUCM9 -->  Cisco handset
>>> >
>>> > The call signalling appears to be working fine and I can successfully
>>>  initiate a call from each end, but the only RTP stream that is working is
>>> from the external Linphone client to the Cisco handset.
>>> >
>>> > Note that CME is being used as a CUBE device, so all SIP and RTP goes
>>> via it.
>>> >
>>> > Looking at the RTP debugs on CME I can see the problem is that the
>>> "Media Dest Addr" is getting set to the external side of the FS SBC rather
>>> than the internal IP address.
>>> >
>>> >
>>> > I tried setting adding:
>>> >
>>> >             <action application="set"
>>> data="disable_rtp_auto_adjust="true" />
>>> >
>>> > to the dialplan on the SBC, but it made no difference.
>>> >
>>> >
>>> > Any suggestions as to what to check next?
>>> >
>>> > Thanks
>>> >
>>> > Peter
>>> >
>>> >
>>> _________________________________________________________________________
>>>
>>>
>>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at davri.com  Thu Aug  8 05:41:11 2013
From: bdfoster at davri.com (Brian Foster)
Date: Wed, 7 Aug 2013 21:41:11 -0400
Subject: [Freeswitch-users] One way audio to CME
In-Reply-To: <CABkS=xrnJFayPeh8du-nScJKKROmHmSOnFiwV_Vug5t9tvh-Vw@mail.gmail.com>
References: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>
	<89239AD2-AE02-4E1C-8692-6E77894D8384@gmail.com>
	<CABkS=xo7NQoTNpeR0dMrLkeBndypfPmTbPEEbUonMu_NJ7wfAA@mail.gmail.com>
	<9C8EC1A4-7DAC-4226-BDD6-F63A9AFD107F@gmail.com>
	<CABkS=xpxRiZ_+SGsjO-CASNSnV66zUxhn_-GmaU+S0_HRS=A1Q@mail.gmail.com>
	<1393ACC3-56D8-49B4-86E7-2F5C80435799@gmail.com>
	<CABkS=xrnJFayPeh8du-nScJKKROmHmSOnFiwV_Vug5t9tvh-Vw@mail.gmail.com>
Message-ID: <CAHgk4BXmfJ1EDRQiV6zd4+k_Zt10jT476ZAP+i1JX+vwNR_bdg@mail.gmail.com>

Your ip for the external profile will show whatever the ip is set to in the
external profile configs. It doesn't matter what is set in the internal
profile.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 7, 2013 9:36 PM, "Peter" <eidevm5 at gmail.com> wrote:

> Hi Anthony
>
> Really appreciate the taking your time to look at this.  It's starting to
> drive me nuts.
>
> My dialplan for calls to CME is:
>
>   <extension name="incoming-cme">
>         <condition field="destination_number" expression="^(3\d\d\d)$">
>             <action application="bridge" data="sofia/internal/${
> destination_number}@10.10.10.203"/>
>         </condition>
>   </extension>
>
>
> My internal sip profile has:
>
>     <param name="rtp-ip" value="10.10.10.206"/>
>     <param name="sip-ip" value="10.10.10.206"/>
>     <param name="ext-rtp-ip" value="10.1.1.206"/>
>     <param name="ext-sip-ip" value="10.1.1.206"/>
>
> The output from:
>
> sofia status
>
> is:
>
> internal    profile                 sip:mod_sofia at 10.1.1.206:5060
> RUNNING (0)
> external    profile                 sip:mod_sofia at 10.1.1.206:5060
> RUNNING (0)
>
> Should internal show 10.10.10.206??
>
> The output from:
>
> sofia status profile internal
>
> shows:
>
> Name                 internal
> Domain Name          N/A
> Auto-NAT             false
> DBName               sofia_reg_internal
> Pres Hosts           10.10.10.206,10.1.1.206
> Dialplan             XML
> Context              public
> Challenge Realm      auto_from
> RTP-IP               10.10.10.206
> Ext-RTP-IP           10.1.1.206
> SIP-IP               10.10.10.206
> Ext-SIP-IP           10.1.1.206
> URL                  sip:mod_sofia at 10.1.1.206:5060
> BIND-URL             sip:mod_sofia at 10.1.1.206:5060;maddr=10.10.10.206
> HOLD-MUSIC           N/A
> OUTBOUND-PROXY       N/A
> CODECS IN            iLBC at 30i,PCMU,PCMA,GSM
> CODECS OUT           iLBC at 30i,PCMU,PCMA,GSM
> TEL-EVENT            101
> DTMF-MODE            rfc2833
> CNG                  13
> SESSION-TO           0
> MAX-DIALOG           0
> NOMEDIA              false
> LATE-NEG             false
> PROXY-MEDIA          false
> ZRTP-PASSTHRU        false
> AGGRESSIVENAT        false
> STUN-ENABLED         true
> STUN-AUTO-DISABLE    false
>
>
> The SIP trace from the Freeswitch SBC is at:
>
> http://pastebin.freeswitch.org/21279
>
> I've been playing around with all sorts of different combinations of
> SIP/RTP IP settings, but still no closer.
>
> Hope you have some insight.
>
> Thanks
>
> Peter
>
> On Wed, Aug 7, 2013 at 6:31 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>
>> Hi Peter,
>>
>> Your debug shows the invite with via/from/contact/rpid all coming from
>> 10.1.1.206, your external side.
>> Check your bridge statement, is it using the correct sip profile?
>> Check your sip profile SBC internal params rtp-ip & sip-ip, make sure
>> they are set correctly to 10.10.10.206
>> Paste up your logs from the sbc including sip trace.
>>
>> Anthony
>>
>>
>> On 7 Aug 2013, at 08:12, Peter <eidevm5 at gmail.com> wrote:
>>
>> Hi Anthony.
>>
>> Yes, the SIP profiles are the same for calls going to Kamailio and to
>> CME/CUBE.
>>
>> Note that CME only has one interface, so binding the source interface
>> doesn't really make much sense.
>>
>> Note that I've simplified my set up a little and the phone that was
>> registered to CUCM is now registered to CME.  However, the result is still
>> the same, ie:  one way audio to the Cisco phone.
>>
>> You can see the SIP debug from CME at:
>>
>> http://pastebin.freeswitch.org/21274
>>
>> The call is coming from 1001 at 10.1.1.204 to 3000 at 10.10.10.203
>>
>> where
>>
>> 10.1.1.204  - Freeswitch where SIP clients register to
>> 10.1.1.206  - External side of Freeswitch SBC
>> 10.10.10.206 - Internal side of Freeswitch SBC
>> 10.10.10.203 - CME
>>
>> Peter
>>
>>
>>
>> On Tue, Aug 6, 2013 at 5:13 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>>
>>> Hi Peter,
>>>
>>> Because the calls are fine when using Kamailio I'm assuming your sip
>>> profiles are fine and you FS SBC config is fine. Are you using the same
>>> profiles?
>>> Yes you are correct. Have you added the commands? Add them as a first
>>> step.
>>> Send on a 'debug ccsip messages'
>>>
>>> Anthony
>>>
>>>
>>>
>>> On 6 Aug 2013, at 05:35, Peter <eidevm5 at gmail.com> wrote:
>>>
>>> Thanks for replying Anthony.
>>>
>>> Keep in mind that I have very little experience with Cisco products, so
>>> I may be missing something fundamental.
>>>
>>> As far as I can see
>>>
>>> voice-class sip bind media source-interface ....
>>>
>>> is just used to bind the SIP or media stream to the appropriate
>>> interface on the CUBE.
>>>
>>> My issue is that the CUBE is trying to initiate the return RTP stream to
>>> the external interface (instead of the internal interface) on the
>>> Freeswitch SBC.
>>>
>>> Is my understanding of the sip bind media command correct?
>>>
>>> Thanks
>>>
>>> Peter
>>>
>>>
>>> On Mon, Aug 5, 2013 at 5:23 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>>>
>>>> On cube make sure you specify the source address on your dial-peers
>>>> voice-class sip bind media|control
>>>> to the correct side. I have seen one way audio when not set.
>>>>
>>>> On 5 Aug 2013, at 06:29, Peter <eidevm5 at gmail.com> wrote:
>>>>
>>>> >
>>>> >
>>>> > I currently have successful two way calls (signalling and media) in
>>>> the following setup
>>>> >
>>>> >
>>>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  ->
>>>> Kamailio --> Internal Linphone
>>>> >
>>>> > However, when I try to call a Cisco handset that is registered to
>>>> CUCM9 via CME in the following config:
>>>> >
>>>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  ->
>>>> CME ->  CUCM9 -->  Cisco handset
>>>> >
>>>> > The call signalling appears to be working fine and I can successfully
>>>>  initiate a call from each end, but the only RTP stream that is working is
>>>> from the external Linphone client to the Cisco handset.
>>>> >
>>>> > Note that CME is being used as a CUBE device, so all SIP and RTP goes
>>>> via it.
>>>> >
>>>> > Looking at the RTP debugs on CME I can see the problem is that the
>>>> "Media Dest Addr" is getting set to the external side of the FS SBC rather
>>>> than the internal IP address.
>>>> >
>>>> >
>>>> > I tried setting adding:
>>>> >
>>>> >             <action application="set"
>>>> data="disable_rtp_auto_adjust="true" />
>>>> >
>>>> > to the dialplan on the SBC, but it made no difference.
>>>> >
>>>> >
>>>> > Any suggestions as to what to check next?
>>>> >
>>>> > Thanks
>>>> >
>>>> > Peter
>>>> >
>>>> >
>>>> _________________________________________________________________________
>>>>
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Thu Aug  8 06:29:46 2013
From: eidevm5 at gmail.com (Peter)
Date: Thu, 8 Aug 2013 12:29:46 +1000
Subject: [Freeswitch-users] One way audio to CME
In-Reply-To: <CAHgk4BXmfJ1EDRQiV6zd4+k_Zt10jT476ZAP+i1JX+vwNR_bdg@mail.gmail.com>
References: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>
	<89239AD2-AE02-4E1C-8692-6E77894D8384@gmail.com>
	<CABkS=xo7NQoTNpeR0dMrLkeBndypfPmTbPEEbUonMu_NJ7wfAA@mail.gmail.com>
	<9C8EC1A4-7DAC-4226-BDD6-F63A9AFD107F@gmail.com>
	<CABkS=xpxRiZ_+SGsjO-CASNSnV66zUxhn_-GmaU+S0_HRS=A1Q@mail.gmail.com>
	<1393ACC3-56D8-49B4-86E7-2F5C80435799@gmail.com>
	<CABkS=xrnJFayPeh8du-nScJKKROmHmSOnFiwV_Vug5t9tvh-Vw@mail.gmail.com>
	<CAHgk4BXmfJ1EDRQiV6zd4+k_Zt10jT476ZAP+i1JX+vwNR_bdg@mail.gmail.com>
Message-ID: <CABkS=xpyeMCX67Au0j=e8R_n22bvoJDYn6XRf5N8M0VWD+SuDw@mail.gmail.com>

Hi Brian.

I did have the IP set in the external profile, but after removing them it
didn't seem to make any difference, ie:  the SIP ports were listening on
the correct IP address.

Have to say I'm getting a little confused as to when and where to use
sip-ip and ext-sip-ip

Should sip-ip ALWAYS be set to your internal IP address and ext-sip-ip to
your external IP address?

If so, which profiles do they need to be set in?   If they are set in both
the internal and external SIP profiles, does that matter?

Thanks

Peter



On Thu, Aug 8, 2013 at 11:41 AM, Brian Foster <bdfoster at davri.com> wrote:

> Your ip for the external profile will show whatever the ip is set to in
> the external profile configs. It doesn't matter what is set in the internal
> profile.
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana
>
> Sent from a mobile device.
> On Aug 7, 2013 9:36 PM, "Peter" <eidevm5 at gmail.com> wrote:
>
>> Hi Anthony
>>
>> Really appreciate the taking your time to look at this.  It's starting to
>> drive me nuts.
>>
>> My dialplan for calls to CME is:
>>
>>   <extension name="incoming-cme">
>>         <condition field="destination_number" expression="^(3\d\d\d)$">
>>             <action application="bridge" data="sofia/internal/${
>> destination_number}@10.10.10.203"/>
>>         </condition>
>>   </extension>
>>
>>
>> My internal sip profile has:
>>
>>     <param name="rtp-ip" value="10.10.10.206"/>
>>     <param name="sip-ip" value="10.10.10.206"/>
>>     <param name="ext-rtp-ip" value="10.1.1.206"/>
>>     <param name="ext-sip-ip" value="10.1.1.206"/>
>>
>> The output from:
>>
>> sofia status
>>
>> is:
>>
>> internal    profile                 sip:mod_sofia at 10.1.1.206:5060
>> RUNNING (0)
>> external    profile                 sip:mod_sofia at 10.1.1.206:5060
>> RUNNING (0)
>>
>> Should internal show 10.10.10.206??
>>
>> The output from:
>>
>> sofia status profile internal
>>
>> shows:
>>
>> Name                 internal
>> Domain Name          N/A
>> Auto-NAT             false
>> DBName               sofia_reg_internal
>> Pres Hosts           10.10.10.206,10.1.1.206
>> Dialplan             XML
>> Context              public
>> Challenge Realm      auto_from
>> RTP-IP               10.10.10.206
>> Ext-RTP-IP           10.1.1.206
>> SIP-IP               10.10.10.206
>> Ext-SIP-IP           10.1.1.206
>> URL                  sip:mod_sofia at 10.1.1.206:5060
>> BIND-URL             sip:mod_sofia at 10.1.1.206:5060;maddr=10.10.10.206
>> HOLD-MUSIC           N/A
>> OUTBOUND-PROXY       N/A
>> CODECS IN            iLBC at 30i,PCMU,PCMA,GSM
>> CODECS OUT           iLBC at 30i,PCMU,PCMA,GSM
>> TEL-EVENT            101
>> DTMF-MODE            rfc2833
>> CNG                  13
>> SESSION-TO           0
>> MAX-DIALOG           0
>> NOMEDIA              false
>> LATE-NEG             false
>> PROXY-MEDIA          false
>> ZRTP-PASSTHRU        false
>> AGGRESSIVENAT        false
>> STUN-ENABLED         true
>> STUN-AUTO-DISABLE    false
>>
>>
>> The SIP trace from the Freeswitch SBC is at:
>>
>> http://pastebin.freeswitch.org/21279
>>
>> I've been playing around with all sorts of different combinations of
>> SIP/RTP IP settings, but still no closer.
>>
>> Hope you have some insight.
>>
>> Thanks
>>
>> Peter
>>
>> On Wed, Aug 7, 2013 at 6:31 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>>
>>> Hi Peter,
>>>
>>> Your debug shows the invite with via/from/contact/rpid all coming from
>>> 10.1.1.206, your external side.
>>> Check your bridge statement, is it using the correct sip profile?
>>> Check your sip profile SBC internal params rtp-ip & sip-ip, make sure
>>> they are set correctly to 10.10.10.206
>>> Paste up your logs from the sbc including sip trace.
>>>
>>> Anthony
>>>
>>>
>>> On 7 Aug 2013, at 08:12, Peter <eidevm5 at gmail.com> wrote:
>>>
>>> Hi Anthony.
>>>
>>> Yes, the SIP profiles are the same for calls going to Kamailio and to
>>> CME/CUBE.
>>>
>>> Note that CME only has one interface, so binding the source interface
>>> doesn't really make much sense.
>>>
>>> Note that I've simplified my set up a little and the phone that was
>>> registered to CUCM is now registered to CME.  However, the result is still
>>> the same, ie:  one way audio to the Cisco phone.
>>>
>>> You can see the SIP debug from CME at:
>>>
>>> http://pastebin.freeswitch.org/21274
>>>
>>> The call is coming from 1001 at 10.1.1.204 to 3000 at 10.10.10.203
>>>
>>> where
>>>
>>> 10.1.1.204  - Freeswitch where SIP clients register to
>>> 10.1.1.206  - External side of Freeswitch SBC
>>> 10.10.10.206 - Internal side of Freeswitch SBC
>>> 10.10.10.203 - CME
>>>
>>> Peter
>>>
>>>
>>>
>>> On Tue, Aug 6, 2013 at 5:13 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>>>
>>>> Hi Peter,
>>>>
>>>> Because the calls are fine when using Kamailio I'm assuming your sip
>>>> profiles are fine and you FS SBC config is fine. Are you using the same
>>>> profiles?
>>>> Yes you are correct. Have you added the commands? Add them as a first
>>>> step.
>>>> Send on a 'debug ccsip messages'
>>>>
>>>> Anthony
>>>>
>>>>
>>>>
>>>> On 6 Aug 2013, at 05:35, Peter <eidevm5 at gmail.com> wrote:
>>>>
>>>> Thanks for replying Anthony.
>>>>
>>>> Keep in mind that I have very little experience with Cisco products, so
>>>> I may be missing something fundamental.
>>>>
>>>> As far as I can see
>>>>
>>>> voice-class sip bind media source-interface ....
>>>>
>>>> is just used to bind the SIP or media stream to the appropriate
>>>> interface on the CUBE.
>>>>
>>>> My issue is that the CUBE is trying to initiate the return RTP stream
>>>> to the external interface (instead of the internal interface) on the
>>>> Freeswitch SBC.
>>>>
>>>> Is my understanding of the sip bind media command correct?
>>>>
>>>> Thanks
>>>>
>>>> Peter
>>>>
>>>>
>>>> On Mon, Aug 5, 2013 at 5:23 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>>>>
>>>>> On cube make sure you specify the source address on your dial-peers
>>>>> voice-class sip bind media|control
>>>>> to the correct side. I have seen one way audio when not set.
>>>>>
>>>>> On 5 Aug 2013, at 06:29, Peter <eidevm5 at gmail.com> wrote:
>>>>>
>>>>> >
>>>>> >
>>>>> > I currently have successful two way calls (signalling and media) in
>>>>> the following setup
>>>>> >
>>>>> >
>>>>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  ->
>>>>> Kamailio --> Internal Linphone
>>>>> >
>>>>> > However, when I try to call a Cisco handset that is registered to
>>>>> CUCM9 via CME in the following config:
>>>>> >
>>>>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  ->
>>>>> CME ->  CUCM9 -->  Cisco handset
>>>>> >
>>>>> > The call signalling appears to be working fine and I can
>>>>> successfully  initiate a call from each end, but the only RTP stream that
>>>>> is working is from the external Linphone client to the Cisco handset.
>>>>> >
>>>>> > Note that CME is being used as a CUBE device, so all SIP and RTP
>>>>> goes via it.
>>>>> >
>>>>> > Looking at the RTP debugs on CME I can see the problem is that the
>>>>> "Media Dest Addr" is getting set to the external side of the FS SBC rather
>>>>> than the internal IP address.
>>>>> >
>>>>> >
>>>>> > I tried setting adding:
>>>>> >
>>>>> >             <action application="set"
>>>>> data="disable_rtp_auto_adjust="true" />
>>>>> >
>>>>> > to the dialplan on the SBC, but it made no difference.
>>>>> >
>>>>> >
>>>>> > Any suggestions as to what to check next?
>>>>> >
>>>>> > Thanks
>>>>> >
>>>>> > Peter
>>>>> >
>>>>> >
>>>>> _________________________________________________________________________
>>>>>
>>>>>
>>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 gmangudai at gmail.com  Thu Aug  8 07:11:27 2013
From: gmangudai at gmail.com (Vincent Xia)
Date: Thu, 8 Aug 2013 11:11:27 +0800
Subject: [Freeswitch-users] FS behind NAT
Message-ID: <CAKQo6t=dgKE-UEkUCmhOihws4OPmCaFx1ZrawRa1xVHx+NbzVQ@mail.gmail.com>

im having my FS box behind NAT, the network topology is something like:

IP PHONE A(Public IP)<------->NAT<------>FS(Private IP, with a public
IP)<------->IP PHONE B(Private IP)

i was trying to have IP PHONE B able to make call to IP PHONE A and have
two-way talk.

with the default FS configuration when IP PHONE B calls IP PHONE A, there's
only one-way talk, A can hear B but B cannot hear A.

then i decided to use stun by modifying:
vars.xml
<X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
 // this is the default config
<X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
 // this is the default config

internal.xml
<param name="ext-rtp-ip" value="stun:stun.freeswtich.org"/>
<param name="ext-sip-ip" value="stun:stun.freeswtich.org"/>

external.xml
<param name="ext-rtp-ip" value="stun:stun.freeswtich.org"/>
<param name="ext-sip-ip" value="stun:stun.freeswtich.org"/>

but while restarting FS, there's error reporting:
2013-08-08 10:33:58.933096 [ERR] sofia_glue.c:1043 STUN Failed!
stun.freeswitch.org:3478 [Bind Error!]
2013-08-08 10:33:58.933096 [ERR] sofia.c:4074 Failed to get external ip.
2013-08-08 10:33:58.934072 [NOTICE] sofia.c:4754 Started Profile external
[sofia_reg_external]
...
2013-08-08 10:34:00.859927 [ERR] sofia_glue.c:1043 STUN Failed!
stun.freeswitch.org:3478 [Bind Error!]
2013-08-08 10:34:00.859927 [ERR] sofia.c:4074 Failed to get external ip.
2013-08-08 10:34:00.860904 [NOTICE] sofia.c:4754 Started Profile internal
[sofia_reg_internal]

so am i still missing some configuration or anything else was wrong?
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From victor.chukalovskiy at gmail.com  Thu Aug  8 08:00:17 2013
From: victor.chukalovskiy at gmail.com (Victor Chukalovskiy)
Date: Thu, 08 Aug 2013 00:00:17 -0400
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <CAMjaK3C=eYeNZsYDvjHg57zobNPOGiAVhkVSRyLHNg6mACO9gg@mail.gmail.com>
References: <5202C90D.3020308@quentustech.com>
	<CAMjaK3C=eYeNZsYDvjHg57zobNPOGiAVhkVSRyLHNg6mACO9gg@mail.gmail.com>
Message-ID: <520317D1.3000409@gmail.com>

So am I as well as another ClueCon attendee I spoke to.

What is the usage scenario you are looking at?

On 13-08-07 06:46 PM, Dmitry Lysenko wrote:
> Yes. I am interested.
>
> Dmitry.
>
>
> 2013/8/8 William King <william.king at quentustech.com 
> <mailto:william.king at quentustech.com>>
>
>     Is anyone on this list interested in FreeSWITCH support for SMPP
>     for SMS
>     messages?
>
>     For more information about the SMPP protocol checkout:
>     http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
>
>     If so feel free to contact me on or off this list.
>     --
>     William King
>     Senior Engineer
>     Quentus Technologies, INC
>     1037 NE 65th St Suite 273
>     Seattle, WA 98115
>     Main:   (877) 211-9337
>     Office: (206) 388-4772
>     Cell:   (253) 686-5518
>     william.king at quentustech.com <mailto:william.king at quentustech.com>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at davri.com  Thu Aug  8 08:04:00 2013
From: bdfoster at davri.com (Brian Foster)
Date: Thu, 8 Aug 2013 00:04:00 -0400
Subject: [Freeswitch-users] FS behind NAT
In-Reply-To: <CAKQo6t=dgKE-UEkUCmhOihws4OPmCaFx1ZrawRa1xVHx+NbzVQ@mail.gmail.com>
References: <CAKQo6t=dgKE-UEkUCmhOihws4OPmCaFx1ZrawRa1xVHx+NbzVQ@mail.gmail.com>
Message-ID: <CAHgk4BVemnaS+Ub-4OWTDORR8TUFae__Nqqt7iT4vq17xoXybg@mail.gmail.com>

You need to have a seperate sofia profile handling your authenticated
traffic outside the lan (your remote ip phones). We call it doublenat. It
makes no difference if your switch has a private IP and public IP or just
private IP, it works the same.

See:
http://wiki.freeswitch.org/wiki/Example_Offsite_phones

http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios

It generally works well for us. We have an FS server tied to a private IP,
one interface.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 7, 2013 11:17 PM, "Vincent Xia" <gmangudai at gmail.com> wrote:

> im having my FS box behind NAT, the network topology is something like:
>
> IP PHONE A(Public IP)<------->NAT<------>FS(Private IP, with a public
> IP)<------->IP PHONE B(Private IP)
>
> i was trying to have IP PHONE B able to make call to IP PHONE A and have
> two-way talk.
>
> with the default FS configuration when IP PHONE B calls IP PHONE A,
> there's only one-way talk, A can hear B but B cannot hear A.
>
> then i decided to use stun by modifying:
> vars.xml
> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
>  // this is the default config
> <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
>  // this is the default config
>
> internal.xml
> <param name="ext-rtp-ip" value="stun:stun.freeswtich.org"/>
> <param name="ext-sip-ip" value="stun:stun.freeswtich.org"/>
>
> external.xml
> <param name="ext-rtp-ip" value="stun:stun.freeswtich.org"/>
> <param name="ext-sip-ip" value="stun:stun.freeswtich.org"/>
>
> but while restarting FS, there's error reporting:
> 2013-08-08 10:33:58.933096 [ERR] sofia_glue.c:1043 STUN Failed!
> stun.freeswitch.org:3478 [Bind Error!]
> 2013-08-08 10:33:58.933096 [ERR] sofia.c:4074 Failed to get external ip.
> 2013-08-08 10:33:58.934072 [NOTICE] sofia.c:4754 Started Profile external
> [sofia_reg_external]
> ...
> 2013-08-08 10:34:00.859927 [ERR] sofia_glue.c:1043 STUN Failed!
> stun.freeswitch.org:3478 [Bind Error!]
> 2013-08-08 10:34:00.859927 [ERR] sofia.c:4074 Failed to get external ip.
> 2013-08-08 10:34:00.860904 [NOTICE] sofia.c:4754 Started Profile internal
> [sofia_reg_internal]
>
> so am i still missing some configuration or anything else was wrong?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 agtmcgarry at gmail.com  Thu Aug  8 13:01:47 2013
From: agtmcgarry at gmail.com (Anthony McGarry)
Date: Thu, 8 Aug 2013 10:01:47 +0100
Subject: [Freeswitch-users] One way audio to CME
In-Reply-To: <CABkS=xrnJFayPeh8du-nScJKKROmHmSOnFiwV_Vug5t9tvh-Vw@mail.gmail.com>
References: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>
	<89239AD2-AE02-4E1C-8692-6E77894D8384@gmail.com>
	<CABkS=xo7NQoTNpeR0dMrLkeBndypfPmTbPEEbUonMu_NJ7wfAA@mail.gmail.com>
	<9C8EC1A4-7DAC-4226-BDD6-F63A9AFD107F@gmail.com>
	<CABkS=xpxRiZ_+SGsjO-CASNSnV66zUxhn_-GmaU+S0_HRS=A1Q@mail.gmail.com>
	<1393ACC3-56D8-49B4-86E7-2F5C80435799@gmail.com>
	<CABkS=xrnJFayPeh8du-nScJKKROmHmSOnFiwV_Vug5t9tvh-Vw@mail.gmail.com>
Message-ID: <BE863ACF-6D4E-46DD-9F6D-67B182247FC5@gmail.com>

Yes internal should show 10.10.10.206
change ext-sip-ip and ext-rtp-ip to "auto-nat" 
Restart and check sofia status

From the wiki
ext-rtp-ip
This is the IP behind which FreeSWITCH is seen from the Internet, so if FreeSWITCH is behind NAT, this is basically the public IP that should be used for RTP.

On 8 Aug 2013, at 02:30, Peter <eidevm5 at gmail.com> wrote:

> Hi Anthony
> 
> Really appreciate the taking your time to look at this.  It's starting to drive me nuts.
> 
> My dialplan for calls to CME is:
> 
>   <extension name="incoming-cme">
>         <condition field="destination_number" expression="^(3\d\d\d)$">
>             <action application="bridge" data="sofia/internal/${destination_number}@10.10.10.203"/>
>         </condition>
>   </extension>
> 
> 
> My internal sip profile has:
> 
>     <param name="rtp-ip" value="10.10.10.206"/>
>     <param name="sip-ip" value="10.10.10.206"/>
>     <param name="ext-rtp-ip" value="10.1.1.206"/>
>     <param name="ext-sip-ip" value="10.1.1.206"/>
> 
> The output from:
> 
> sofia status
> 
> is:
> 
> internal    profile                 sip:mod_sofia at 10.1.1.206:5060    RUNNING (0)
> external    profile                 sip:mod_sofia at 10.1.1.206:5060    RUNNING (0)
> 
> Should internal show 10.10.10.206??
> 
> The output from:
> 
> sofia status profile internal
> 
> shows:
> 
> Name                 internal
> Domain Name          N/A
> Auto-NAT             false
> DBName               sofia_reg_internal
> Pres Hosts           10.10.10.206,10.1.1.206
> Dialplan             XML
> Context              public
> Challenge Realm      auto_from
> RTP-IP               10.10.10.206
> Ext-RTP-IP           10.1.1.206
> SIP-IP               10.10.10.206
> Ext-SIP-IP           10.1.1.206
> URL                  sip:mod_sofia at 10.1.1.206:5060
> BIND-URL             sip:mod_sofia at 10.1.1.206:5060;maddr=10.10.10.206
> HOLD-MUSIC           N/A
> OUTBOUND-PROXY       N/A
> CODECS IN            iLBC at 30i,PCMU,PCMA,GSM
> CODECS OUT           iLBC at 30i,PCMU,PCMA,GSM
> TEL-EVENT            101
> DTMF-MODE            rfc2833
> CNG                  13
> SESSION-TO           0
> MAX-DIALOG           0
> NOMEDIA              false
> LATE-NEG             false
> PROXY-MEDIA          false
> ZRTP-PASSTHRU        false
> AGGRESSIVENAT        false
> STUN-ENABLED         true
> STUN-AUTO-DISABLE    false
> 
> 
> The SIP trace from the Freeswitch SBC is at:
> 
> http://pastebin.freeswitch.org/21279
> 
> I've been playing around with all sorts of different combinations of SIP/RTP IP settings, but still no closer.
> 
> Hope you have some insight.
> 
> Thanks
> 
> Peter
> 
> On Wed, Aug 7, 2013 at 6:31 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
> Hi Peter,
> 
> Your debug shows the invite with via/from/contact/rpid all coming from 10.1.1.206, your external side. 
> Check your bridge statement, is it using the correct sip profile? 
> Check your sip profile SBC internal params rtp-ip & sip-ip, make sure they are set correctly to 10.10.10.206
> Paste up your logs from the sbc including sip trace.
> 
> Anthony
> 
> 
> On 7 Aug 2013, at 08:12, Peter <eidevm5 at gmail.com> wrote:
> 
>> Hi Anthony.
>> 
>> Yes, the SIP profiles are the same for calls going to Kamailio and to CME/CUBE.
>> 
>> Note that CME only has one interface, so binding the source interface doesn't really make much sense.
>> 
>> Note that I've simplified my set up a little and the phone that was registered to CUCM is now registered to CME.  However, the result is still the same, ie:  one way audio to the Cisco phone.
>> 
>> You can see the SIP debug from CME at:
>> 
>> http://pastebin.freeswitch.org/21274
>> 
>> The call is coming from 1001 at 10.1.1.204 to 3000 at 10.10.10.203
>> 
>> where
>> 
>> 10.1.1.204  - Freeswitch where SIP clients register to
>> 10.1.1.206  - External side of Freeswitch SBC
>> 10.10.10.206 - Internal side of Freeswitch SBC
>> 10.10.10.203 - CME
>> 
>> Peter
>> 
>> 
>> 
>> On Tue, Aug 6, 2013 at 5:13 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
>> Hi Peter,
>> 
>> Because the calls are fine when using Kamailio I'm assuming your sip profiles are fine and you FS SBC config is fine. Are you using the same profiles?
>> Yes you are correct. Have you added the commands? Add them as a first step.
>> Send on a 'debug ccsip messages' 
>> 
>> Anthony 
>> 
>> 
>> 
>> On 6 Aug 2013, at 05:35, Peter <eidevm5 at gmail.com> wrote:
>> 
>>> Thanks for replying Anthony.
>>> 
>>> Keep in mind that I have very little experience with Cisco products, so I may be missing something fundamental.
>>> 
>>> As far as I can see
>>> 
>>> voice-class sip bind media source-interface ....
>>> 
>>> is just used to bind the SIP or media stream to the appropriate interface on the CUBE.
>>> 
>>> My issue is that the CUBE is trying to initiate the return RTP stream to the external interface (instead of the internal interface) on the Freeswitch SBC.
>>> 
>>> Is my understanding of the sip bind media command correct?
>>> 
>>> Thanks
>>> 
>>> Peter
>>> 
>>> 
>>> On Mon, Aug 5, 2013 at 5:23 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
>>> On cube make sure you specify the source address on your dial-peers
>>> voice-class sip bind media|control
>>> to the correct side. I have seen one way audio when not set.
>>> 
>>> On 5 Aug 2013, at 06:29, Peter <eidevm5 at gmail.com> wrote:
>>> 
>>> >
>>> >
>>> > I currently have successful two way calls (signalling and media) in the following setup
>>> >
>>> >
>>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> Kamailio --> Internal Linphone
>>> >
>>> > However, when I try to call a Cisco handset that is registered to CUCM9 via CME in the following config:
>>> >
>>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> CME ->  CUCM9 -->  Cisco handset
>>> >
>>> > The call signalling appears to be working fine and I can successfully  initiate a call from each end, but the only RTP stream that is working is from the external Linphone client to the Cisco handset.
>>> >
>>> > Note that CME is being used as a CUBE device, so all SIP and RTP goes via it.
>>> >
>>> > Looking at the RTP debugs on CME I can see the problem is that the "Media Dest Addr" is getting set to the external side of the FS SBC rather than the internal IP address.
>>> >
>>> >
>>> > I tried setting adding:
>>> >
>>> >             <action application="set" data="disable_rtp_auto_adjust="true" />
>>> >
>>> > to the dialplan on the SBC, but it made no difference.
>>> >
>>> >
>>> > Any suggestions as to what to check next?
>>> >
>>> > Thanks
>>> >
>>> > Peter
>>> >
>>> > _________________________________________________________________________
>>> 
>>> 
>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 agtmcgarry at gmail.com  Thu Aug  8 13:07:46 2013
From: agtmcgarry at gmail.com (Anthony McGarry)
Date: Thu, 8 Aug 2013 10:07:46 +0100
Subject: [Freeswitch-users] One way audio to CME
In-Reply-To: <CABkS=xpyeMCX67Au0j=e8R_n22bvoJDYn6XRf5N8M0VWD+SuDw@mail.gmail.com>
References: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>
	<89239AD2-AE02-4E1C-8692-6E77894D8384@gmail.com>
	<CABkS=xo7NQoTNpeR0dMrLkeBndypfPmTbPEEbUonMu_NJ7wfAA@mail.gmail.com>
	<9C8EC1A4-7DAC-4226-BDD6-F63A9AFD107F@gmail.com>
	<CABkS=xpxRiZ_+SGsjO-CASNSnV66zUxhn_-GmaU+S0_HRS=A1Q@mail.gmail.com>
	<1393ACC3-56D8-49B4-86E7-2F5C80435799@gmail.com>
	<CABkS=xrnJFayPeh8du-nScJKKROmHmSOnFiwV_Vug5t9tvh-Vw@mail.gmail.com>
	<CAHgk4BXmfJ1EDRQiV6zd4+k_Zt10jT476ZAP+i1JX+vwNR_bdg@mail.gmail.com>
	<CABkS=xpyeMCX67Au0j=e8R_n22bvoJDYn6XRf5N8M0VWD+SuDw@mail.gmail.com>
Message-ID: <072F50C1-88FC-47DB-A281-61CF3D90FA05@gmail.com>

sip-ip & rtp-ip should be set per profile. So if you have 5 profiles each profile.xml will have these params set to whatever IP address the profile should use.
In your case 

external profile 
<param name="sip-ip" value="10.1.1.206"/>
<param name="rtp-ip" value="10.1.1.206"/>

internal profile
<param name="sip-ip" value="10.10.10.206"/>
<param name="rtp-ip" value="10.10.10.206"/>

you only need to worry about ext-sip-ip and ext-rtp-ip if there is NAT in the path. 
Better to leave them in each profile to
<param name="ext-sip-ip" value="auto-nat"/>
<param name="ext-rtp-ip" value="auto-nat"/>


On 8 Aug 2013, at 03:29, Peter <eidevm5 at gmail.com> wrote:

> Hi Brian.
> 
> I did have the IP set in the external profile, but after removing them it didn't seem to make any difference, ie:  the SIP ports were listening on the correct IP address.
> 
> Have to say I'm getting a little confused as to when and where to use sip-ip and ext-sip-ip
> 
> Should sip-ip ALWAYS be set to your internal IP address and ext-sip-ip to your external IP address?
> 
> If so, which profiles do they need to be set in?   If they are set in both the internal and external SIP profiles, does that matter?  
> 
> Thanks
> 
> Peter
> 
> 
> 
> On Thu, Aug 8, 2013 at 11:41 AM, Brian Foster <bdfoster at davri.com> wrote:
> Your ip for the external profile will show whatever the ip is set to in the external profile configs. It doesn't matter what is set in the internal profile.
> 
> Thank you,
> 
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana
> 
> Sent from a mobile device.
> 
> On Aug 7, 2013 9:36 PM, "Peter" <eidevm5 at gmail.com> wrote:
> Hi Anthony
> 
> Really appreciate the taking your time to look at this.  It's starting to drive me nuts.
> 
> My dialplan for calls to CME is:
> 
>   <extension name="incoming-cme">
>         <condition field="destination_number" expression="^(3\d\d\d)$">
>             <action application="bridge" data="sofia/internal/${destination_number}@10.10.10.203"/>
>         </condition>
>   </extension>
> 
> 
> My internal sip profile has:
> 
>     <param name="rtp-ip" value="10.10.10.206"/>
>     <param name="sip-ip" value="10.10.10.206"/>
>     <param name="ext-rtp-ip" value="10.1.1.206"/>
>     <param name="ext-sip-ip" value="10.1.1.206"/>
> 
> The output from:
> 
> sofia status
> 
> is:
> 
> internal    profile                 sip:mod_sofia at 10.1.1.206:5060    RUNNING (0)
> external    profile                 sip:mod_sofia at 10.1.1.206:5060    RUNNING (0)
> 
> Should internal show 10.10.10.206??
> 
> The output from:
> 
> sofia status profile internal
> 
> shows:
> 
> Name                 internal
> Domain Name          N/A
> Auto-NAT             false
> DBName               sofia_reg_internal
> Pres Hosts           10.10.10.206,10.1.1.206
> Dialplan             XML
> Context              public
> Challenge Realm      auto_from
> RTP-IP               10.10.10.206
> Ext-RTP-IP           10.1.1.206
> SIP-IP               10.10.10.206
> Ext-SIP-IP           10.1.1.206
> URL                  sip:mod_sofia at 10.1.1.206:5060
> BIND-URL             sip:mod_sofia at 10.1.1.206:5060;maddr=10.10.10.206
> HOLD-MUSIC           N/A
> OUTBOUND-PROXY       N/A
> CODECS IN            iLBC at 30i,PCMU,PCMA,GSM
> CODECS OUT           iLBC at 30i,PCMU,PCMA,GSM
> TEL-EVENT            101
> DTMF-MODE            rfc2833
> CNG                  13
> SESSION-TO           0
> MAX-DIALOG           0
> NOMEDIA              false
> LATE-NEG             false
> PROXY-MEDIA          false
> ZRTP-PASSTHRU        false
> AGGRESSIVENAT        false
> STUN-ENABLED         true
> STUN-AUTO-DISABLE    false
> 
> 
> The SIP trace from the Freeswitch SBC is at:
> 
> http://pastebin.freeswitch.org/21279
> 
> I've been playing around with all sorts of different combinations of SIP/RTP IP settings, but still no closer.
> 
> Hope you have some insight.
> 
> Thanks
> 
> Peter
> 
> On Wed, Aug 7, 2013 at 6:31 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
> Hi Peter,
> 
> Your debug shows the invite with via/from/contact/rpid all coming from 10.1.1.206, your external side. 
> Check your bridge statement, is it using the correct sip profile? 
> Check your sip profile SBC internal params rtp-ip & sip-ip, make sure they are set correctly to 10.10.10.206
> Paste up your logs from the sbc including sip trace.
> 
> Anthony
> 
> 
> On 7 Aug 2013, at 08:12, Peter <eidevm5 at gmail.com> wrote:
> 
>> Hi Anthony.
>> 
>> Yes, the SIP profiles are the same for calls going to Kamailio and to CME/CUBE.
>> 
>> Note that CME only has one interface, so binding the source interface doesn't really make much sense.
>> 
>> Note that I've simplified my set up a little and the phone that was registered to CUCM is now registered to CME.  However, the result is still the same, ie:  one way audio to the Cisco phone.
>> 
>> You can see the SIP debug from CME at:
>> 
>> http://pastebin.freeswitch.org/21274
>> 
>> The call is coming from 1001 at 10.1.1.204 to 3000 at 10.10.10.203
>> 
>> where
>> 
>> 10.1.1.204  - Freeswitch where SIP clients register to
>> 10.1.1.206  - External side of Freeswitch SBC
>> 10.10.10.206 - Internal side of Freeswitch SBC
>> 10.10.10.203 - CME
>> 
>> Peter
>> 
>> 
>> 
>> On Tue, Aug 6, 2013 at 5:13 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
>> Hi Peter,
>> 
>> Because the calls are fine when using Kamailio I'm assuming your sip profiles are fine and you FS SBC config is fine. Are you using the same profiles?
>> Yes you are correct. Have you added the commands? Add them as a first step.
>> Send on a 'debug ccsip messages' 
>> 
>> Anthony 
>> 
>> 
>> 
>> On 6 Aug 2013, at 05:35, Peter <eidevm5 at gmail.com> wrote:
>> 
>>> Thanks for replying Anthony.
>>> 
>>> Keep in mind that I have very little experience with Cisco products, so I may be missing something fundamental.
>>> 
>>> As far as I can see
>>> 
>>> voice-class sip bind media source-interface ....
>>> 
>>> is just used to bind the SIP or media stream to the appropriate interface on the CUBE.
>>> 
>>> My issue is that the CUBE is trying to initiate the return RTP stream to the external interface (instead of the internal interface) on the Freeswitch SBC.
>>> 
>>> Is my understanding of the sip bind media command correct?
>>> 
>>> Thanks
>>> 
>>> Peter
>>> 
>>> 
>>> On Mon, Aug 5, 2013 at 5:23 PM, Anthony McGarry <agtmcgarry at gmail.com> wrote:
>>> On cube make sure you specify the source address on your dial-peers
>>> voice-class sip bind media|control
>>> to the correct side. I have seen one way audio when not set.
>>> 
>>> On 5 Aug 2013, at 06:29, Peter <eidevm5 at gmail.com> wrote:
>>> 
>>> >
>>> >
>>> > I currently have successful two way calls (signalling and media) in the following setup
>>> >
>>> >
>>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> Kamailio --> Internal Linphone
>>> >
>>> > However, when I try to call a Cisco handset that is registered to CUCM9 via CME in the following config:
>>> >
>>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router  -> CME ->  CUCM9 -->  Cisco handset
>>> >
>>> > The call signalling appears to be working fine and I can successfully  initiate a call from each end, but the only RTP stream that is working is from the external Linphone client to the Cisco handset.
>>> >
>>> > Note that CME is being used as a CUBE device, so all SIP and RTP goes via it.
>>> >
>>> > Looking at the RTP debugs on CME I can see the problem is that the "Media Dest Addr" is getting set to the external side of the FS SBC rather than the internal IP address.
>>> >
>>> >
>>> > I tried setting adding:
>>> >
>>> >             <action application="set" data="disable_rtp_auto_adjust="true" />
>>> >
>>> > to the dialplan on the SBC, but it made no difference.
>>> >
>>> >
>>> > Any suggestions as to what to check next?
>>> >
>>> > Thanks
>>> >
>>> > Peter
>>> >
>>> > _________________________________________________________________________
>>> 
>>> 
>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 intralanman at freeswitch.org  Thu Aug  8 13:23:21 2013
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 8 Aug 2013 05:23:21 -0400
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <520317D1.3000409@gmail.com>
References: <5202C90D.3020308@quentustech.com>
	<CAMjaK3C=eYeNZsYDvjHg57zobNPOGiAVhkVSRyLHNg6mACO9gg@mail.gmail.com>
	<520317D1.3000409@gmail.com>
Message-ID: <CAB+Kzh6477KJa4M9PxvrWWh=A4rcJnD7NttvqLcJKBUXctQneQ@mail.gmail.com>

So, to the best of my knowledge,  SMPP is strictly for SMS so you can route
SMS to your clients via FS to SIMPLE / dingaling / etc clients

-Ray
On Aug 7, 2013 11:05 PM, "Victor Chukalovskiy" <
victor.chukalovskiy at gmail.com> wrote:

>  So am I as well as another ClueCon attendee I spoke to.
>
> What is the usage scenario you are looking at?
>
>  On 13-08-07 06:46 PM, Dmitry Lysenko wrote:
>
> Yes. I am interested.
>
>  Dmitry.
>
>
> 2013/8/8 William King <william.king at quentustech.com>
>
>> Is anyone on this list interested in FreeSWITCH support for SMPP for SMS
>> messages?
>>
>> For more information about the SMPP protocol checkout:
>> http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
>>
>> If so feel free to contact me on or off this list.
>> --
>> William King
>> Senior Engineer
>> Quentus Technologies, INC
>> 1037 NE 65th St Suite 273
>> Seattle, WA 98115
>> Main:   (877) 211-9337
>> Office: (206) 388-4772
>> Cell:   (253) 686-5518
>> william.king at quentustech.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 rajat.toshniwal at tekmindz.com  Thu Aug  8 14:47:38 2013
From: rajat.toshniwal at tekmindz.com (Rajat toshniwal)
Date: Thu, 08 Aug 2013 16:17:38 +0530
Subject: [Freeswitch-users] Freeswitch with Digium T316 timed out,
	T316 timed out, 
Message-ID: <5203774A.7040906@tekmindz.com>

Hi

I am trying to deploy freeswitch with Digium TE121 card for my office 
setup, but it is continuously showing Signaling is up and channels are 
down except D channel.
Our Architecture is like
We have freeswitch installed with libpri1.4 and Dahdi.
I am from India and here we are having E1 trunk.

Dahdi Configuration is

cat system.conf
# Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug  7 19:39:07 2013
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
# Dahdi Configuration File
#
# This file is parsed by the Dahdi Configurator, dahdi_cfg
#
# Span 1: WCT1/0 "Wildcard TE121 Card 0" (MASTER)
span=1,1,0,ccs,hdb3,crc4
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31

# Global data

loadzone    = uk
defaultzone    = uk



cat modules
# Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules) 
on Wed Aug  7 19:37:48 2013
# If you edit this file and execute /usr/sbin/dahdi_genconf again,
# your manual changes will be LOST.
wcte12xp
# Xorcom Astribank Devices
xpp_usb


dahdi_hardware
pci:0000:02:08.0     wcte12xp+    d161:8000 Wildcard TE121

dahdi_scan
[1]
active=yes
alarms=OK
description=Wildcard TE121 Card 0
name=WCT1/0
manufacturer=Digium
devicetype=Wildcard TE121 (VPMOCT032)
location=PCI Bus 02 Slot 09
basechan=1
totchans=31
irq=0
type=digital-E1
syncsrc=1
lbo=0 db (CSU)/0-133 feet (DSX-1)
coding_opts=AMI,HDB3
framing_opts=CCS,CRC4
coding=HDB3
framing=CCS/CRC4


Card is properly installed and recognized by Dahdi

Freetdm is compiled with libpri and configuration is like
cat /usr/local/freeswitch/conf/freetdm.conf
[general]
cpu_monitor => yes
cpu_monitoring_interval => 2000 ; monitor usage every 2 seconds
cpu_set_alarm_threshold => 90 ; whenever 90% of global CPU usage is 
reached, trigger the alarm.
cpu_reset_alarm_threshold => 80 ; when the CPU usage decreases at 80%, 
clear the alarm.
cpu_alarm_action => reject,warn ; Start rejecting calls when the CPU 
alarm is triggered and also print warnings.

[span zt myDAHDISpan]
trunk_type => E1
group => g1
b-channel => 1-15
d-channel => 16
b-channel => 17-31


cat /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml
<configuration name="freetdm.conf" description="FreeTDM Configuration">
<settings>
<param name="debug" value="1"/>
</settings>
<libpri_spans>
<span name="myDAHDISpan">
<!-- Log Levels: none, alert, crit, err, warning, notice, info, debug -->
<param name="switch" value="euroisdn"/>
<param name="node" value="cpe"/>
<param name="dialect" value="q931"/>
<param name="debug" value="all"/>
<param name="dialplan" value="XML"/>
<param name="context" value="public"/>
<param name="l1" value="alaw"/>
</span>
</libpri_spans>
</configuration>


Freeswitch logs are showing

2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c10][1:10] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c11][1:11] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c12][1:12] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c13][1:13] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c14][1:14] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c15][1:15] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c17][1:17] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c18][1:18] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c19][1:19] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c20][1:20] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c21][1:21] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c22][1:22] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c23][1:23] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c24][1:24] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c25][1:25] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c26][1:26] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c27][1:27] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c28][1:28] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c29][1:29] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c30][1:30] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c31][1:31] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:23.673847 [WARNING] ftdm_io.c:3022 [s1c5][1:5] Channel 
not opened, proceeding anyway
2013-08-08 15:38:48.613848 [WARNING] ftmod_libpri.c:1975 [s1c6][1:6] -- 
T316 timed out, resending RESTART request
2013-08-08 15:38:48.613848 [WARNING] ftmod_libpri.c:1975 [s1c7][1:7] -- 
T316 timed out, resending RESTART request
2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c8][1:8] -- 
T316 timed out, resending RESTART request
2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c9][1:9] -- 
T316 timed out, resending RESTART request
2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c10][1:10] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c11][1:11] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c12][1:12] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c13][1:13] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c14][1:14] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c15][1:15] 
-- T316 timed out, resending RESTART request
2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c17][1:17] 
-- T316 timed out, resending RESTART request



Libpri logs are showing

2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
DL-DATA request
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
Protocol Discriminator: Q.931 (8)  len=13
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
Message Type: RESTART (70)
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] 
TEI=0 Transmitting N(S)=4, window is open V(A)=2 K=7
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
TEI: 0 State 7(Multi-frame established)
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
V(A)=2, V(S)=4, V(R)=6
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
T200_id=8192, N200=3, T203_id=0
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > [ 
00 01 08 0c 08 02 00 00 46 18 03 a9 83 8d 79 01 80 ]
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
Informational frame:
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
SAPI: 00  C/R: 0 EA: 0
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] >  
TEI: 000        EA: 1
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
N(S): 004   0: 0
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
N(R): 006   P: 0
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 13 
bytes of data
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
Protocol Discriminator: Q.931 (8)  len=13
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
Message Type: RESTART (70)
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
[18 03 a9 83 8d]
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)  Spare: 0  
Exclusive  Dchan: 0
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] 
 >                       ChanSel: As indicated in following octets
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] 
 >                       Ext: 1  Coding: 0  Number Specified  Channel 
Type: 3
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] 
 >                       Ext: 1  Channel: 13 Type: CPE]
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
[79 01 80]
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
Restart Indicator (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated 
Channel (0) ]
2013-08-08 15:41:18.633844 [DEBUG] lpwrap_pri.c:199 [s1c16][1:16] -- 
Starting timer 0x2678478 with timeout 30000 ms
2013-08-08 15:41:18.633844 [WARNING] ftmod_libpri.c:1975 [s1c14][1:14] 
-- T316 timed out, resending RESTART request
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
DL-DATA request
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
Protocol Discriminator: Q.931 (8)  len=13
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
Message Type: RESTART (70)
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] 
TEI=0 Transmitting N(S)=5, window is open V(A)=2 K=7
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
TEI: 0 State 7(Multi-frame established)
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
V(A)=2, V(S)=5, V(R)=6
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
T200_id=8192, N200=3, T203_id=0
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > [ 
00 01 0a 0c 08 02 00 00 46 18 03 a9 83 8e 79 01 80 ]
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
Informational frame:
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
SAPI: 00  C/R: 0 EA: 0
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] >  
TEI: 000        EA: 1
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
N(S): 005   0: 0
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
N(R): 006   P: 0
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 13 
bytes of data
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
Protocol Discriminator: Q.931 (8)  len=13
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
Message Type: RESTART (70)
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
[18 03 a9 83 8e]
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)  Spare: 0  
Exclusive  Dchan: 0
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] 
 >                       ChanSel: As indicated in following octets
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] 
 >                       Ext: 1  Coding: 0  Number Specified  Channel 
Type: 3
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] 
 >                       Ext: 1  Channel: 14 Type: CPE]
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
[79 01 80]
2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16] > 
Restart Indicator (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated 
Channel (0) ]
2013-08-08 15:41:18.633844 [DEBUG] lpwrap_pri.c:199 [s1c16][1:16] -- 
Starting timer 0x26784d8 with timeout 30000 ms
2013-08-08 15:41:18.633844 [WARNING] ftmod_libpri.c:1975 [s1c15][1:15] 
-- T316 timed out, resending RESTART request


ftdm dump 1 n (where n is from 1 to 15 and  17 to 31) is showing state 
as down

freeswitch at dst1> ftdm dump 1 1

span_id: 1
chan_id: n
physical_span_id: 1
physical_chan_id: n
physical_status: ok
physical_status_red: 0
physical_status_yellow: 0
physical_status_rai: 0
physical_status_blue: 0
physical_status_ais: 0
physical_status_general: 0
signaling_status: UP
type: B
state: DOWN
last_state: RESTART
txgain: 0.00
rxgain: 0.00
cid_date:
cid_name:
cid_num:
ani:
aniII:
dnis:
rdnis:
cause: NONE
session: (none)

I have tried many configurations, but I am not able to fix that issue , 
If I put my Pri line into panasonic PBX, it is working properly.

Kindly help me in solving this issue.

Regards
Rajat Toshniwal
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From rajat.toshniwal at tekmindz.com  Thu Aug  8 17:32:04 2013
From: rajat.toshniwal at tekmindz.com (Rajat toshniwal)
Date: Thu, 08 Aug 2013 19:02:04 +0530
Subject: [Freeswitch-users] Freeswitch with Digium T316 timed out,
 T316 timed out, 
In-Reply-To: <5203774A.7040906@tekmindz.com>
References: <5203774A.7040906@tekmindz.com>
Message-ID: <52039DD4.9070109@tekmindz.com>

Forget to mention that I have already tested below mentioned configuration

   1. Go to FS source directory and change dir to this
      path libs/freetdm/src/ftmod/ftmod_libpri
   2. change #define T316_TIMEOUT_MS_DEFAULT to 10000
   3. #define T316_TIMEOUT_MS_MAX to 300000
   4. #define T316_ATTEMPT_LIMIT_DEFAULT to 30 (max. channels in PRI lines)

It didn't work

Also we are using the latest version of Freeswitch 1.2.12 and libpri1.4.14

Regards
Rajat Toshniwal



On Thursday 08 August 2013 04:17 PM, Rajat toshniwal wrote:
> Hi
>
> I am trying to deploy freeswitch with Digium TE121 card for my office
> setup, but it is continuously showing Signaling is up and channels are
> down except D channel.
> Our Architecture is like
> We have freeswitch installed with libpri1.4 and Dahdi.
> I am from India and here we are having E1 trunk.
>
> Dahdi Configuration is
>
> cat system.conf
> # Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug  7 19:39:07 2013
> # If you edit this file and execute /usr/sbin/dahdi_genconf again,
> # your manual changes will be LOST.
> # Dahdi Configuration File
> #
> # This file is parsed by the Dahdi Configurator, dahdi_cfg
> #
> # Span 1: WCT1/0 "Wildcard TE121 Card 0" (MASTER)
> span=1,1,0,ccs,hdb3,crc4
> # termtype: te
> bchan=1-15,17-31
> dchan=16
> echocanceller=mg2,1-15,17-31
>
> # Global data
>
> loadzone    = uk
> defaultzone    = uk
>
>
>
> cat modules
> # Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules)
> on Wed Aug  7 19:37:48 2013
> # If you edit this file and execute /usr/sbin/dahdi_genconf again,
> # your manual changes will be LOST.
> wcte12xp
> # Xorcom Astribank Devices
> xpp_usb
>
>
> dahdi_hardware
> pci:0000:02:08.0     wcte12xp+    d161:8000 Wildcard TE121
>
> dahdi_scan
> [1]
> active=yes
> alarms=OK
> description=Wildcard TE121 Card 0
> name=WCT1/0
> manufacturer=Digium
> devicetype=Wildcard TE121 (VPMOCT032)
> location=PCI Bus 02 Slot 09
> basechan=1
> totchans=31
> irq=0
> type=digital-E1
> syncsrc=1
> lbo=0 db (CSU)/0-133 feet (DSX-1)
> coding_opts=AMI,HDB3
> framing_opts=CCS,CRC4
> coding=HDB3
> framing=CCS/CRC4
>
>
> Card is properly installed and recognized by Dahdi
>
> Freetdm is compiled with libpri and configuration is like
> cat /usr/local/freeswitch/conf/freetdm.conf
> [general]
> cpu_monitor =>  yes
> cpu_monitoring_interval =>  2000 ; monitor usage every 2 seconds
> cpu_set_alarm_threshold =>  90 ; whenever 90% of global CPU usage is
> reached, trigger the alarm.
> cpu_reset_alarm_threshold =>  80 ; when the CPU usage decreases at 80%,
> clear the alarm.
> cpu_alarm_action =>  reject,warn ; Start rejecting calls when the CPU
> alarm is triggered and also print warnings.
>
> [span zt myDAHDISpan]
> trunk_type =>  E1
> group =>  g1
> b-channel =>  1-15
> d-channel =>  16
> b-channel =>  17-31
>
>
> cat /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml
> <configuration name="freetdm.conf" description="FreeTDM Configuration">
> <settings>
> <param name="debug" value="1"/>
> </settings>
> <libpri_spans>
> <span name="myDAHDISpan">
> <!-- Log Levels: none, alert, crit, err, warning, notice, info, debug -->
> <param name="switch" value="euroisdn"/>
> <param name="node" value="cpe"/>
> <param name="dialect" value="q931"/>
> <param name="debug" value="all"/>
> <param name="dialplan" value="XML"/>
> <param name="context" value="public"/>
> <param name="l1" value="alaw"/>
> </span>
> </libpri_spans>
> </configuration>
>
>
> Freeswitch logs are showing
>
> 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c10][1:10]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c11][1:11]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c12][1:12]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c13][1:13]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c14][1:14]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c15][1:15]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c17][1:17]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c18][1:18]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c19][1:19]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c20][1:20]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c21][1:21]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c22][1:22]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c23][1:23]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c24][1:24]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c25][1:25]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c26][1:26]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c27][1:27]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c28][1:28]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c29][1:29]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c30][1:30]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c31][1:31]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:23.673847 [WARNING] ftdm_io.c:3022 [s1c5][1:5] Channel
> not opened, proceeding anyway
> 2013-08-08 15:38:48.613848 [WARNING] ftmod_libpri.c:1975 [s1c6][1:6] --
> T316 timed out, resending RESTART request
> 2013-08-08 15:38:48.613848 [WARNING] ftmod_libpri.c:1975 [s1c7][1:7] --
> T316 timed out, resending RESTART request
> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c8][1:8] --
> T316 timed out, resending RESTART request
> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c9][1:9] --
> T316 timed out, resending RESTART request
> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c10][1:10]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c11][1:11]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c12][1:12]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c13][1:13]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c14][1:14]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c15][1:15]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c17][1:17]
> -- T316 timed out, resending RESTART request
>
>
>
> Libpri logs are showing
>
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> DL-DATA request
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> Protocol Discriminator: Q.931 (8)  len=13
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> Message Type: RESTART (70)
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
> TEI=0 Transmitting N(S)=4, window is open V(A)=2 K=7
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> TEI: 0 State 7(Multi-frame established)
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> V(A)=2, V(S)=4, V(R)=6
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> T200_id=8192, N200=3, T203_id=0
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>  [
> 00 01 08 0c 08 02 00 00 46 18 03 a9 83 8d 79 01 80 ]
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> Informational frame:
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> SAPI: 00  C/R: 0 EA: 0
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> TEI: 000        EA: 1
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> N(S): 004   0: 0
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> N(R): 006   P: 0
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>  13
> bytes of data
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> Protocol Discriminator: Q.931 (8)  len=13
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> Message Type: RESTART (70)
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> [18 03 a9 83 8d]
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)  Spare: 0
> Exclusive  Dchan: 0
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>   >                        ChanSel: As indicated in following octets
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>   >                        Ext: 1  Coding: 0  Number Specified  Channel
> Type: 3
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>   >                        Ext: 1  Channel: 13 Type: CPE]
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> [79 01 80]
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> Restart Indicator (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated
> Channel (0) ]
> 2013-08-08 15:41:18.633844 [DEBUG] lpwrap_pri.c:199 [s1c16][1:16] --
> Starting timer 0x2678478 with timeout 30000 ms
> 2013-08-08 15:41:18.633844 [WARNING] ftmod_libpri.c:1975 [s1c14][1:14]
> -- T316 timed out, resending RESTART request
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> DL-DATA request
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> Protocol Discriminator: Q.931 (8)  len=13
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> Message Type: RESTART (70)
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
> TEI=0 Transmitting N(S)=5, window is open V(A)=2 K=7
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> TEI: 0 State 7(Multi-frame established)
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> V(A)=2, V(S)=5, V(R)=6
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> T200_id=8192, N200=3, T203_id=0
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>  [
> 00 01 0a 0c 08 02 00 00 46 18 03 a9 83 8e 79 01 80 ]
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> Informational frame:
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> SAPI: 00  C/R: 0 EA: 0
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> TEI: 000        EA: 1
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> N(S): 005   0: 0
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> N(R): 006   P: 0
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>  13
> bytes of data
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> Protocol Discriminator: Q.931 (8)  len=13
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> Message Type: RESTART (70)
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> [18 03 a9 83 8e]
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)  Spare: 0
> Exclusive  Dchan: 0
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>   >                        ChanSel: As indicated in following octets
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>   >                        Ext: 1  Coding: 0  Number Specified  Channel
> Type: 3
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>   >                        Ext: 1  Channel: 14 Type: CPE]
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> [79 01 80]
> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
> Restart Indicator (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated
> Channel (0) ]
> 2013-08-08 15:41:18.633844 [DEBUG] lpwrap_pri.c:199 [s1c16][1:16] --
> Starting timer 0x26784d8 with timeout 30000 ms
> 2013-08-08 15:41:18.633844 [WARNING] ftmod_libpri.c:1975 [s1c15][1:15]
> -- T316 timed out, resending RESTART request
>
>
> ftdm dump 1 n (where n is from 1 to 15 and  17 to 31) is showing state
> as down
>
> freeswitch at dst1>  ftdm dump 1 1
>
> span_id: 1
> chan_id: n
> physical_span_id: 1
> physical_chan_id: n
> physical_status: ok
> physical_status_red: 0
> physical_status_yellow: 0
> physical_status_rai: 0
> physical_status_blue: 0
> physical_status_ais: 0
> physical_status_general: 0
> signaling_status: UP
> type: B
> state: DOWN
> last_state: RESTART
> txgain: 0.00
> rxgain: 0.00
> cid_date:
> cid_name:
> cid_num:
> ani:
> aniII:
> dnis:
> rdnis:
> cause: NONE
> session: (none)
>
> I have tried many configurations, but I am not able to fix that issue ,
> If I put my Pri line into panasonic PBX, it is working properly.
>
> Kindly help me in solving this issue.
>
> Regards
> Rajat Toshniwal
> ----------------------------------------------------------------------------------
> Disclaimer: The information contained in this communication is confidential, private, proprietary, or otherwise privileged and is intended only for the use of the addressee. Unauthorized use, disclosure, distribution or copying is strictly prohibited and may be unlawful. If you have received this communication in error, please delete this message and notify the sender immediately - Samin TekMindz India Pvt. Ltd.
> ----------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>


----------------------------------------------------------------------------------
Disclaimer: The information contained in this communication is confidential, private, proprietary, or otherwise privileged and is intended only for the use of the addressee. Unauthorized use, disclosure, distribution or copying is strictly prohibited and may be unlawful. If you have received this communication in error, please delete this message and notify the sender immediately - Samin TekMindz India Pvt. Ltd.
----------------------------------------------------------------------------------
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From jaasmailing at gmail.com  Thu Aug  8 18:15:39 2013
From: jaasmailing at gmail.com (Carlo Dimaggio)
Date: Thu, 08 Aug 2013 16:15:39 +0200
Subject: [Freeswitch-users] Bypass_media_after_bridge doesn't works
In-Reply-To: <A514647D-DA5C-4647-987B-95029C5640B3@jerris.com>
References: <51EE6B35.1000007@gmail.com>
	<A514647D-DA5C-4647-987B-95029C5640B3@jerris.com>
Message-ID: <5203A80B.6050104@gmail.com>

Hi Michael,

After your mail, I did several test and looked into the packets, but the 
phones doesn't insert any reference to the NAT_ADDRESS.
Freeswitch adds the parameter "received=NAT_ADDRESS" in the via header 
(for the aggressive-nat-detection) and sometimes changes the SDP connect 
information to the NAT_ADDRESS instead of the PHONE_PRIVATE_ADDRESS.

Could the bypass_media_after_bridge fail in some calls?

Regards,



Il 23/07/13 15.50, Michael Jerris ha scritto:
> in bypass media, we pass along the sdp the other side sends.  If they 
> are sending the wrong address, we will happily send that along.  If 
> the other end can't figure it out with its own nat fixing, it won't work.
>
>
> On Jul 23, 2013, at 7:38 AM, Carlo Dimaggio <jaasmailing at gmail.com 
> <mailto:jaasmailing at gmail.com>> wrote:
>
>> Hi all,
>>
>> I have an installation with several phones (yealink) behind NAT and 
>> freeswitch withaggressive-nat-detection=true in internal SIP Profile.
>> In dialplan I set bypass_media_after_bridgein order to eliminate 
>> on-way audio problems (and use less bandwidth), but I'm encountering 
>> a strange issue.
>> When I call two extensions, sometimes freeswitch bypassthe media 
>> between them (so the SDP sent by freeswitch have the private IP) and 
>> sometimes freeswitch send the SDP with the NAT IP(in this case there 
>> is no audio).
>>
>> What could be the problem?
>> Is this a bug or my misconfiguration?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveu at coppice.org  Thu Aug  8 18:57:45 2013
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 08 Aug 2013 22:57:45 +0800
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <CAB+Kzh6477KJa4M9PxvrWWh=A4rcJnD7NttvqLcJKBUXctQneQ@mail.gmail.com>
References: <5202C90D.3020308@quentustech.com>
	<CAMjaK3C=eYeNZsYDvjHg57zobNPOGiAVhkVSRyLHNg6mACO9gg@mail.gmail.com>
	<520317D1.3000409@gmail.com>
	<CAB+Kzh6477KJa4M9PxvrWWh=A4rcJnD7NttvqLcJKBUXctQneQ@mail.gmail.com>
Message-ID: <5203B1E9.6020300@coppice.org>

SMPP is strictly for SMS, and you need a direct connect to the network's 
SMSC to use it.

This sudden interest in SMPP is puzzling. Several times over the life of 
FreeSwitch I've offered to resurrect a SMPP platform that was heavily 
used about 12 to 15 years ago, and make it open source. That platform 
has SGIP and a couple of other protocols, too. I've never had enough 
positive response to make me go ahead with the work.

Regards,
Steve

On 08/08/2013 05:23 PM, Raymond Chandler wrote:
>
> So, to the best of my knowledge,  SMPP is strictly for SMS so you can 
> route SMS to your clients via FS to SIMPLE / dingaling / etc clients
>
> -Ray
>
> On Aug 7, 2013 11:05 PM, "Victor Chukalovskiy" 
> <victor.chukalovskiy at gmail.com <mailto:victor.chukalovskiy at gmail.com>> 
> wrote:
>
>     So am I as well as another ClueCon attendee I spoke to.
>
>     What is the usage scenario you are looking at?
>     On 13-08-07 06:46 PM, Dmitry Lysenko wrote:
>>     Yes. I am interested.
>>
>>     Dmitry.
>>
>>
>>     2013/8/8 William King <william.king at quentustech.com
>>     <mailto:william.king at quentustech.com>>
>>
>>         Is anyone on this list interested in FreeSWITCH support for
>>         SMPP for SMS
>>         messages?
>>
>>         For more information about the SMPP protocol checkout:
>>         http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
>>
>>         If so feel free to contact me on or off this list.
>>         --
>>         William King
>>         Senior Engineer
>>         Quentus Technologies, INC
>>         1037 NE 65th St Suite 273
>>         Seattle, WA 98115
>>         Main: (877) 211-9337 <tel:%28877%29%20211-9337>
>>         Office: (206) 388-4772 <tel:%28206%29%20388-4772>
>>         Cell: (253) 686-5518 <tel:%28253%29%20686-5518>
>>         william.king at quentustech.com
>>         <mailto:william.king at quentustech.com>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.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
>>
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.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
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 stefano at i360tecnologia.com.br  Thu Aug  8 16:28:10 2013
From: stefano at i360tecnologia.com.br (Stefano)
Date: Thu, 8 Aug 2013 09:28:10 -0300
Subject: [Freeswitch-users] RES:  Potential FreeSWITCH support for SMPP
In-Reply-To: <5202C90D.3020308@quentustech.com>
References: <5202C90D.3020308@quentustech.com>
Message-ID: <024d01ce9432$c0d281e0$427785a0$@i360tecnologia.com.br>

Yes, I need SMPP on FreeSWITCH.


Att,



-----Mensagem original-----
De: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] Em nome de William
King
Enviada em: quarta-feira, 7 de agosto de 2013 19:24
Para: FreeSWITCH Users Help
Assunto: [Freeswitch-users] Potential FreeSWITCH support for SMPP

Is anyone on this list interested in FreeSWITCH support for SMPP for SMS
messages?

For more information about the SMPP protocol checkout:
http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer

If so feel free to contact me on or off this list.
--
William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 stuart.mills3 at btopenworld.com  Thu Aug  8 20:06:16 2013
From: stuart.mills3 at btopenworld.com (Stuart Mills)
Date: Thu, 8 Aug 2013 17:06:16 +0100
Subject: [Freeswitch-users] Call at once and XML-CDR
In-Reply-To: <1599944384-1375845943-cardhu_decombobulator_blackberry.rim.net-1994125046-@b14.c5.bise6.blackberry>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC><000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com><1138329210A14BE4AEE53D7419427A15@PBPC><CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com><CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com><3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC><CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com><B782F8192B3E46608CE6666897F86BEB@PBPC><CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com><CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com><CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com><F496A3CCE6994A408714AE2DCE31BAC6@PBPC><0BB9E38C82674637982EDAF416B7B552@PBPC>
	<1599944384-1375845943-cardhu_decombobulator_blackberry.rim.net-1994125046-@b14.c5.bise6.blackberry>
Message-ID: <B5969BB0C6B54895851816D7C7994256@PBPC>

I think I may have some more information regarding the app_log stopping it's 
output to XML-CDR files.

I've just managed to re-create the issue using a single call -
{aleg_uuid=801ec328-0040-11e3-9180-076be430ad07,module_id=1712,originate_timeout=15}[leg_timeout=15]sofia/gateway/sip_gateway/447535612620

Could any of my parameters be messing up the app_log output to the xml cdr 
files?

Call at once still seems to split the app_log across 2 cdr's though which 
seems a bit odd to me.

Stuart

-----Original Message----- 
From: lloyd.aloysius at gmail.com
Sent: Wednesday, August 07, 2013 4:25 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Call at once and XML-CDR

Freeswitch pastebin link below

http://pastebin.freeswitch.org/

Lloyd
Sent from my BlackBerry device on the Rogers Wireless Network

-----Original Message-----
From: "Stuart Mills" <stuart.mills3 at btopenworld.com>
Sender: freeswitch-users-bounces at lists.freeswitch.org
Date: Wed, 7 Aug 2013 01:15:53
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] Call at once and XML-CDR

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
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http://www.freeswitch.org

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
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http://www.freeswitch.org 



From msc at freeswitch.org  Thu Aug  8 20:33:50 2013
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Aug 2013 09:33:50 -0700
Subject: [Freeswitch-users] freeDTM detect if line is in use
In-Reply-To: <5202C0C7.7080502@xtronics.com>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<000001403f6a98c1-d082f485-5034-425b-bf7e-510f87f1ef0d-000000@email.amazonses.com>
	<1138329210A14BE4AEE53D7419427A15@PBPC>
	<CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
	<CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
	<3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
	<CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>
	<B782F8192B3E46608CE6666897F86BEB@PBPC>
	<CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com>
	<CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com>
	<CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com>
	<F496A3CCE6994A408714AE2DCE31BAC6@PBPC>
	<0BB9E38C82674637982EDAF416B7B552@PBPC>
	<1599944384-1375845943-cardhu_decombobulator_blackberry.rim.net-1994125046-@b14.c5.bise6.blackberry>
	<5201D22B.4080104@xtronics.com>
	<CAA4nhyA4y2xKcP=7MgCv_oQ8zKy9ufE8dipVQQsFvnTWDkbQYQ@mail.gmail.com>
	<5202C0C7.7080502@xtronics.com>
Message-ID: <CAKzWOxVZF+h002vGa75zZ59+m4V039pnojoxPz6JFv8jYdtkkw@mail.gmail.com>

In fs_cli use "/help" to show fs_cli commands. Use "help" to show
FreeSWITCH commands. Like Brian said, if you just press <tab> you'll get a
quick list of all the tab-complete-enabled commands. Using the "help"
command is good because it will give you a quick description of each
command.

-MC


On Wed, Aug 7, 2013 at 2:48 PM, Karl Schmidt <karl at xtronics.com> wrote:

> On 08/07/2013 11:18 AM, Moises Silva wrote:
> > You can execute CLI commands (which in FreeSWITCH lingo are called APIs)
> from the dialplan using
> > dollar-sign syntax such as:
> >
> > <condition field="${ftdm_usage(1 1)}" expression="^0$">
> > </condition>
>
>
> Testing from the cli:
>
>
> freeswitch at internal> ftdm_usage 1 1
> 0
> freeswitch at internal> eval ${ftdm_usage (1 1)}
> -ERR invalid span
>
> freeswitch at internal> eval ${ftdm_usage 1 1}
> 0
>
> I think we need to drop the parenthesis?
>
> Added this to the wiki for the next guy..
>
>
>
> Also, is there a command to list all of the available commands from the
> cli? Something like ls
> /usr/bin would do for us in bash.
>
> Or even a place in the source code that would list all the ftdi commands?
>
> I'm getting things going - finding the learning curve steep in spite of
> having read all 3 books, but
> I'm getting old..
>
> One thing that would be very useful would be some dialplan log command
> that would dump all the
> available variables to the cli - so you can at least know what you have to
> work with.
>
>
>
> --------------------------------------------------------------------------------
> Karl Schmidt                                  EMail Karl at xtronics.com
> Transtronics, Inc.                              WEB
> http://secure.transtronics.com
> 3209 West 9th Street                             Ph (785) 841-3089
> Lawrence, KS 66049                              FAX (785) 841-0434
>
> They can print money; they cannot print wealth.
> Wealth only comes from productive behavior.
> Printing money only destroys the ability of wealth
> producers to produce more wealth. KPS
>
> --------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From msc at freeswitch.org  Thu Aug  8 20:37:11 2013
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Aug 2013 09:37:11 -0700
Subject: [Freeswitch-users] FS behind NAT
In-Reply-To: <CAKQo6t=dgKE-UEkUCmhOihws4OPmCaFx1ZrawRa1xVHx+NbzVQ@mail.gmail.com>
References: <CAKQo6t=dgKE-UEkUCmhOihws4OPmCaFx1ZrawRa1xVHx+NbzVQ@mail.gmail.com>
Message-ID: <CAKzWOxVeTjLMxBRxY1iL=Ypnaa0CdJxYNDS1XN6ugGg58LPhKw@mail.gmail.com>

On Wed, Aug 7, 2013 at 8:11 PM, Vincent Xia <gmangudai at gmail.com> wrote:

> im having my FS box behind NAT, the network topology is something like:
>
> IP PHONE A(Public IP)<------->NAT<------>FS(Private IP, with a public
> IP)<------->IP PHONE B(Private IP)
>
> i was trying to have IP PHONE B able to make call to IP PHONE A and have
> two-way talk.
>
> with the default FS configuration when IP PHONE B calls IP PHONE A,
> there's only one-way talk, A can hear B but B cannot hear A.
>
> then i decided to use stun by modifying:
> vars.xml
> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/>
>  // this is the default config
> <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/>
>  // this is the default config
>
> internal.xml
> <param name="ext-rtp-ip" value="stun:stun.freeswtich.org"/>
> <param name="ext-sip-ip" value="stun:stun.freeswtich.org"/>
>
> external.xml
> <param name="ext-rtp-ip" value="stun:stun.freeswtich.org"/>
> <param name="ext-sip-ip" value="stun:stun.freeswtich.org"/>
>

You also may have a typo: you have "stun.freeswtich.org" instead of "
stun.freeswitch.org"

-MC



>
> but while restarting FS, there's error reporting:
> 2013-08-08 10:33:58.933096 [ERR] sofia_glue.c:1043 STUN Failed!
> stun.freeswitch.org:3478 [Bind Error!]
> 2013-08-08 10:33:58.933096 [ERR] sofia.c:4074 Failed to get external ip.
> 2013-08-08 10:33:58.934072 [NOTICE] sofia.c:4754 Started Profile external
> [sofia_reg_external]
> ...
> 2013-08-08 10:34:00.859927 [ERR] sofia_glue.c:1043 STUN Failed!
> stun.freeswitch.org:3478 [Bind Error!]
> 2013-08-08 10:34:00.859927 [ERR] sofia.c:4074 Failed to get external ip.
> 2013-08-08 10:34:00.860904 [NOTICE] sofia.c:4754 Started Profile internal
> [sofia_reg_internal]
>
> so am i still missing some configuration or anything else was wrong?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From ivan at c3i.bg  Thu Aug  8 21:01:53 2013
From: ivan at c3i.bg (Ivan)
Date: Thu, 08 Aug 2013 20:01:53 +0300
Subject: [Freeswitch-users] FreeSwitch and Yealink
In-Reply-To: <20130807060000.E564FF4002@mail.mydcs.ca>
References: <20130807044439.9A5F8F4002@mail.mydcs.ca> <5201DF92.9010800@c3i.bg>
	<20130807060000.E564FF4002@mail.mydcs.ca>
Message-ID: <5203CF01.8050501@c3i.bg>

You have to set the variable in the dialplan, before bridging. You can 
also set it directly in the bridge command.

More details at:

http://wiki.freeswitch.org/wiki/Codec_Negotiation

But the only way to really know what is going on is to enable siptrace 
on your sofia profile and see what is offered before your endpoint 
hangups (if you post your debug, please post on freeswitch's pastebin).

ivan


On 08/07/2013 08:59 AM, Paul wrote:
> Ivan thanks for your reply.
>
> I'm willing to try anything :)
>
> Where would I set those variables? In the dialplan out to the sip trunk?
>
> Thanks
>
> Paul
>
> On Tue, 6 Aug, 2013 at 10:48 PM, Ivan <ivan at c3i.bg> wrote:
>> Hangups with the cause you describe can be caused by a failed codec
>> negotiation. Are you sure it's not the case ? Does the remote phone
>> send a sip "not acceptable here" ? If needed you can force the
>> selection of the "outgoing" codec:
>> http://wiki.freeswitch.org/wiki/Channel_Variables#absolute_codec_string ivan
>> On 08/07/2013 07:44 AM, Paul wrote:
>>
>>     Hi guys, Has anyone had any issues using FreeSwitch with Yealink
>>     phones? My phones connect to FS via openvpn tunnel. All incoming
>>     calls work no problem, call comes through phones ring everyone can
>>     hear each other no issues, having a very strange issue though on
>>     the outgoing calls. As soon as the destination party picks up
>>     (this external calls) the call hangs up. short FS LOG:
>>     switch_ivr_bridge.c:475 Channel [sofia/internal/105 at 10.0.0.34] has
>>     been answered sofia.c:6528 Hangup sofia/internal/105 at 10.0.0.34
>>     [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup
>>     sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] So
>>     FS thinks the phone sent a BYE packet (which I can see with
>>     siptrace) but the phone's timer keeps going as if it thinks the
>>     call is supposed to keep going. Internal extension to extension
>>     works fine (even if the extensions are at a different physical
>>     location and subnet). I setup a second account to one of my
>>     asterisk servers and outgoing/incoming work just fine, so it seems
>>     this strange combination of FS and Yealink ... does it on 2 models
>>     T32G and T38G (only phones I have). I have updated firwmare to
>>     their latest version (which in the comments say freeswitch ready)
>>     Wondering if anyone else had any experience with these, or has
>>     some thoughts? Thanks Paul
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org http://www.freeswitchsolutions.com
>>     
>>      Official FreeSWITCH Sites
>>     http://www.freeswitch.org http://wiki.freeswitch.org
>>     http://www.cluecon.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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
>> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
>> CudaTel Communication Server  Official
>> FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org
>> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 karl at xtronics.com  Thu Aug  8 23:44:46 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Thu, 08 Aug 2013 14:44:46 -0500
Subject: [Freeswitch-users] freeDTM detect if line is in use
In-Reply-To: <CAKzWOxVZF+h002vGa75zZ59+m4V039pnojoxPz6JFv8jYdtkkw@mail.gmail.com>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<1138329210A14BE4AEE53D7419427A15@PBPC>
	<CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
	<CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
	<3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
	<CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>
	<B782F8192B3E46608CE6666897F86BEB@PBPC>
	<CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com>
	<CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com>
	<CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com>
	<F496A3CCE6994A408714AE2DCE31BAC6@PBPC>
	<0BB9E38C82674637982EDAF416B7B552@PBPC>
	<1599944384-1375845943-cardhu_decombobulator_blackberry.rim.net-1994125046-@b14.c5.bise6.blackberry>
	<5201D22B.4080104@xtronics.com>
	<CAA4nhyA4y2xKcP=7MgCv_oQ8zKy9ufE8dipVQQsFvnTWDkbQYQ@mail.gmail.com>
	<5202C0C7.7080502@xtronics.com>
	<CAKzWOxVZF+h002vGa75zZ59+m4V039pnojoxPz6JFv8jYdtkkw@mail.gmail.com>
Message-ID: <5203F52E.7080502@xtronics.com>

On 08/08/2013 11:33 AM, Michael Collins wrote:
> In fs_cli use "/help" to show fs_cli commands. Use "help" to show FreeSWITCH commands. Like Brian
> said, if you just press <tab> you'll get a quick list of all the tab-complete-enabled commands.
> Using the "help" command is good because it will give you a quick description of each command.
>
> -MC

I could have sworn that I had tried that already and it didn't work - it certainly works now (and I 
feel a bit stupid) - I had been using up-arrow and F6 already.

I've been testing various SIP clients from Debian stable - both ekiga and qutecom fail to do call 
transfers (works well with linphone). Didn't find anything with google - is there more than one way 
for a SIP client to push a call transfer? Some setup bit on the freeswitch end?






--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

There is no security in life, only opportunity - Mark Twain
--------------------------------------------------------------------------------


From msc at freeswitch.org  Fri Aug  9 00:06:29 2013
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 8 Aug 2013 13:06:29 -0700
Subject: [Freeswitch-users] freeDTM detect if line is in use
In-Reply-To: <5203F52E.7080502@xtronics.com>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<1138329210A14BE4AEE53D7419427A15@PBPC>
	<CAGvuWYY0AtO5WVo6hDBbouu4VhP6bbcFvPgQJ5K6Jyo1TRPZ6Q@mail.gmail.com>
	<CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
	<3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
	<CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>
	<B782F8192B3E46608CE6666897F86BEB@PBPC>
	<CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com>
	<CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com>
	<CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com>
	<F496A3CCE6994A408714AE2DCE31BAC6@PBPC>
	<0BB9E38C82674637982EDAF416B7B552@PBPC>
	<1599944384-1375845943-cardhu_decombobulator_blackberry.rim.net-1994125046-@b14.c5.bise6.blackberry>
	<5201D22B.4080104@xtronics.com>
	<CAA4nhyA4y2xKcP=7MgCv_oQ8zKy9ufE8dipVQQsFvnTWDkbQYQ@mail.gmail.com>
	<5202C0C7.7080502@xtronics.com>
	<CAKzWOxVZF+h002vGa75zZ59+m4V039pnojoxPz6JFv8jYdtkkw@mail.gmail.com>
	<5203F52E.7080502@xtronics.com>
Message-ID: <CAKzWOxUdSGDoesNWXtHToMJ=rJZmT8i1D1T7b8mokQjjJz-84Q@mail.gmail.com>

The default dialplan has an example of using *1 or *4 to do a blind or attn
xfer respectively. Look for "Local_Extension" in conf/dialplan/default.xml.


-MC


On Thu, Aug 8, 2013 at 12:44 PM, Karl Schmidt <karl at xtronics.com> wrote:

> On 08/08/2013 11:33 AM, Michael Collins wrote:
> > In fs_cli use "/help" to show fs_cli commands. Use "help" to show
> FreeSWITCH commands. Like Brian
> > said, if you just press <tab> you'll get a quick list of all the
> tab-complete-enabled commands.
> > Using the "help" command is good because it will give you a quick
> description of each command.
> >
> > -MC
>
> I could have sworn that I had tried that already and it didn't work - it
> certainly works now (and I
> feel a bit stupid) - I had been using up-arrow and F6 already.
>
> I've been testing various SIP clients from Debian stable - both ekiga and
> qutecom fail to do call
> transfers (works well with linphone). Didn't find anything with google -
> is there more than one way
> for a SIP client to push a call transfer? Some setup bit on the freeswitch
> end?
>
>
>
>
>
>
>
> --------------------------------------------------------------------------------
> Karl Schmidt                                  EMail Karl at xtronics.com
> Transtronics, Inc.                              WEB
> http://secure.transtronics.com
> 3209 West 9th Street                             Ph (785) 841-3089
> Lawrence, KS 66049                              FAX (785) 841-0434
>
> There is no security in life, only opportunity - Mark Twain
>
> --------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From bdfoster at davri.com  Fri Aug  9 00:38:28 2013
From: bdfoster at davri.com (Brian Foster)
Date: Thu, 8 Aug 2013 16:38:28 -0400
Subject: [Freeswitch-users] OpenWRT Router changes public IP to it's private
	IP
Message-ID: <CAHgk4BVjSodyDpHYnuX_tE_DM=NbfaxzOnLruk=joNwhsq0XyQ@mail.gmail.com>

We've got an openwrt router at a site, it's private IP is 10.0.0.1.
Whenever we get an inbound call on the external profile (and probably
outbound calls to, haven't checked yet), it changes the contact IP to the
router's private IP, so that every call we get looks like it's coming from
thr router. We've had some intermittent audio problems recently and we're
trying to narrow things down a bit. Is this the expected behavior for NAT?

Sorry might seem like a dumb question, but my job entails much more than
keeping the company's server's happy and I'm starting to lose my grip and
I'm definitely not experienced with NAT.

All of my routers in the past have given FS the public IP of the contacting
server. I have web servers behind the same NAT and doing the same thing,
showing the private IP of the router instead of showing the public IP of
the client. So I know it's not an issue with FS. Just wanted to see if
anyone has exoerienced this before. Makes it difficult to use fail2ban on
my servers, as it continually jails my router.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
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From jleung at v10networks.ca  Fri Aug  9 00:55:32 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Thu, 8 Aug 2013 13:55:32 -0700
Subject: [Freeswitch-users] OpenWRT Router changes public IP to it's
	private	IP
In-Reply-To: <CAHgk4BVjSodyDpHYnuX_tE_DM=NbfaxzOnLruk=joNwhsq0XyQ@mail.gmail.com>
References: <CAHgk4BVjSodyDpHYnuX_tE_DM=NbfaxzOnLruk=joNwhsq0XyQ@mail.gmail.com>
Message-ID: <008c01ce9479$a0325590$e09700b0$@v10networks.ca>

Use STUN. Ext-rtp-ip and ext-sip-ip addresses are a great way to start
looking into this.

 

You can define stun servers as stun:stunserver.here.tld

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
Foster
Sent: Thursday, August 8, 2013 1:38 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] OpenWRT Router changes public IP to it's private
IP

 

We've got an openwrt router at a site, it's private IP is 10.0.0.1. Whenever
we get an inbound call on the external profile (and probably outbound calls
to, haven't checked yet), it changes the contact IP to the router's private
IP, so that every call we get looks like it's coming from thr router. We've
had some intermittent audio problems recently and we're trying to narrow
things down a bit. Is this the expected behavior for NAT? 

Sorry might seem like a dumb question, but my job entails much more than
keeping the company's server's happy and I'm starting to lose my grip and
I'm definitely not experienced with NAT.

All of my routers in the past have given FS the public IP of the contacting
server. I have web servers behind the same NAT and doing the same thing,
showing the private IP of the router instead of showing the public IP of the
client. So I know it's not an issue with FS. Just wanted to see if anyone
has exoerienced this before. Makes it difficult to use fail2ban on my
servers, as it continually jails my router.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.

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From moises.silva at gmail.com  Fri Aug  9 00:55:45 2013
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 8 Aug 2013 15:55:45 -0500
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <5203B1E9.6020300@coppice.org>
References: <5202C90D.3020308@quentustech.com>
	<CAMjaK3C=eYeNZsYDvjHg57zobNPOGiAVhkVSRyLHNg6mACO9gg@mail.gmail.com>
	<520317D1.3000409@gmail.com>
	<CAB+Kzh6477KJa4M9PxvrWWh=A4rcJnD7NttvqLcJKBUXctQneQ@mail.gmail.com>
	<5203B1E9.6020300@coppice.org>
Message-ID: <CAA4nhyAOrTuKTb5bQLw+By4=Ji9pM-wDpi7L70dThiyOujMCTQ@mail.gmail.com>

On Thu, Aug 8, 2013 at 9:57 AM, Steve Underwood <steveu at coppice.org> wrote:

> SMPP is strictly for SMS, and you need a direct connect to the network's
> SMSC to use it.
>
> This sudden interest in SMPP is puzzling. Several times over the life of
> FreeSwitch I've offered to resurrect a SMPP platform that was heavily
> used about 12 to 15 years ago, and make it open source. That platform
> has SGIP and a couple of other protocols, too. I've never had enough
> positive response to make me go ahead with the work.
>
>
Nice, if you have it why not just put the code on github (or elsewhere) and
let the community take care of integrating it/fixing it to work with
FreeSWITCH? (Assuming you don't have the time/interest on doing it).

Nothing peaks the interest as a little of code browsing :)

*Moises Silva
**Manager, Software Engineering***

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



**<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>

Products<http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Solutions<http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Events<http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Contact<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Wiki<http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Facebook<http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
 | Twitter<http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`|
| YouTube<http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
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From michel.brabants at gmail.com  Fri Aug  9 00:57:37 2013
From: michel.brabants at gmail.com (Michel Brabants)
Date: Thu, 8 Aug 2013 22:57:37 +0200
Subject: [Freeswitch-users] OpenWRT Router changes public IP to it's
 private IP
In-Reply-To: <CAHgk4BVjSodyDpHYnuX_tE_DM=NbfaxzOnLruk=joNwhsq0XyQ@mail.gmail.com>
References: <CAHgk4BVjSodyDpHYnuX_tE_DM=NbfaxzOnLruk=joNwhsq0XyQ@mail.gmail.com>
Message-ID: <CAHt+dNHt3paDPceiJT1e695cCKAZqxV5QWdPXzX+dQHUCdu-8w@mail.gmail.com>

Hey,
Are you running the SIP nat module on openers? There are 2 SIP modules and
One of the 2 is responsible for changing SIP packets. You can disable it.
The other One is responsible for opening ports. I don't use both as I fond
it top risky after some tests. However, it may depend on your environment.

Michel
Op 8 aug. 2013 22:42 schreef "Brian Foster" <bdfoster at davri.com> het
volgende:

> We've got an openwrt router at a site, it's private IP is 10.0.0.1.
> Whenever we get an inbound call on the external profile (and probably
> outbound calls to, haven't checked yet), it changes the contact IP to the
> router's private IP, so that every call we get looks like it's coming from
> thr router. We've had some intermittent audio problems recently and we're
> trying to narrow things down a bit. Is this the expected behavior for NAT?
>
> Sorry might seem like a dumb question, but my job entails much more than
> keeping the company's server's happy and I'm starting to lose my grip and
> I'm definitely not experienced with NAT.
>
> All of my routers in the past have given FS the public IP of the
> contacting server. I have web servers behind the same NAT and doing the
> same thing, showing the private IP of the router instead of showing the
> public IP of the client. So I know it's not an issue with FS. Just wanted
> to see if anyone has exoerienced this before. Makes it difficult to use
> fail2ban on my servers, as it continually jails my router.
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana
>
> Sent from a mobile device.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Fri Aug  9 01:45:46 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 8 Aug 2013 16:45:46 -0500
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <5203B1E9.6020300@coppice.org>
References: <5202C90D.3020308@quentustech.com>
	<CAMjaK3C=eYeNZsYDvjHg57zobNPOGiAVhkVSRyLHNg6mACO9gg@mail.gmail.com>
	<520317D1.3000409@gmail.com>
	<CAB+Kzh6477KJa4M9PxvrWWh=A4rcJnD7NttvqLcJKBUXctQneQ@mail.gmail.com>
	<5203B1E9.6020300@coppice.org>
Message-ID: <CAKbxfG_HFR8GMFc0SjTQHOerFfX8jzcjxXrOb6oikyLCpKa2tg@mail.gmail.com>

On Thu, Aug 8, 2013 at 9:57 AM, Steve Underwood <steveu at coppice.org> wrote:

> SMPP is strictly for SMS, and you need a direct connect to the network's
> SMSC to use it.
>
> This sudden interest in SMPP is puzzling. Several times over the life of
> FreeSwitch I've offered to resurrect a SMPP platform that was heavily
> used about 12 to 15 years ago, and make it open source. That platform
> has SGIP and a couple of other protocols, too. I've never had enough
> positive response to make me go ahead with the work.
>
> Regards,
> Steve
>


In my talk at ClueCon someone asked me if we have any plans for it in FS.
I said maybe if enough people showed an interest in it.  So I suspect this
thread is spawned from that comment =D
I'd say you ought to begin the resurrection if you are still willing =D

-- 
Anthony Minessale II

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 bdfoster at davri.com  Fri Aug  9 01:48:54 2013
From: bdfoster at davri.com (Brian Foster)
Date: Thu, 8 Aug 2013 17:48:54 -0400
Subject: [Freeswitch-users] OpenWRT Router changes public IP to it's
 private IP
In-Reply-To: <008c01ce9479$a0325590$e09700b0$@v10networks.ca>
References: <CAHgk4BVjSodyDpHYnuX_tE_DM=NbfaxzOnLruk=joNwhsq0XyQ@mail.gmail.com>
	<008c01ce9479$a0325590$e09700b0$@v10networks.ca>
Message-ID: <CAHgk4BXLaUF3X0M5rpCxJ8q=TMd8g1AS_bGaZgwYqpuSy1o1KQ@mail.gmail.com>

Stun is used on external profile, but I'll double check to see if they are
correct.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 8, 2013 5:08 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:

> Use STUN. Ext-rtp-ip and ext-sip-ip addresses are a great way to start
> looking into this.****
>
> ** **
>
> You can define stun servers as stun:stunserver.here.tld****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian Foster
> *Sent:* Thursday, August 8, 2013 1:38 PM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] OpenWRT Router changes public IP to it's
> private IP****
>
> ** **
>
> We've got an openwrt router at a site, it's private IP is 10.0.0.1.
> Whenever we get an inbound call on the external profile (and probably
> outbound calls to, haven't checked yet), it changes the contact IP to the
> router's private IP, so that every call we get looks like it's coming from
> thr router. We've had some intermittent audio problems recently and we're
> trying to narrow things down a bit. Is this the expected behavior for NAT?
> ****
>
> Sorry might seem like a dumb question, but my job entails much more than
> keeping the company's server's happy and I'm starting to lose my grip and
> I'm definitely not experienced with NAT.****
>
> All of my routers in the past have given FS the public IP of the
> contacting server. I have web servers behind the same NAT and doing the
> same thing, showing the private IP of the router instead of showing the
> public IP of the client. So I know it's not an issue with FS. Just wanted
> to see if anyone has exoerienced this before. Makes it difficult to use
> fail2ban on my servers, as it continually jails my router.****
>
> Thank you,****
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana****
>
> Sent from a mobile device.****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at davri.com  Fri Aug  9 01:51:49 2013
From: bdfoster at davri.com (Brian Foster)
Date: Thu, 8 Aug 2013 17:51:49 -0400
Subject: [Freeswitch-users] OpenWRT Router changes public IP to it's
 private IP
In-Reply-To: <CAHt+dNHt3paDPceiJT1e695cCKAZqxV5QWdPXzX+dQHUCdu-8w@mail.gmail.com>
References: <CAHgk4BVjSodyDpHYnuX_tE_DM=NbfaxzOnLruk=joNwhsq0XyQ@mail.gmail.com>
	<CAHt+dNHt3paDPceiJT1e695cCKAZqxV5QWdPXzX+dQHUCdu-8w@mail.gmail.com>
Message-ID: <CAHgk4BVEDfF74HriR2VNepU5fg-9PTXTYSechnXLndO6QMwy8g@mail.gmail.com>

I doubt either one are used, atleast I haven't installed it. It's running
Backfire and the router itself is a Netgear WNDR3700. Unless it's installed
by default...

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 8, 2013 5:08 PM, "Michel Brabants" <michel.brabants at gmail.com> wrote:

> Hey,
> Are you running the SIP nat module on openers? There are 2 SIP modules and
> One of the 2 is responsible for changing SIP packets. You can disable it.
> The other One is responsible for opening ports. I don't use both as I fond
> it top risky after some tests. However, it may depend on your environment.
>
> Michel
> Op 8 aug. 2013 22:42 schreef "Brian Foster" <bdfoster at davri.com> het
> volgende:
>
>> We've got an openwrt router at a site, it's private IP is 10.0.0.1.
>> Whenever we get an inbound call on the external profile (and probably
>> outbound calls to, haven't checked yet), it changes the contact IP to the
>> router's private IP, so that every call we get looks like it's coming from
>> thr router. We've had some intermittent audio problems recently and we're
>> trying to narrow things down a bit. Is this the expected behavior for NAT?
>>
>> Sorry might seem like a dumb question, but my job entails much more than
>> keeping the company's server's happy and I'm starting to lose my grip and
>> I'm definitely not experienced with NAT.
>>
>> All of my routers in the past have given FS the public IP of the
>> contacting server. I have web servers behind the same NAT and doing the
>> same thing, showing the private IP of the router instead of showing the
>> public IP of the client. So I know it's not an issue with FS. Just wanted
>> to see if anyone has exoerienced this before. Makes it difficult to use
>> fail2ban on my servers, as it continually jails my router.
>>
>> Thank you,
>>
>> Brian Foster
>> Project Manager/Owner's Rep.
>> Davri Investments, Inc.
>> O: 317-787-2686 x2102
>> M: 317-600-9753
>> E: bdfoster at davri.com
>> Indianapolis, Indiana
>>
>> Sent from a mobile device.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jleung at v10networks.ca  Fri Aug  9 01:55:54 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Thu, 8 Aug 2013 14:55:54 -0700
Subject: [Freeswitch-users] OpenWRT Router changes public IP to it's
	private IP
In-Reply-To: <CAHgk4BXLaUF3X0M5rpCxJ8q=TMd8g1AS_bGaZgwYqpuSy1o1KQ@mail.gmail.com>
References: <CAHgk4BVjSodyDpHYnuX_tE_DM=NbfaxzOnLruk=joNwhsq0XyQ@mail.gmail.com>	<008c01ce9479$a0325590$e09700b0$@v10networks.ca>
	<CAHgk4BXLaUF3X0M5rpCxJ8q=TMd8g1AS_bGaZgwYqpuSy1o1KQ@mail.gmail.com>
Message-ID: <00af01ce9482$0f36ec00$2da4c400$@v10networks.ca>

You also may want to see if OpenWRT is doing something called symmetric NAT
too. That can cause STUN's port detection technique to fail completely.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
Foster
Sent: Thursday, August 8, 2013 2:49 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] OpenWRT Router changes public IP to it's
private IP

 

Stun is used on external profile, but I'll double check to see if they are
correct.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.

On Aug 8, 2013 5:08 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:

Use STUN. Ext-rtp-ip and ext-sip-ip addresses are a great way to start
looking into this.

 

You can define stun servers as stun:stunserver.here.tld

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian
Foster
Sent: Thursday, August 8, 2013 1:38 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] OpenWRT Router changes public IP to it's private
IP

 

We've got an openwrt router at a site, it's private IP is 10.0.0.1. Whenever
we get an inbound call on the external profile (and probably outbound calls
to, haven't checked yet), it changes the contact IP to the router's private
IP, so that every call we get looks like it's coming from thr router. We've
had some intermittent audio problems recently and we're trying to narrow
things down a bit. Is this the expected behavior for NAT? 

Sorry might seem like a dumb question, but my job entails much more than
keeping the company's server's happy and I'm starting to lose my grip and
I'm definitely not experienced with NAT.

All of my routers in the past have given FS the public IP of the contacting
server. I have web servers behind the same NAT and doing the same thing,
showing the private IP of the router instead of showing the public IP of the
client. So I know it's not an issue with FS. Just wanted to see if anyone
has exoerienced this before. Makes it difficult to use fail2ban on my
servers, as it continually jails my router.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102 <tel:317-787-2686%20x2102> 
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 bdfoster at davri.com  Fri Aug  9 02:11:47 2013
From: bdfoster at davri.com (Brian Foster)
Date: Thu, 8 Aug 2013 18:11:47 -0400
Subject: [Freeswitch-users] OpenWRT Router changes public IP to it's
 private IP
In-Reply-To: <00af01ce9482$0f36ec00$2da4c400$@v10networks.ca>
References: <CAHgk4BVjSodyDpHYnuX_tE_DM=NbfaxzOnLruk=joNwhsq0XyQ@mail.gmail.com>
	<008c01ce9479$a0325590$e09700b0$@v10networks.ca>
	<CAHgk4BXLaUF3X0M5rpCxJ8q=TMd8g1AS_bGaZgwYqpuSy1o1KQ@mail.gmail.com>
	<00af01ce9482$0f36ec00$2da4c400$@v10networks.ca>
Message-ID: <CAHgk4BU=HcoesBu4qQKYAAygtTyi-ggL-26w6XKymzJfKmiMAw@mail.gmail.com>

Like DNAT vs SNAT? Forwarding rules are set to DNAT in luci if that is what
you are referring to.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 8, 2013 6:00 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:

> You also may want to see if OpenWRT is doing something called symmetric
> NAT too. That can cause STUN?s port detection technique to fail completely.
> ****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian Foster
> *Sent:* Thursday, August 8, 2013 2:49 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] OpenWRT Router changes public IP to
> it's private IP****
>
> ** **
>
> Stun is used on external profile, but I'll double check to see if they are
> correct.****
>
> Thank you,****
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana****
>
> Sent from a mobile device.****
>
> On Aug 8, 2013 5:08 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:****
>
> Use STUN. Ext-rtp-ip and ext-sip-ip addresses are a great way to start
> looking into this.****
>
>  ****
>
> You can define stun servers as stun:stunserver.here.tld****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian Foster
> *Sent:* Thursday, August 8, 2013 1:38 PM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] OpenWRT Router changes public IP to it's
> private IP****
>
>  ****
>
> We've got an openwrt router at a site, it's private IP is 10.0.0.1.
> Whenever we get an inbound call on the external profile (and probably
> outbound calls to, haven't checked yet), it changes the contact IP to the
> router's private IP, so that every call we get looks like it's coming from
> thr router. We've had some intermittent audio problems recently and we're
> trying to narrow things down a bit. Is this the expected behavior for NAT?
> ****
>
> Sorry might seem like a dumb question, but my job entails much more than
> keeping the company's server's happy and I'm starting to lose my grip and
> I'm definitely not experienced with NAT.****
>
> All of my routers in the past have given FS the public IP of the
> contacting server. I have web servers behind the same NAT and doing the
> same thing, showing the private IP of the router instead of showing the
> public IP of the client. So I know it's not an issue with FS. Just wanted
> to see if anyone has exoerienced this before. Makes it difficult to use
> fail2ban on my servers, as it continually jails my router.****
>
> Thank you,****
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana****
>
> Sent from a mobile device.****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Fri Aug  9 05:03:04 2013
From: eidevm5 at gmail.com (Peter)
Date: Fri, 9 Aug 2013 11:03:04 +1000
Subject: [Freeswitch-users] One way audio to CME
In-Reply-To: <072F50C1-88FC-47DB-A281-61CF3D90FA05@gmail.com>
References: <CABkS=xp=mczqaJBmK8E_DOt0jWpvXu2x01V_QdU2_e0NOXWFig@mail.gmail.com>
	<89239AD2-AE02-4E1C-8692-6E77894D8384@gmail.com>
	<CABkS=xo7NQoTNpeR0dMrLkeBndypfPmTbPEEbUonMu_NJ7wfAA@mail.gmail.com>
	<9C8EC1A4-7DAC-4226-BDD6-F63A9AFD107F@gmail.com>
	<CABkS=xpxRiZ_+SGsjO-CASNSnV66zUxhn_-GmaU+S0_HRS=A1Q@mail.gmail.com>
	<1393ACC3-56D8-49B4-86E7-2F5C80435799@gmail.com>
	<CABkS=xrnJFayPeh8du-nScJKKROmHmSOnFiwV_Vug5t9tvh-Vw@mail.gmail.com>
	<CAHgk4BXmfJ1EDRQiV6zd4+k_Zt10jT476ZAP+i1JX+vwNR_bdg@mail.gmail.com>
	<CABkS=xpyeMCX67Au0j=e8R_n22bvoJDYn6XRf5N8M0VWD+SuDw@mail.gmail.com>
	<072F50C1-88FC-47DB-A281-61CF3D90FA05@gmail.com>
Message-ID: <CABkS=xqNggjjqA8aa1kG_pqm3+iyYcyiSudgwTaFmhGXEk7yEA@mail.gmail.com>

Ah, it all makes sense now.  I was misunderstanding the way the way the
sip/rtp IP settings worked.

Thank you very much for sorting this out.  Much appreciated.


On Thu, Aug 8, 2013 at 7:07 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:

> sip-ip & rtp-ip should be set per profile. So if you have 5 profiles each
> profile.xml will have these params set to whatever IP address the profile
> should use.
> In your case
>
> external profile
> <param name="sip-ip" value="10.1.1.206"/>
> <param name="rtp-ip" value="10.1.1.206"/>
>
> internal profile
> <param name="sip-ip" value="10.10.10.206"/>
> <param name="rtp-ip" value="10.10.10.206"/>
>
> you only need to worry about ext-sip-ip and ext-rtp-ip if there is NAT in
> the path.
> Better to leave them in each profile to
> <param name="ext-sip-ip" value="auto-nat"/>
> <param name="ext-rtp-ip" value="auto-nat"/>
>
>
> On 8 Aug 2013, at 03:29, Peter <eidevm5 at gmail.com> wrote:
>
> Hi Brian.
>
> I did have the IP set in the external profile, but after removing them it
> didn't seem to make any difference, ie:  the SIP ports were listening on
> the correct IP address.
>
> Have to say I'm getting a little confused as to when and where to use
> sip-ip and ext-sip-ip
>
> Should sip-ip ALWAYS be set to your internal IP address and ext-sip-ip to
> your external IP address?
>
> If so, which profiles do they need to be set in?   If they are set in both
> the internal and external SIP profiles, does that matter?
>
> Thanks
>
> Peter
>
>
>
> On Thu, Aug 8, 2013 at 11:41 AM, Brian Foster <bdfoster at davri.com> wrote:
>
>> Your ip for the external profile will show whatever the ip is set to in
>> the external profile configs. It doesn't matter what is set in the internal
>> profile.
>>
>> Thank you,
>>
>> Brian Foster
>> Project Manager/Owner's Rep.
>> Davri Investments, Inc.
>> O: 317-787-2686 x2102
>> M: 317-600-9753
>> E: bdfoster at davri.com
>> Indianapolis, Indiana
>>
>> Sent from a mobile device.
>> On Aug 7, 2013 9:36 PM, "Peter" <eidevm5 at gmail.com> wrote:
>>
>>> Hi Anthony
>>>
>>> Really appreciate the taking your time to look at this.  It's starting
>>> to drive me nuts.
>>>
>>> My dialplan for calls to CME is:
>>>
>>>   <extension name="incoming-cme">
>>>         <condition field="destination_number" expression="^(3\d\d\d)$">
>>>             <action application="bridge" data="sofia/internal/${
>>> destination_number}@10.10.10.203"/>
>>>         </condition>
>>>   </extension>
>>>
>>>
>>> My internal sip profile has:
>>>
>>>     <param name="rtp-ip" value="10.10.10.206"/>
>>>     <param name="sip-ip" value="10.10.10.206"/>
>>>     <param name="ext-rtp-ip" value="10.1.1.206"/>
>>>     <param name="ext-sip-ip" value="10.1.1.206"/>
>>>
>>> The output from:
>>>
>>> sofia status
>>>
>>> is:
>>>
>>> internal    profile                 sip:mod_sofia at 10.1.1.206:5060
>>> RUNNING (0)
>>> external    profile                 sip:mod_sofia at 10.1.1.206:5060
>>> RUNNING (0)
>>>
>>> Should internal show 10.10.10.206??
>>>
>>> The output from:
>>>
>>> sofia status profile internal
>>>
>>> shows:
>>>
>>> Name                 internal
>>> Domain Name          N/A
>>> Auto-NAT             false
>>> DBName               sofia_reg_internal
>>> Pres Hosts           10.10.10.206,10.1.1.206
>>> Dialplan             XML
>>> Context              public
>>> Challenge Realm      auto_from
>>> RTP-IP               10.10.10.206
>>> Ext-RTP-IP           10.1.1.206
>>> SIP-IP               10.10.10.206
>>> Ext-SIP-IP           10.1.1.206
>>> URL                  sip:mod_sofia at 10.1.1.206:5060
>>> BIND-URL             sip:mod_sofia at 10.1.1.206:5060;maddr=10.10.10.206
>>> HOLD-MUSIC           N/A
>>> OUTBOUND-PROXY       N/A
>>> CODECS IN            iLBC at 30i,PCMU,PCMA,GSM
>>> CODECS OUT           iLBC at 30i,PCMU,PCMA,GSM
>>> TEL-EVENT            101
>>> DTMF-MODE            rfc2833
>>> CNG                  13
>>> SESSION-TO           0
>>> MAX-DIALOG           0
>>> NOMEDIA              false
>>> LATE-NEG             false
>>> PROXY-MEDIA          false
>>> ZRTP-PASSTHRU        false
>>> AGGRESSIVENAT        false
>>> STUN-ENABLED         true
>>> STUN-AUTO-DISABLE    false
>>>
>>>
>>> The SIP trace from the Freeswitch SBC is at:
>>>
>>> http://pastebin.freeswitch.org/21279
>>>
>>> I've been playing around with all sorts of different combinations of
>>> SIP/RTP IP settings, but still no closer.
>>>
>>> Hope you have some insight.
>>>
>>> Thanks
>>>
>>> Peter
>>>
>>> On Wed, Aug 7, 2013 at 6:31 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>>>
>>>> Hi Peter,
>>>>
>>>> Your debug shows the invite with via/from/contact/rpid all coming from
>>>> 10.1.1.206, your external side.
>>>> Check your bridge statement, is it using the correct sip profile?
>>>> Check your sip profile SBC internal params rtp-ip & sip-ip, make sure
>>>> they are set correctly to 10.10.10.206
>>>> Paste up your logs from the sbc including sip trace.
>>>>
>>>> Anthony
>>>>
>>>>
>>>> On 7 Aug 2013, at 08:12, Peter <eidevm5 at gmail.com> wrote:
>>>>
>>>> Hi Anthony.
>>>>
>>>> Yes, the SIP profiles are the same for calls going to Kamailio and to
>>>> CME/CUBE.
>>>>
>>>> Note that CME only has one interface, so binding the source interface
>>>> doesn't really make much sense.
>>>>
>>>> Note that I've simplified my set up a little and the phone that was
>>>> registered to CUCM is now registered to CME.  However, the result is still
>>>> the same, ie:  one way audio to the Cisco phone.
>>>>
>>>> You can see the SIP debug from CME at:
>>>>
>>>> http://pastebin.freeswitch.org/21274
>>>>
>>>> The call is coming from 1001 at 10.1.1.204 to 3000 at 10.10.10.203
>>>>
>>>> where
>>>>
>>>> 10.1.1.204  - Freeswitch where SIP clients register to
>>>> 10.1.1.206  - External side of Freeswitch SBC
>>>> 10.10.10.206 - Internal side of Freeswitch SBC
>>>> 10.10.10.203 - CME
>>>>
>>>> Peter
>>>>
>>>>
>>>>
>>>> On Tue, Aug 6, 2013 at 5:13 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>>>>
>>>>> Hi Peter,
>>>>>
>>>>> Because the calls are fine when using Kamailio I'm assuming your sip
>>>>> profiles are fine and you FS SBC config is fine. Are you using the same
>>>>> profiles?
>>>>> Yes you are correct. Have you added the commands? Add them as a first
>>>>> step.
>>>>> Send on a 'debug ccsip messages'
>>>>>
>>>>> Anthony
>>>>>
>>>>>
>>>>>
>>>>> On 6 Aug 2013, at 05:35, Peter <eidevm5 at gmail.com> wrote:
>>>>>
>>>>> Thanks for replying Anthony.
>>>>>
>>>>> Keep in mind that I have very little experience with Cisco products,
>>>>> so I may be missing something fundamental.
>>>>>
>>>>> As far as I can see
>>>>>
>>>>> voice-class sip bind media source-interface ....
>>>>>
>>>>> is just used to bind the SIP or media stream to the appropriate
>>>>> interface on the CUBE.
>>>>>
>>>>> My issue is that the CUBE is trying to initiate the return RTP stream
>>>>> to the external interface (instead of the internal interface) on the
>>>>> Freeswitch SBC.
>>>>>
>>>>> Is my understanding of the sip bind media command correct?
>>>>>
>>>>> Thanks
>>>>>
>>>>> Peter
>>>>>
>>>>>
>>>>> On Mon, Aug 5, 2013 at 5:23 PM, Anthony McGarry <agtmcgarry at gmail.com>wrote:
>>>>>
>>>>>> On cube make sure you specify the source address on your dial-peers
>>>>>> voice-class sip bind media|control
>>>>>> to the correct side. I have seen one way audio when not set.
>>>>>>
>>>>>> On 5 Aug 2013, at 06:29, Peter <eidevm5 at gmail.com> wrote:
>>>>>>
>>>>>> >
>>>>>> >
>>>>>> > I currently have successful two way calls (signalling and media) in
>>>>>> the following setup
>>>>>> >
>>>>>> >
>>>>>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router
>>>>>>  -> Kamailio --> Internal Linphone
>>>>>> >
>>>>>> > However, when I try to call a Cisco handset that is registered to
>>>>>> CUCM9 via CME in the following config:
>>>>>> >
>>>>>> > External Linphone -->  Freeswitch  --> Freeswitch SBC  -> Router
>>>>>>  -> CME ->  CUCM9 -->  Cisco handset
>>>>>> >
>>>>>> > The call signalling appears to be working fine and I can
>>>>>> successfully  initiate a call from each end, but the only RTP stream that
>>>>>> is working is from the external Linphone client to the Cisco handset.
>>>>>> >
>>>>>> > Note that CME is being used as a CUBE device, so all SIP and RTP
>>>>>> goes via it.
>>>>>> >
>>>>>> > Looking at the RTP debugs on CME I can see the problem is that the
>>>>>> "Media Dest Addr" is getting set to the external side of the FS SBC rather
>>>>>> than the internal IP address.
>>>>>> >
>>>>>> >
>>>>>> > I tried setting adding:
>>>>>> >
>>>>>> >             <action application="set"
>>>>>> data="disable_rtp_auto_adjust="true" />
>>>>>> >
>>>>>> > to the dialplan on the SBC, but it made no difference.
>>>>>> >
>>>>>> >
>>>>>> > Any suggestions as to what to check next?
>>>>>> >
>>>>>> > Thanks
>>>>>> >
>>>>>> > Peter
>>>>>> >
>>>>>> >
>>>>>> _________________________________________________________________________
>>>>>>
>>>>>>
>>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 rajat.toshniwal at tekmindz.com  Fri Aug  9 10:25:33 2013
From: rajat.toshniwal at tekmindz.com (Rajat toshniwal)
Date: Fri, 09 Aug 2013 11:55:33 +0530
Subject: [Freeswitch-users] Freeswitch with Digium T316 timed out,
 T316 timed out, 
In-Reply-To: <52039DD4.9070109@tekmindz.com>
References: <5203774A.7040906@tekmindz.com> <52039DD4.9070109@tekmindz.com>
Message-ID: <52048B5D.8000207@tekmindz.com>

Any Clues ??

On Thursday 08 August 2013 07:02 PM, Rajat toshniwal wrote:
> Forget to mention that I have already tested below mentioned 
> configuration
>
>    1. Go to FS source directory and change dir to this
>       path libs/freetdm/src/ftmod/ftmod_libpri
>    2. change #define T316_TIMEOUT_MS_DEFAULT to 10000
>    3. #define T316_TIMEOUT_MS_MAX to 300000
>    4. #define T316_ATTEMPT_LIMIT_DEFAULT to 30 (max. channels in PRI
>       lines)
>
> It didn't work
>
> Also we are using the latest version of Freeswitch 1.2.12 and libpri1.4.14
>
> Regards
> Rajat Toshniwal
>
>
>
> On Thursday 08 August 2013 04:17 PM, Rajat toshniwal wrote:
>> Hi
>>
>> I am trying to deploy freeswitch with Digium TE121 card for my office
>> setup, but it is continuously showing Signaling is up and channels are
>> down except D channel.
>> Our Architecture is like
>> We have freeswitch installed with libpri1.4 and Dahdi.
>> I am from India and here we are having E1 trunk.
>>
>> Dahdi Configuration is
>>
>> cat system.conf
>> # Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug  7 19:39:07 2013
>> # If you edit this file and execute /usr/sbin/dahdi_genconf again,
>> # your manual changes will be LOST.
>> # Dahdi Configuration File
>> #
>> # This file is parsed by the Dahdi Configurator, dahdi_cfg
>> #
>> # Span 1: WCT1/0 "Wildcard TE121 Card 0" (MASTER)
>> span=1,1,0,ccs,hdb3,crc4
>> # termtype: te
>> bchan=1-15,17-31
>> dchan=16
>> echocanceller=mg2,1-15,17-31
>>
>> # Global data
>>
>> loadzone    = uk
>> defaultzone    = uk
>>
>>
>>
>> cat modules
>> # Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules)
>> on Wed Aug  7 19:37:48 2013
>> # If you edit this file and execute /usr/sbin/dahdi_genconf again,
>> # your manual changes will be LOST.
>> wcte12xp
>> # Xorcom Astribank Devices
>> xpp_usb
>>
>>
>> dahdi_hardware
>> pci:0000:02:08.0     wcte12xp+    d161:8000 Wildcard TE121
>>
>> dahdi_scan
>> [1]
>> active=yes
>> alarms=OK
>> description=Wildcard TE121 Card 0
>> name=WCT1/0
>> manufacturer=Digium
>> devicetype=Wildcard TE121 (VPMOCT032)
>> location=PCI Bus 02 Slot 09
>> basechan=1
>> totchans=31
>> irq=0
>> type=digital-E1
>> syncsrc=1
>> lbo=0 db (CSU)/0-133 feet (DSX-1)
>> coding_opts=AMI,HDB3
>> framing_opts=CCS,CRC4
>> coding=HDB3
>> framing=CCS/CRC4
>>
>>
>> Card is properly installed and recognized by Dahdi
>>
>> Freetdm is compiled with libpri and configuration is like
>> cat /usr/local/freeswitch/conf/freetdm.conf
>> [general]
>> cpu_monitor =>  yes
>> cpu_monitoring_interval =>  2000 ; monitor usage every 2 seconds
>> cpu_set_alarm_threshold =>  90 ; whenever 90% of global CPU usage is
>> reached, trigger the alarm.
>> cpu_reset_alarm_threshold =>  80 ; when the CPU usage decreases at 80%,
>> clear the alarm.
>> cpu_alarm_action =>  reject,warn ; Start rejecting calls when the CPU
>> alarm is triggered and also print warnings.
>>
>> [span zt myDAHDISpan]
>> trunk_type =>  E1
>> group =>  g1
>> b-channel =>  1-15
>> d-channel =>  16
>> b-channel =>  17-31
>>
>>
>> cat /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml
>> <configuration name="freetdm.conf" description="FreeTDM Configuration">
>> <settings>
>> <param name="debug" value="1"/>
>> </settings>
>> <libpri_spans>
>> <span name="myDAHDISpan">
>> <!-- Log Levels: none, alert, crit, err, warning, notice, info, debug -->
>> <param name="switch" value="euroisdn"/>
>> <param name="node" value="cpe"/>
>> <param name="dialect" value="q931"/>
>> <param name="debug" value="all"/>
>> <param name="dialplan" value="XML"/>
>> <param name="context" value="public"/>
>> <param name="l1" value="alaw"/>
>> </span>
>> </libpri_spans>
>> </configuration>
>>
>>
>> Freeswitch logs are showing
>>
>> 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c10][1:10]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c11][1:11]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c12][1:12]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c13][1:13]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c14][1:14]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c15][1:15]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c17][1:17]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c18][1:18]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c19][1:19]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c20][1:20]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c21][1:21]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c22][1:22]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c23][1:23]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c24][1:24]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c25][1:25]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c26][1:26]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c27][1:27]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c28][1:28]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c29][1:29]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c30][1:30]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c31][1:31]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:23.673847 [WARNING] ftdm_io.c:3022 [s1c5][1:5] Channel
>> not opened, proceeding anyway
>> 2013-08-08 15:38:48.613848 [WARNING] ftmod_libpri.c:1975 [s1c6][1:6] --
>> T316 timed out, resending RESTART request
>> 2013-08-08 15:38:48.613848 [WARNING] ftmod_libpri.c:1975 [s1c7][1:7] --
>> T316 timed out, resending RESTART request
>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c8][1:8] --
>> T316 timed out, resending RESTART request
>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c9][1:9] --
>> T316 timed out, resending RESTART request
>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c10][1:10]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c11][1:11]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c12][1:12]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c13][1:13]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c14][1:14]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c15][1:15]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c17][1:17]
>> -- T316 timed out, resending RESTART request
>>
>>
>>
>> Libpri logs are showing
>>
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> DL-DATA request
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> Protocol Discriminator: Q.931 (8)  len=13
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> Message Type: RESTART (70)
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>> TEI=0 Transmitting N(S)=4, window is open V(A)=2 K=7
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> TEI: 0 State 7(Multi-frame established)
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> V(A)=2, V(S)=4, V(R)=6
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> T200_id=8192, N200=3, T203_id=0
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>  [
>> 00 01 08 0c 08 02 00 00 46 18 03 a9 83 8d 79 01 80 ]
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> Informational frame:
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> SAPI: 00  C/R: 0 EA: 0
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> TEI: 000        EA: 1
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> N(S): 004   0: 0
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> N(R): 006   P: 0
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>  13
>> bytes of data
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> Protocol Discriminator: Q.931 (8)  len=13
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> Message Type: RESTART (70)
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> [18 03 a9 83 8d]
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)  Spare: 0
>> Exclusive  Dchan: 0
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>   >                        ChanSel: As indicated in following octets
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>   >                        Ext: 1  Coding: 0  Number Specified  Channel
>> Type: 3
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>   >                        Ext: 1  Channel: 13 Type: CPE]
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> [79 01 80]
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> Restart Indicator (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated
>> Channel (0) ]
>> 2013-08-08 15:41:18.633844 [DEBUG] lpwrap_pri.c:199 [s1c16][1:16] --
>> Starting timer 0x2678478 with timeout 30000 ms
>> 2013-08-08 15:41:18.633844 [WARNING] ftmod_libpri.c:1975 [s1c14][1:14]
>> -- T316 timed out, resending RESTART request
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> DL-DATA request
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> Protocol Discriminator: Q.931 (8)  len=13
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> Message Type: RESTART (70)
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>> TEI=0 Transmitting N(S)=5, window is open V(A)=2 K=7
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> TEI: 0 State 7(Multi-frame established)
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> V(A)=2, V(S)=5, V(R)=6
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> T200_id=8192, N200=3, T203_id=0
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>  [
>> 00 01 0a 0c 08 02 00 00 46 18 03 a9 83 8e 79 01 80 ]
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> Informational frame:
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> SAPI: 00  C/R: 0 EA: 0
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> TEI: 000        EA: 1
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> N(S): 005   0: 0
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> N(R): 006   P: 0
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>  13
>> bytes of data
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> Protocol Discriminator: Q.931 (8)  len=13
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> Message Type: RESTART (70)
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> [18 03 a9 83 8e]
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)  Spare: 0
>> Exclusive  Dchan: 0
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>   >                        ChanSel: As indicated in following octets
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>   >                        Ext: 1  Coding: 0  Number Specified  Channel
>> Type: 3
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>   >                        Ext: 1  Channel: 14 Type: CPE]
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> [79 01 80]
>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>> Restart Indicator (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated
>> Channel (0) ]
>> 2013-08-08 15:41:18.633844 [DEBUG] lpwrap_pri.c:199 [s1c16][1:16] --
>> Starting timer 0x26784d8 with timeout 30000 ms
>> 2013-08-08 15:41:18.633844 [WARNING] ftmod_libpri.c:1975 [s1c15][1:15]
>> -- T316 timed out, resending RESTART request
>>
>>
>> ftdm dump 1 n (where n is from 1 to 15 and  17 to 31) is showing state
>> as down
>>
>> freeswitch at dst1>  ftdm dump 1 1
>>
>> span_id: 1
>> chan_id: n
>> physical_span_id: 1
>> physical_chan_id: n
>> physical_status: ok
>> physical_status_red: 0
>> physical_status_yellow: 0
>> physical_status_rai: 0
>> physical_status_blue: 0
>> physical_status_ais: 0
>> physical_status_general: 0
>> signaling_status: UP
>> type: B
>> state: DOWN
>> last_state: RESTART
>> txgain: 0.00
>> rxgain: 0.00
>> cid_date:
>> cid_name:
>> cid_num:
>> ani:
>> aniII:
>> dnis:
>> rdnis:
>> cause: NONE
>> session: (none)
>>
>> I have tried many configurations, but I am not able to fix that issue ,
>> If I put my Pri line into panasonic PBX, it is working properly.
>>
>> Kindly help me in solving this issue.
>>
>> Regards
>> Rajat Toshniwal
>> ----------------------------------------------------------------------------------
>> Disclaimer: The information contained in this communication is confidential, private, proprietary, or otherwise privileged and is intended only for the use of the addressee. Unauthorized use, disclosure, distribution or copying is strictly prohibited and may be unlawful. If you have received this communication in error, please delete this message and notify the sender immediately - Samin TekMindz India Pvt. Ltd.
>> ----------------------------------------------------------------------------------
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
>
> ------------------------------------------------------------------------
> Disclaimer: The information contained in this communication is 
> confidential, private, proprietary, or otherwise privileged and is 
> intended only for the use of the addressee.Unauthorized use, 
> disclosure, distribution or copying is strictly prohibited and may be 
> unlawful. If you have received this communication in error, please 
> delete this message and notify the sender immediately - Samin TekMindz 
> India Pvt.Ltd.
> ------------------------------------------------------------------------
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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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Disclaimer: The information contained in this communication is confidential, private, proprietary, or otherwise privileged and is intended only for the use of the addressee. Unauthorized use, disclosure, distribution or copying is strictly prohibited and may be unlawful. If you have received this communication in error, please delete this message and notify the sender immediately - Samin TekMindz India Pvt. Ltd.
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From william.king at quentustech.com  Fri Aug  9 11:00:33 2013
From: william.king at quentustech.com (William King)
Date: Fri, 09 Aug 2013 02:00:33 -0500
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <CAB+Kzh6477KJa4M9PxvrWWh=A4rcJnD7NttvqLcJKBUXctQneQ@mail.gmail.com>
References: <5202C90D.3020308@quentustech.com>
	<CAMjaK3C=eYeNZsYDvjHg57zobNPOGiAVhkVSRyLHNg6mACO9gg@mail.gmail.com>
	<520317D1.3000409@gmail.com>
	<CAB+Kzh6477KJa4M9PxvrWWh=A4rcJnD7NttvqLcJKBUXctQneQ@mail.gmail.com>
Message-ID: <52049391.4080105@quentustech.com>

>From what I've seen, retail level SMS carriers(not sms aggregators)
usually offer a mix of HTTP, XMPP, SIP SIMPLE, or SMPP interfaces. I've
seen the higher volume SMS connections prefer SMPP or HTTP.

I'm exploring to see if there is a business case for FreeSWITCH support
for SMPP(either direct support, or interface support through an
independent application) that is not already covered by the current FS
feature set.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 08/08/2013 04:23 AM, Raymond Chandler wrote:
> So, to the best of my knowledge,  SMPP is strictly for SMS so you can
> route SMS to your clients via FS to SIMPLE / dingaling / etc clients
> 
> -Ray
> 
> On Aug 7, 2013 11:05 PM, "Victor Chukalovskiy"
> <victor.chukalovskiy at gmail.com <mailto:victor.chukalovskiy at gmail.com>>
> wrote:
> 
>     So am I as well as another ClueCon attendee I spoke to.
> 
>     What is the usage scenario you are looking at?
> 
>     On 13-08-07 06:46 PM, Dmitry Lysenko wrote:
>>     Yes. I am interested.
>>
>>     Dmitry.
>>
>>
>>     2013/8/8 William King <william.king at quentustech.com
>>     <mailto:william.king at quentustech.com>>
>>
>>         Is anyone on this list interested in FreeSWITCH support for
>>         SMPP for SMS
>>         messages?
>>
>>         For more information about the SMPP protocol checkout:
>>         http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
>>
>>         If so feel free to contact me on or off this list.
>>         --
>>         William King
>>         Senior Engineer
>>         Quentus Technologies, INC
>>         1037 NE 65th St Suite 273
>>         Seattle, WA 98115
>>         Main:   (877) 211-9337 <tel:%28877%29%20211-9337>
>>         Office: (206) 388-4772 <tel:%28206%29%20388-4772>
>>         Cell:   (253) 686-5518 <tel:%28253%29%20686-5518>
>>         william.king at quentustech.com <mailto:william.king at quentustech.com>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.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
>>
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.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
> 
> 
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
> 
>     
>     
> 
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nandy1925 at gmail.com  Fri Aug  9 13:15:29 2013
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 9 Aug 2013 17:15:29 +0800
Subject: [Freeswitch-users] FreeSwitch and Yealink
In-Reply-To: <5203CF01.8050501@c3i.bg>
References: <20130807044439.9A5F8F4002@mail.mydcs.ca> <5201DF92.9010800@c3i.bg>
	<20130807060000.E564FF4002@mail.mydcs.ca> <5203CF01.8050501@c3i.bg>
Message-ID: <CAHKqMb7xjKY3W1XRnwQFKZrGGetT76c0J6fyA1hDrQeJGWNYeA@mail.gmail.com>

Try to set verbose_sdp=true in vars.xml. I just experienced a problem with
a China phone using Broadcom chipset.

/Nandy

On Fri, Aug 9, 2013 at 1:01 AM, Ivan <ivan at c3i.bg> wrote:

> You have to set the variable in the dialplan, before bridging. You can
> also set it directly in the bridge command.
>
> More details at:
>
> http://wiki.freeswitch.org/wiki/Codec_Negotiation
>
> But the only way to really know what is going on is to enable siptrace
> on your sofia profile and see what is offered before your endpoint
> hangups (if you post your debug, please post on freeswitch's pastebin).
>
> ivan
>
>
> On 08/07/2013 08:59 AM, Paul wrote:
> > Ivan thanks for your reply.
> >
> > I'm willing to try anything :)
> >
> > Where would I set those variables? In the dialplan out to the sip trunk?
> >
> > Thanks
> >
> > Paul
> >
> > On Tue, 6 Aug, 2013 at 10:48 PM, Ivan <ivan at c3i.bg> wrote:
> >> Hangups with the cause you describe can be caused by a failed codec
> >> negotiation. Are you sure it's not the case ? Does the remote phone
> >> send a sip "not acceptable here" ? If needed you can force the
> >> selection of the "outgoing" codec:
> >> http://wiki.freeswitch.org/wiki/Channel_Variables#absolute_codec_stringivan
> >> On 08/07/2013 07:44 AM, Paul wrote:
> >>
> >>     Hi guys, Has anyone had any issues using FreeSwitch with Yealink
> >>     phones? My phones connect to FS via openvpn tunnel. All incoming
> >>     calls work no problem, call comes through phones ring everyone can
> >>     hear each other no issues, having a very strange issue though on
> >>     the outgoing calls. As soon as the destination party picks up
> >>     (this external calls) the call hangs up. short FS LOG:
> >>     switch_ivr_bridge.c:475 Channel [sofia/internal/105 at 10.0.0.34] has
> >>     been answered sofia.c:6528 Hangup sofia/internal/105 at 10.0.0.34
> >>     [CS_EXECUTE] [ORIGINATOR_CANCEL] switch_ivr_bridge.c:721 Hangup
> >>     sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] So
> >>     FS thinks the phone sent a BYE packet (which I can see with
> >>     siptrace) but the phone's timer keeps going as if it thinks the
> >>     call is supposed to keep going. Internal extension to extension
> >>     works fine (even if the extensions are at a different physical
> >>     location and subnet). I setup a second account to one of my
> >>     asterisk servers and outgoing/incoming work just fine, so it seems
> >>     this strange combination of FS and Yealink ... does it on 2 models
> >>     T32G and T38G (only phones I have). I have updated firwmare to
> >>     their latest version (which in the comments say freeswitch ready)
> >>     Wondering if anyone else had any experience with these, or has
> >>     some thoughts? Thanks Paul
> >>
> _________________________________________________________________________
> >>     Professional FreeSWITCH Consulting Services:
> >>     consulting at freeswitch.org http://www.freeswitchsolutions.com
> >>     
> >>      Official FreeSWITCH Sites
> >>     http://www.freeswitch.org http://wiki.freeswitch.org
> >>     http://www.cluecon.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
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services: consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The
> >> CudaTel Communication Server  Official
> >> FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org
> >> http://www.cluecon.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
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sravi123 at yahoo.com  Fri Aug  9 16:36:07 2013
From: sravi123 at yahoo.com (Ravi)
Date: Fri, 9 Aug 2013 05:36:07 -0700 (PDT)
Subject: [Freeswitch-users] Wanpipe - Wanrouter help - freetdm and freeswitch
Message-ID: <1376051767.14626.YahooMailNeo@web160501.mail.bf1.yahoo.com>

Hello Everyone !

I am not sure if this error is relevant for this mailing list. My apologies if this is not the forum. I am trying to configure the Sangoma card, with the freetdm installation for Freeswitch. When I complete the wanpipe installation, and try wanrouter status, and wanrouter start I get the following:

[root at bfree-server log]# wanrouter start

Starting up device: wanpipe1


wanconfig: WAN device wanpipe1 driver load failed !!
: ioctl(wanpipe1,ROUTER_SETUP) failed:
:	22 - Invalid argument


Wanpipe driver did not load properly
Please check /var/log/wanrouter and
/var/log/messages for errors

Configuring interfaces: w1g1 w1g1: unknown interface: No such device

done.
___________________________________________________________________

Can anyone help me understand why this is happening ? And how to resolve it. 


Thanks.
Ravi
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From khorsmann at gmail.com  Fri Aug  9 20:14:05 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Fri, 9 Aug 2013 18:14:05 +0200
Subject: [Freeswitch-users] Wanpipe - Wanrouter help - freetdm and
	freeswitch
In-Reply-To: <1376051767.14626.YahooMailNeo@web160501.mail.bf1.yahoo.com>
References: <1376051767.14626.YahooMailNeo@web160501.mail.bf1.yahoo.com>
Message-ID: <CAFArqsb0KAMJsUCL71fxFBpSQ6SdCyuxDqmq1sRbBkiY+ksAjA@mail.gmail.com>

Hi Ravi,


best forum is read the fine sangoma wiki for proper driver installation
http://wiki.sangoma.com/ and if that didnt work, try the nice sangoma
support.

Sangoma + FreeSWITCH is a working combination. Whats unknown is <??? Linux
Distro> and Wanpipe Driver <??? Version). That should you collect before
contact sangoma techdesk.


2013/8/9 Ravi <sravi123 at yahoo.com>

> Hello Everyone !
>
> I am not sure if this error is relevant for this mailing list. My
> apologies if this is not the forum. I am trying to configure the Sangoma
> card, with the freetdm installation for Freeswitch. When I complete the
> wanpipe installation, and try wanrouter status, and wanrouter start I get
> the following:
>
> [root at bfree-server log]# wanrouter start
>
> Starting up device: wanpipe1
>
>
> wanconfig: WAN device wanpipe1 driver load failed !!
> : ioctl(wanpipe1,ROUTER_SETUP) failed:
> : 22 - Invalid argument
>
>
> Wanpipe driver did not load properly
> Please check /var/log/wanrouter and
> /var/log/messages for errors
>
> Configuring interfaces: w1g1 w1g1: unknown interface: No such device
>
> done.
> ___________________________________________________________________
>
> Can anyone help me understand why this is happening ? And how to resolve
> it.
>
> Thanks.
> Ravi
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From nickolayr at gmail.com  Fri Aug  9 21:59:26 2013
From: nickolayr at gmail.com (Nikolay Rogoshchenkov)
Date: Fri, 9 Aug 2013 13:59:26 -0400
Subject: [Freeswitch-users] FreeSwitch and Yealink
In-Reply-To: <20130807044439.9A5F8F4002@mail.mydcs.ca>
References: <20130807044439.9A5F8F4002@mail.mydcs.ca>
Message-ID: <CAAQCs31MW_pFNiD=p=udTp4YmDueu=9rY8uDhozrYTU5JGy-uw@mail.gmail.com>

Check RTP Packet Size too.

--
Rogoshchenkov Nikolay


On Wed, Aug 7, 2013 at 12:44 AM, Paul <pasha at prosperity4ever.com> wrote:

> Hi guys,
>
> Has anyone had any issues using FreeSwitch with Yealink phones? My phones
> connect to FS via openvpn tunnel. All incoming calls work no problem, call
> comes through phones ring everyone can hear each other no issues, having a
> very strange issue though on the outgoing calls. As soon as the destination
> party picks up (this external calls) the call hangs up.
>
> short FS LOG:
>
> switch_ivr_bridge.c:475 Channel [sofia/internal/105 at 10.0.0.34] has been
> answered
> sofia.c:6528 Hangup sofia/internal/105 at 10.0.0.34 [CS_EXECUTE]
> [ORIGINATOR_CANCEL]
> switch_ivr_bridge.c:721 Hangup sofia/external/2503004900[CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>
> So FS thinks the phone sent a BYE packet (which I can see with siptrace)
> but the phone's timer keeps going as if it thinks the call is supposed to
> keep going.
>
> Internal extension to extension works fine (even if the extensions are at
> a different physical location and subnet).
>
> I setup a second account to one of my asterisk servers and
> outgoing/incoming work just fine, so it seems this strange combination of
> FS and Yealink ... does it on 2 models T32G and T38G (only phones I have).
>
> I have updated firwmare to their latest version (which in the comments say
> freeswitch ready)
>
> Wondering if anyone else had any experience with these, or has some
> thoughts?
>
> Thanks
>
> Paul
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 karl at xtronics.com  Sat Aug 10 00:24:01 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Fri, 09 Aug 2013 15:24:01 -0500
Subject: [Freeswitch-users] freeDTM detect if line is in use
In-Reply-To: <CAKzWOxUdSGDoesNWXtHToMJ=rJZmT8i1D1T7b8mokQjjJz-84Q@mail.gmail.com>
References: <C3ABD749BB4545119E17C1257F270BA9@PBPC>
	<CAGvuWYYdNb-QUYwm7uK+L57XUm0D8spah10qm=wswitDLpSrxw@mail.gmail.com>
	<3F5D7CD6DA584E7F95D17F532A2E13A4@PBPC>
	<CAKzWOxVKLT9iSe1OCcy5tG+D+rqq_qiSEC5D=XNQKec8DSUm+g@mail.gmail.com>
	<B782F8192B3E46608CE6666897F86BEB@PBPC>
	<CAKzWOxUb9T8cB7G=eCr5hXoyycWcad-PH1sHewdtm-bHNTepwA@mail.gmail.com>
	<CAB+Kzh5wNV=xSzZmTpEHVAjkJ_96dseoDA_A82yFJZ-1vJevwQ@mail.gmail.com>
	<CAHgk4BUUrdG_527UQW3TUvATmt3Zp1=c8wz=MDwPEXZXmtosMw@mail.gmail.com>
	<F496A3CCE6994A408714AE2DCE31BAC6@PBPC>
	<0BB9E38C82674637982EDAF416B7B552@PBPC>
	<1599944384-1375845943-cardhu_decombobulator_blackberry.rim.net-1994125046-@b14.c5.bise6.blackberry>
	<5201D22B.4080104@xtronics.com>
	<CAA4nhyA4y2xKcP=7MgCv_oQ8zKy9ufE8dipVQQsFvnTWDkbQYQ@mail.gmail.com>
	<5202C0C7.7080502@xtronics.com>
	<CAKzWOxVZF+h002vGa75zZ59+m4V039pnojoxPz6JFv8jYdtkkw@mail.gmail.com>
	<5203F52E.7080502@xtronics.com>
	<CAKzWOxUdSGDoesNWXtHToMJ=rJZmT8i1D1T7b8mokQjjJz-84Q@mail.gmail.com>
Message-ID: <52054FE1.8020009@xtronics.com>

On 08/08/2013 03:06 PM, Michael Collins wrote:
> The default dialplan has an example of using *1 or *4 to do a blind or attn xfer respectively. Look
> for "Local_Extension" in conf/dialplan/default.xml.
>

Thanks - Yes, I know and use that feature - the bit I'm trying to understand is why linphone's 
transfer function works and ekiga and qutecom fail.

This might apply to stand-alone SIP phones - which I am shopping for - will the transfer button 
work? Either there is a similar bug in wheezy's version of both ekiga and qutecom or there is more 
than one way to do a transfer from a SIP clients perspective.
.,.

Also - this makes me think I should probably get a good book on SIP ( such reading also makes a 
wonderful cure for insomnia ) - any titles to recommend?



--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

Experience is something you never get,
until just after you needed it.
--------------------------------------------------------------------------------


From pasha at prosperity4ever.com  Sat Aug 10 04:56:23 2013
From: pasha at prosperity4ever.com (Paul)
Date: Sat, 10 Aug 2013 00:49:23 -0007
Subject: [Freeswitch-users] OpenWRT Router changes public IP to it's
 private IP
In-Reply-To: <CAHgk4BU=HcoesBu4qQKYAAygtTyi-ggL-26w6XKymzJfKmiMAw@mail.gmail.com>
References: <CAHgk4BVjSodyDpHYnuX_tE_DM=NbfaxzOnLruk=joNwhsq0XyQ@mail.gmail.com>
	<008c01ce9479$a0325590$e09700b0$@v10networks.ca>
	<CAHgk4BXLaUF3X0M5rpCxJ8q=TMd8g1AS_bGaZgwYqpuSy1o1KQ@mail.gmail.com>
	<00af01ce9482$0f36ec00$2da4c400$@v10networks.ca>
	<CAHgk4BU=HcoesBu4qQKYAAygtTyi-ggL-26w6XKymzJfKmiMAw@mail.gmail.com>
Message-ID: <20130810005634.438C657E002@mail.mydcs.ca>

Hi Brian,

Typically what happens is your external source sends the packet to your 
external IP (which is the WAN ip of the openWRT router), the router 
receives it, and if there are NAT rules defined, it will rewrite the 
source (in the DNAT case) to the internal IP of your FS (or webserver, 
etc), you can have it do it based on port it comes in on, ip it comes 
from, etc. It also should do the same on the way out of your network so 
that the 2 devices can have a bi-directional communication.

If your SIP packets on FS are coming with the source as 10.0.0.1 (your 
router's internal interface) it sounds to me like your NAT isn't 
configured right. Double check all of your NAT re-write rules.

I did this over an openVPN setup (haven't done it from WAN -> LAN on 
openWRT, but in the openWRT example I remember it being quite simple, 
you didn't have to write all the rules manually, you just had to enable 
masquerading on the proper zones and it took care of NAT.

I think you would have to write it manually in your case because it 
sounds like you're looking to route different traffic coming in to the 
same IP to different internal systems depending on what port they come 
in on.

Try something along the lines 
of:?https://forum.openwrt.org/viewtopic.php?id=35106

Not sure if I'm helping or confusing you more :)

Paul

On Thu, 8 Aug, 2013 at 3:11 PM, Brian Foster <bdfoster at davri.com> wrote:
> Like DNAT vs SNAT? Forwarding rules are set to DNAT in luci if that 
> is what you are referring to.
> 
> Thank you,
> 
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana
> 
> Sent from a mobile device.
> 
> On Aug 8, 2013 6:00 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:
>> You also may want to see if OpenWRT is doing something called 
>> symmetric NAT too. That can cause STUN?s port detection technique 
>> to fail completely.
>> 
>> ?
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org 
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
>> Brian Foster
>> Sent: Thursday, August 8, 2013 2:49 PM
>> To: FreeSWITCH Users Help
>> Subject: Re: [Freeswitch-users] OpenWRT Router changes public IP to 
>> it's private IP
>> 
>> ?
>> 
>> Stun is used on external profile, but I'll double check to see if 
>> they are correct.
>> 
>> Thank you,
>> 
>> Brian Foster
>> Project Manager/Owner's Rep.
>> Davri Investments, Inc.
>> O: 317-787-2686 x2102
>> M: 317-600-9753
>> E: bdfoster at davri.com
>> Indianapolis, Indiana
>> 
>> Sent from a mobile device.
>> 
>> On Aug 8, 2013 5:08 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:
>> 
>> Use STUN. Ext-rtp-ip and ext-sip-ip addresses are a great way to 
>> start looking into this.
>> 
>> ?
>> 
>> You can define stun servers as stun:stunserver.here.tld
>> 
>> ?
>> 
>> From: freeswitch-users-bounces at lists.freeswitch.org 
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
>> Brian Foster
>> Sent: Thursday, August 8, 2013 1:38 PM
>> To: FreeSWITCH Users Help
>> Subject: [Freeswitch-users] OpenWRT Router changes public IP to it's 
>> private IP
>> 
>> ?
>> 
>> We've got an openwrt router at a site, it's private IP is 10.0.0.1. 
>> Whenever we get an inbound call on the external profile (and 
>> probably outbound calls to, haven't checked yet), it changes the 
>> contact IP to the router's private IP, so that every call we get 
>> looks like it's coming from thr router. We've had some intermittent 
>> audio problems recently and we're trying to narrow things down a 
>> bit. Is this the expected behavior for NAT?
>> 
>> Sorry might seem like a dumb question, but my job entails much more 
>> than keeping the company's server's happy and I'm starting to lose 
>> my grip and I'm definitely not experienced with NAT.
>> 
>> All of my routers in the past have given FS the public IP of the 
>> contacting server. I have web servers behind the same NAT and doing 
>> the same thing, showing the private IP of the router instead of 
>> showing the public IP of the client. So I know it's not an issue 
>> with FS. Just wanted to see if anyone has exoerienced this before. 
>> Makes it difficult to use fail2ban on my servers, as it continually 
>> jails my router.
>> 
>> Thank you,
>> 
>> Brian Foster
>> Project Manager/Owner's Rep.
>> Davri Investments, Inc.
>> O: 317-787-2686 x2102
>> M: 317-600-9753
>> E: bdfoster at davri.com
>> Indianapolis, Indiana
>> 
>> Sent from a mobile device.
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 pasha at prosperity4ever.com  Sat Aug 10 05:01:14 2013
From: pasha at prosperity4ever.com (Paul)
Date: Sat, 10 Aug 2013 00:54:14 -0007
Subject: [Freeswitch-users] FreeSwitch and Yealink
In-Reply-To: <CAAQCs31MW_pFNiD=p=udTp4YmDueu=9rY8uDhozrYTU5JGy-uw@mail.gmail.com>
References: <20130807044439.9A5F8F4002@mail.mydcs.ca>
	<CAAQCs31MW_pFNiD=p=udTp4YmDueu=9rY8uDhozrYTU5JGy-uw@mail.gmail.com>
Message-ID: <20130810010116.3D80657E003@mail.mydcs.ca>

I'm gonna try all of your guys' suggestions.?

Ivan, I do not think its codec related, I made /log 11 and siptrace on 
all profiles (internal/external) as well as pcap capture of the phone 
itself. For some strange reason the moment the PBX send a packe tot ye 
Yealink phone that the remote party picked up and the call should be 
bridges, the Yealink phone replies with a BYE packet, I'm thinking it 
might be a bug in the Yealink firmware, so needless to say I have 
gotten a hold of their support and opened a ticket, see if something 
comes of that, but meanwhile going to try some of the suggestions in 
this thread and see if I can get any close to work around.

Paul

On Fri, 9 Aug, 2013 at 10:59 AM, Nikolay Rogoshchenkov 
<nickolayr at gmail.com> wrote:
> Check?RTP Packet Size too.
> 
> --
> Rogoshchenkov Nikolay
> 
> 
> On Wed, Aug 7, 2013 at 12:44 AM, Paul <pasha at prosperity4ever.com> 
> wrote:
>> Hi guys,
>> 
>> Has anyone had any issues using FreeSwitch with Yealink phones? My 
>> phones connect to FS via openvpn tunnel. All incoming calls work no 
>> problem, call comes through phones ring everyone can hear each other 
>> no issues, having a very strange issue though on the outgoing calls. 
>> As soon as the destination party picks up (this external calls) the 
>> call hangs up.
>> 
>> short FS LOG:
>> 
>> switch_ivr_bridge.c:475 Channel [sofia/internal/105 at 10.0.0.34] has 
>> been answered
>> sofia.c:6528 Hangup sofia/internal/105 at 10.0.0.34 [CS_EXECUTE] 
>> [ORIGINATOR_CANCEL]
>> switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 
>> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]?
>> 
>> So FS thinks the phone sent a BYE packet (which I can see with 
>> siptrace) but the phone's timer keeps going as if it thinks the call 
>> is supposed to keep going.
>> 
>> Internal extension to extension works fine (even if the extensions 
>> are at a different physical location and subnet).
>> 
>> I setup a second account to one of my asterisk servers and 
>> outgoing/incoming work just fine, so it seems this strange 
>> combination of FS and Yealink ... does it on 2 models T32G and T38G 
>> (only phones I have).
>> 
>> I have updated firwmare to their latest version (which in the 
>> comments say freeswitch ready)
>> 
>> Wondering if anyone else had any experience with these, or has some 
>> thoughts?
>> 
>> Thanks
>> 
>> Paul
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 bdfoster at davri.com  Sat Aug 10 10:29:35 2013
From: bdfoster at davri.com (Brian Foster)
Date: Sat, 10 Aug 2013 02:29:35 -0400
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <52049391.4080105@quentustech.com>
References: <5202C90D.3020308@quentustech.com>
	<CAMjaK3C=eYeNZsYDvjHg57zobNPOGiAVhkVSRyLHNg6mACO9gg@mail.gmail.com>
	<520317D1.3000409@gmail.com>
	<CAB+Kzh6477KJa4M9PxvrWWh=A4rcJnD7NttvqLcJKBUXctQneQ@mail.gmail.com>
	<52049391.4080105@quentustech.com>
Message-ID: <CAHgk4BXVnZA3i+Efp3w-JexNnr8PkwKCyWK02wGqgMKjfzrQ1Q@mail.gmail.com>

Thank you William. I will definitely check this out on Monday. Have a great
weekend.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 9, 2013 3:09 AM, "William King" <william.king at quentustech.com> wrote:

> >From what I've seen, retail level SMS carriers(not sms aggregators)
> usually offer a mix of HTTP, XMPP, SIP SIMPLE, or SMPP interfaces. I've
> seen the higher volume SMS connections prefer SMPP or HTTP.
>
> I'm exploring to see if there is a business case for FreeSWITCH support
> for SMPP(either direct support, or interface support through an
> independent application) that is not already covered by the current FS
> feature set.
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
> On 08/08/2013 04:23 AM, Raymond Chandler wrote:
> > So, to the best of my knowledge,  SMPP is strictly for SMS so you can
> > route SMS to your clients via FS to SIMPLE / dingaling / etc clients
> >
> > -Ray
> >
> > On Aug 7, 2013 11:05 PM, "Victor Chukalovskiy"
> > <victor.chukalovskiy at gmail.com <mailto:victor.chukalovskiy at gmail.com>>
> > wrote:
> >
> >     So am I as well as another ClueCon attendee I spoke to.
> >
> >     What is the usage scenario you are looking at?
> >
> >     On 13-08-07 06:46 PM, Dmitry Lysenko wrote:
> >>     Yes. I am interested.
> >>
> >>     Dmitry.
> >>
> >>
> >>     2013/8/8 William King <william.king at quentustech.com
> >>     <mailto:william.king at quentustech.com>>
> >>
> >>         Is anyone on this list interested in FreeSWITCH support for
> >>         SMPP for SMS
> >>         messages?
> >>
> >>         For more information about the SMPP protocol checkout:
> >>         http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
> >>
> >>         If so feel free to contact me on or off this list.
> >>         --
> >>         William King
> >>         Senior Engineer
> >>         Quentus Technologies, INC
> >>         1037 NE 65th St Suite 273
> >>         Seattle, WA 98115
> >>         Main:   (877) 211-9337 <tel:%28877%29%20211-9337>
> >>         Office: (206) 388-4772 <tel:%28206%29%20388-4772>
> >>         Cell:   (253) 686-5518 <tel:%28253%29%20686-5518>
> >>         william.king at quentustech.com <mailto:
> william.king at quentustech.com>
> >>
> >>
> _________________________________________________________________________
> >>         Professional FreeSWITCH Consulting Services:
> >>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >>         http://www.freeswitchsolutions.com
> >>
> >>         
> >>         
> >>
> >>         Official FreeSWITCH Sites
> >>         http://www.freeswitch.org
> >>         http://wiki.freeswitch.org
> >>         http://www.cluecon.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
> >>
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >>     Professional FreeSWITCH Consulting Services:
> >>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >>     http://www.freeswitchsolutions.com
> >>
> >>     
> >>     
> >>
> >>     Official FreeSWITCH Sites
> >>     http://www.freeswitch.org
> >>     http://wiki.freeswitch.org
> >>     http://www.cluecon.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
> >
> >
> >
> _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com
> >
> >     
> >     
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org
> >     http://wiki.freeswitch.org
> >     http://www.cluecon.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
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sravi123 at yahoo.com  Sat Aug 10 10:35:19 2013
From: sravi123 at yahoo.com (Ravi S)
Date: Sat, 10 Aug 2013 12:05:19 +0530
Subject: [Freeswitch-users] Wanpipe - Wanrouter help - freetdm and
	freeswitch
In-Reply-To: <CAFArqsb0KAMJsUCL71fxFBpSQ6SdCyuxDqmq1sRbBkiY+ksAjA@mail.gmail.com>
References: <1376051767.14626.YahooMailNeo@web160501.mail.bf1.yahoo.com>
	<CAFArqsb0KAMJsUCL71fxFBpSQ6SdCyuxDqmq1sRbBkiY+ksAjA@mail.gmail.com>
Message-ID: <5205DF27.7080306@yahoo.com>

Thanks Karsten for the response.

I tried he wiki pages on Sangoma. There is little to no documentation on 
the issue that I am facing. Through Sangoma standard support, I get one 
email response a day, at the most, which is slower than I would expect.  
Any quicker response, I have to buy their support, which I am willing to 
take if I am not able to resolve.

Thanks.
Ravi



On 09-08-2013 21:44, Karsten Horsmann wrote:
> Hi Ravi,
>
>
> best forum is read the fine sangoma wiki for proper driver 
> installation http://wiki.sangoma.com/ and if that didnt work, try the 
> nice sangoma support.
>
> Sangoma + FreeSWITCH is a working combination. Whats unknown is <??? 
> Linux Distro> and Wanpipe Driver <??? Version). That should you 
> collect before contact sangoma techdesk.
>
>
> 2013/8/9 Ravi <sravi123 at yahoo.com <mailto:sravi123 at yahoo.com>>
>
>     Hello Everyone !
>
>     I am not sure if this error is relevant for this mailing list. My
>     apologies if this is not the forum. I am trying to configure the
>     Sangoma card, with the freetdm installation for Freeswitch. When I
>     complete the wanpipe installation, and try wanrouter status, and
>     wanrouter start I get the following:
>
>     [root at bfree-server log]# wanrouter start
>
>     Starting up device: wanpipe1
>
>
>     wanconfig: WAN device wanpipe1 driver load failed !!
>     : ioctl(wanpipe1,ROUTER_SETUP) failed:
>     : 22 - Invalid argument
>
>
>     Wanpipe driver did not load properly
>     Please check /var/log/wanrouter and
>     /var/log/messages for errors
>
>     Configuring interfaces: w1g1 w1g1: unknown interface: No such device
>
>     done.
>     ___________________________________________________________________
>
>     Can anyone help me understand why this is happening ? And how to
>     resolve it.
>
>     Thanks.
>     Ravi
>
>
>
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
>
> -- 
> Mit freundlichen Gr??en
> *Karsten Horsmann*
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at davri.com  Sat Aug 10 10:48:33 2013
From: bdfoster at davri.com (Brian Foster)
Date: Sat, 10 Aug 2013 02:48:33 -0400
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <CAHgk4BXVnZA3i+Efp3w-JexNnr8PkwKCyWK02wGqgMKjfzrQ1Q@mail.gmail.com>
References: <5202C90D.3020308@quentustech.com>
	<CAMjaK3C=eYeNZsYDvjHg57zobNPOGiAVhkVSRyLHNg6mACO9gg@mail.gmail.com>
	<520317D1.3000409@gmail.com>
	<CAB+Kzh6477KJa4M9PxvrWWh=A4rcJnD7NttvqLcJKBUXctQneQ@mail.gmail.com>
	<52049391.4080105@quentustech.com>
	<CAHgk4BXVnZA3i+Efp3w-JexNnr8PkwKCyWK02wGqgMKjfzrQ1Q@mail.gmail.com>
Message-ID: <CAHgk4BVQp5i4mpGH7Px65Xoj+P4AC0z9XTcpEuGaAHNP_=u=wg@mail.gmail.com>

Yea it would help if I replied to tge correct thread...

Carry on!

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 10, 2013 2:29 AM, "Brian Foster" <bdfoster at davri.com> wrote:

> Thank you William. I will definitely check this out on Monday. Have a
> great weekend.
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana
>
> Sent from a mobile device.
> On Aug 9, 2013 3:09 AM, "William King" <william.king at quentustech.com>
> wrote:
>
>> >From what I've seen, retail level SMS carriers(not sms aggregators)
>> usually offer a mix of HTTP, XMPP, SIP SIMPLE, or SMPP interfaces. I've
>> seen the higher volume SMS connections prefer SMPP or HTTP.
>>
>> I'm exploring to see if there is a business case for FreeSWITCH support
>> for SMPP(either direct support, or interface support through an
>> independent application) that is not already covered by the current FS
>> feature set.
>>
>> William King
>> Senior Engineer
>> Quentus Technologies, INC
>> 1037 NE 65th St Suite 273
>> Seattle, WA 98115
>> Main:   (877) 211-9337
>> Office: (206) 388-4772
>> Cell:   (253) 686-5518
>> william.king at quentustech.com
>>
>> On 08/08/2013 04:23 AM, Raymond Chandler wrote:
>> > So, to the best of my knowledge,  SMPP is strictly for SMS so you can
>> > route SMS to your clients via FS to SIMPLE / dingaling / etc clients
>> >
>> > -Ray
>> >
>> > On Aug 7, 2013 11:05 PM, "Victor Chukalovskiy"
>> > <victor.chukalovskiy at gmail.com <mailto:victor.chukalovskiy at gmail.com>>
>> > wrote:
>> >
>> >     So am I as well as another ClueCon attendee I spoke to.
>> >
>> >     What is the usage scenario you are looking at?
>> >
>> >     On 13-08-07 06:46 PM, Dmitry Lysenko wrote:
>> >>     Yes. I am interested.
>> >>
>> >>     Dmitry.
>> >>
>> >>
>> >>     2013/8/8 William King <william.king at quentustech.com
>> >>     <mailto:william.king at quentustech.com>>
>> >>
>> >>         Is anyone on this list interested in FreeSWITCH support for
>> >>         SMPP for SMS
>> >>         messages?
>> >>
>> >>         For more information about the SMPP protocol checkout:
>> >>         http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
>> >>
>> >>         If so feel free to contact me on or off this list.
>> >>         --
>> >>         William King
>> >>         Senior Engineer
>> >>         Quentus Technologies, INC
>> >>         1037 NE 65th St Suite 273
>> >>         Seattle, WA 98115
>> >>         Main:   (877) 211-9337 <tel:%28877%29%20211-9337>
>> >>         Office: (206) 388-4772 <tel:%28206%29%20388-4772>
>> >>         Cell:   (253) 686-5518 <tel:%28253%29%20686-5518>
>> >>         william.king at quentustech.com <mailto:
>> william.king at quentustech.com>
>> >>
>> >>
>> _________________________________________________________________________
>> >>         Professional FreeSWITCH Consulting Services:
>> >>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> >>         http://www.freeswitchsolutions.com
>> >>
>> >>         
>> >>         
>> >>
>> >>         Official FreeSWITCH Sites
>> >>         http://www.freeswitch.org
>> >>         http://wiki.freeswitch.org
>> >>         http://www.cluecon.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
>> >>
>> >>
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >>     Professional FreeSWITCH Consulting Services:
>> >>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> >>     http://www.freeswitchsolutions.com
>> >>
>> >>     
>> >>     
>> >>
>> >>     Official FreeSWITCH Sites
>> >>     http://www.freeswitch.org
>> >>     http://wiki.freeswitch.org
>> >>     http://www.cluecon.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
>> >
>> >
>> >
>> _________________________________________________________________________
>> >     Professional FreeSWITCH Consulting Services:
>> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> >     http://www.freeswitchsolutions.com
>> >
>> >     
>> >     
>> >
>> >     Official FreeSWITCH Sites
>> >     http://www.freeswitch.org
>> >     http://wiki.freeswitch.org
>> >     http://www.cluecon.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
>> >
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.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
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 bdfoster at davri.com  Sat Aug 10 10:54:29 2013
From: bdfoster at davri.com (Brian Foster)
Date: Sat, 10 Aug 2013 02:54:29 -0400
Subject: [Freeswitch-users] OpenWRT Router changes public IP to it's
 private IP
In-Reply-To: <20130810005634.438C657E002@mail.mydcs.ca>
References: <CAHgk4BVjSodyDpHYnuX_tE_DM=NbfaxzOnLruk=joNwhsq0XyQ@mail.gmail.com>
	<008c01ce9479$a0325590$e09700b0$@v10networks.ca>
	<CAHgk4BXLaUF3X0M5rpCxJ8q=TMd8g1AS_bGaZgwYqpuSy1o1KQ@mail.gmail.com>
	<00af01ce9482$0f36ec00$2da4c400$@v10networks.ca>
	<CAHgk4BU=HcoesBu4qQKYAAygtTyi-ggL-26w6XKymzJfKmiMAw@mail.gmail.com>
	<20130810005634.438C657E002@mail.mydcs.ca>
Message-ID: <CAHgk4BWN4gt0wpAp3OBOvjkP8oVLR5Zj1RfkVNxNXCsRdR=-2g@mail.gmail.com>

Paul, thanks for the reply. I'll definitely check this out on Monday.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 9, 2013 9:03 PM, "Paul" <pasha at prosperity4ever.com> wrote:

> Hi Brian,
>
> Typically what happens is your external source sends the packet to your
> external IP (which is the WAN ip of the openWRT router), the router
> receives it, and if there are NAT rules defined, it will rewrite the source
> (in the DNAT case) to the internal IP of your FS (or webserver, etc), you
> can have it do it based on port it comes in on, ip it comes from, etc. It
> also should do the same on the way out of your network so that the 2
> devices can have a bi-directional communication.
>
> If your SIP packets on FS are coming with the source as 10.0.0.1 (your
> router's internal interface) it sounds to me like your NAT isn't configured
> right. Double check all of your NAT re-write rules.
>
> I did this over an openVPN setup (haven't done it from WAN -> LAN on
> openWRT, but in the openWRT example I remember it being quite simple, you
> didn't have to write all the rules manually, you just had to enable
> masquerading on the proper zones and it took care of NAT.
>
> I think you would have to write it manually in your case because it sounds
> like you're looking to route different traffic coming in to the same IP to
> different internal systems depending on what port they come in on.
>
> Try something along the lines of:
> https://forum.openwrt.org/viewtopic.php?id=35106
>
> Not sure if I'm helping or confusing you more :)
>
> Paul
>
> On Thu, 8 Aug, 2013 at 3:11 PM, Brian Foster <bdfoster at davri.com> wrote:
>
> Like DNAT vs SNAT? Forwarding rules are set to DNAT in luci if that is
> what you are referring to.
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana
>
> Sent from a mobile device.
> On Aug 8, 2013 6:00 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:
>
>> You also may want to see if OpenWRT is doing something called symmetric
>> NAT too. That can cause STUN?s port detection technique to fail completely.
>> ****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian
>> Foster
>> *Sent:* Thursday, August 8, 2013 2:49 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] OpenWRT Router changes public IP to
>> it's private IP****
>>
>> ** **
>>
>> Stun is used on external profile, but I'll double check to see if they
>> are correct.****
>>
>> Thank you,****
>>
>> Brian Foster
>> Project Manager/Owner's Rep.
>> Davri Investments, Inc.
>> O: 317-787-2686 x2102
>> M: 317-600-9753
>> E: bdfoster at davri.com
>> Indianapolis, Indiana****
>>
>> Sent from a mobile device.****
>>
>> On Aug 8, 2013 5:08 PM, "Jeff Leung" <jleung at v10networks.ca> wrote:****
>>
>> Use STUN. Ext-rtp-ip and ext-sip-ip addresses are a great way to start
>> looking into this.****
>>
>>  ****
>>
>> You can define stun servers as stun:stunserver.here.tld****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian
>> Foster
>> *Sent:* Thursday, August 8, 2013 1:38 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] OpenWRT Router changes public IP to it's
>> private IP****
>>
>>  ****
>>
>> We've got an openwrt router at a site, it's private IP is 10.0.0.1.
>> Whenever we get an inbound call on the external profile (and probably
>> outbound calls to, haven't checked yet), it changes the contact IP to the
>> router's private IP, so that every call we get looks like it's coming from
>> thr router. We've had some intermittent audio problems recently and we're
>> trying to narrow things down a bit. Is this the expected behavior for NAT?
>> ****
>>
>> Sorry might seem like a dumb question, but my job entails much more than
>> keeping the company's server's happy and I'm starting to lose my grip and
>> I'm definitely not experienced with NAT.****
>>
>> All of my routers in the past have given FS the public IP of the
>> contacting server. I have web servers behind the same NAT and doing the
>> same thing, showing the private IP of the router instead of showing the
>> public IP of the client. So I know it's not an issue with FS. Just wanted
>> to see if anyone has exoerienced this before. Makes it difficult to use
>> fail2ban on my servers, as it continually jails my router.****
>>
>> Thank you,****
>>
>> Brian Foster
>> Project Manager/Owner's Rep.
>> Davri Investments, Inc.
>> O: 317-787-2686 x2102
>> M: 317-600-9753
>> E: bdfoster at davri.com
>> Indianapolis, Indiana****
>>
>> Sent from a mobile device.****
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveayre at gmail.com  Sat Aug 10 12:54:59 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Sat, 10 Aug 2013 09:54:59 +0100
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAGvuWYbACF=vR=wMUBG5u1qY4HoGEEqApT0t-4FEa6AzSecMHw@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<52001B43.9080901@freeswitch.org>
	<CAMKwYLCtQV1wCVzPxwhNj2d23eK__uPDYPb98p0ZiJLRqx0GKQ@mail.gmail.com>
	<CAMvhjJt8v0XVnGqBs5pMgEyNR3tDV6GuMFR5Dsb1x6XMCMO4tw@mail.gmail.com>
	<CABkS=xptY82XhGFxfOh5ED1P04mZtxKV3Fiy2KmwT+x=8oGmAA@mail.gmail.com>
	<CAMvhjJuY3NSinMpu7N=wzh7Y+6zUxw4UfsGFhM6j-ECzY4tXjw@mail.gmail.com>
	<CAFiqYumS-4QvsgcH48TugtUC3__2PYQSk7Sg-QZru8qDikoFHw@mail.gmail.com>
	<CAGvuWYbACF=vR=wMUBG5u1qY4HoGEEqApT0t-4FEa6AzSecMHw@mail.gmail.com>
Message-ID: <CAFiqYummdrTUZNwLjYXw__PNLYe=qvH4-7nEicFuff31BSphcw@mail.gmail.com>

That's not strictly speaking true. It is possible to firewall localhost
connections. Taking iptables as an example unless there's a rule to allow
all connections from lo it'll be treated just like any other.

-Steve



On 7 August 2013 16:50, Lloyd Aloysius <lloyd.aloysius at sunteltech.ca> wrote:

> If you are accessing fs_cli locally , no need to worry about your
> firewall. By default fs_cli should work. Check freeswitch process status.
>
> You can try the following method to test freeswitch , go to your
> freeswitch/bin folder and run ./freeswitch manually and see the switch is
> loading without any issue.
>
> Lloyd
>
>
> *
> *
> *  <http://www.sunteltech.ca/blog/>*
>
>
> On Wed, Aug 7, 2013 at 10:11 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> 1) Check mod_event_socket is enabled to load on start in
>> autoload_configs/modules.conf.xml
>>
>> 2) Check freeswitch is running:
>> $ ps -C freeswitch -f
>> UID        PID  PPID  C STIME TTY          TIME CMD
>> 999      12993     1  1 14:44 ?        00:00:23 /usr/bin/freeswitch -u
>> freeswitch -g freeswitch -nc -rp -nonat
>>
>> 3) Check freeswitch is listening on the ESL socket:
>> $ netstat -anp | grep :8021
>> tcp        0      0 127.0.0.1:8021          0.0.0.0:*
>> LISTEN      12993/freeswitch
>>
>> 4) Check the freeswitch.log file to check mod_event_socket tried to load,
>> and for any errors while loading. Or for startup errors if freeswitch isn't
>> running at all.
>>
>> Note if it's listening on 127.0.0.1:8021 then you won't be able to
>> connect from remote machines. If you need that set listen-ip to 0.0.0.0 and
>> make sure you change the default password and you should also consider
>> using ACLs or a firewall to restrict who can connect.
>>
>> -Steve
>>
>>
>>
>> On 7 August 2013 08:21, Ashish Mishra <itsme.kunnu at gmail.com> wrote:
>>
>>> Sir i would like to give you some information about how am i launching
>>> fs_cli...I have installed freeswitch in my ubuntu machine now on the same
>>> machine i am trying to launch fs_cli but i am getting an error which i have
>>> already mentioned...Also as far as the output of iptables -L is concerned i
>>> am getiing:
>>>
>>> Chain INPUT (policy ACCEPT)
>>> target        prot opt source        destination
>>>
>>> Chain FORWARD (policy ACCEPT)
>>> target        prot opt source        destination
>>>
>>> Chain OUTPUT (policy ACCEPT)
>>> target        prot opt source        destination
>>>
>>>
>>> On Wed, Aug 7, 2013 at 4:44 AM, Peter <eidevm5 at gmail.com> wrote:
>>>
>>>> Do you have a firewall running?
>>>>
>>>> What does
>>>>
>>>> iptables -L
>>>>
>>>> display?
>>>>
>>>>
>>>> On Tue, Aug 6, 2013 at 5:51 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>
>>>>> Yes my freeswitch is running...
>>>>> On Aug 6, 2013 3:41 AM, "Gustavo Salazar" <guga.salazar.loor at gmail.com>
>>>>> wrote:
>>>>>
>>>>>>
>>>>>>
>>>>>> Is Freeswitch running?
>>>>>> I have seen a similar error when I tried to start the cli and
>>>>>> Freeswitch was not running .
>>>>>>
>>>>>>
>>>>>>
>>>>>> 2013/8/5 Yossi Neiman <mishehu at freeswitch.org>
>>>>>>
>>>>>>>  Do you have mod_event_socket loaded?
>>>>>>>
>>>>>>> -Yossi
>>>>>>>
>>>>>>>
>>>>>>> On 08/05/2013 04:00 PM, Ashish Mishra wrote:
>>>>>>>
>>>>>>> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on
>>>>>>> which i have installed freeswitch) it gives me the following error :
>>>>>>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Gustavo Salazar
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 khorsmann at gmail.com  Sat Aug 10 21:54:13 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Sat, 10 Aug 2013 19:54:13 +0200
Subject: [Freeswitch-users] WebRTC sipml5 client hangs up after 120 seconds
 on bridged calls to sip phone
Message-ID: <CAFArqsY-sBqFYu1xAw_K48tTrNHKxxAYy7ZYj3pLF_wm7p+WUA@mail.gmail.com>

Hello,

i play around with freeswitch master on centos 6.4 and webrtc to sip-phones.
if i call from the browser (chrome) to a snom, the call ends after 120
seconds.
I see that the sipml5 client is sending then a bye message.

The other direction works fine. From snom to the webrtc phone, no hangup
after 120 seconds.
And from webrtc to the moh extension works fine too.

Any special settings to resolve that?

I made an sofia trace, both phones are on the same sip-profile.
http://pastebin.freeswitch.org/21291


my git version is d1268e81036ce8ce00de8ee22f387cdbf43a7203

--
Kind Regards
Mit freundlichen Gr??en
*Karsten Horsmann*
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From smontour at verizon.net  Fri Aug  9 22:29:48 2013
From: smontour at verizon.net (Sami Montour)
Date: Fri, 09 Aug 2013 13:29:48 -0500
Subject: [Freeswitch-users] Real-time billing application for the FreeSWITCH
 (mod_lua, mod_perl or ESL)
Message-ID: <002101ce952e$6e3fc3f0$4abf4bd0$@verizon.net>

Hello Everyone,

I am a long time user of OpenSIPS and have recently been tasked with
developing a call control/real-time billing application for the FreeSWITCH.
I would like to develop the application in either C/C++, Perl or lua
(totally new to lua) in this order. 

 

I have used ESL to develop some basic functionality of the application using
Perl. For instance, when the call comes in, I pass over control from
FreeSWITCH to the call control application using outbound TCP socket. Then
the call control application listens for CHANNEL_ANSWER event to start
billing process and CHANNEL_HANGUP event to stop billing. The application
does some database work as well to retrieve and check user status and
balance and all that stuff. My main concern is with load testing because of
the TCP connection between the FreeSWITCH and call control application and
also TCP connection to database to retrieve subscriber info. The basic
functionality seems to work fine but haven't tested it with high call
volume, which I am about to do.

 

The last couple of days I have been experimenting with mod_lua but can't
find a method under "Session" to get information related to CHANNEL_ANSWER
and CHANNEL_HANGUP events. The "session:geVariable" does not have variables
related to session answer time or hangup time, something similar to
"$e->getHeader("Caller-Channel-Answered-Time")"  in ESL. I must be missing
something very simple here. Again, my experience with mod_lua is only a
couple of days old.

 

I am new Freeswitch and thought I would seek the advice of all the experts
and people with more experience with Freeswitch. For this type of
application, am I better off with mod_lua, or mod_perl, or shall I stick
with ESL using Perl? How about ESL with C/C++? Will I gain some performance
gain there?

 

Any feedback would be appreciated. Thanks.

 

Sam

 

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From crazygabry at gmail.com  Sat Aug 10 02:32:05 2013
From: crazygabry at gmail.com (crazygabry)
Date: Fri, 9 Aug 2013 15:32:05 -0700 (PDT)
Subject: [Freeswitch-users] Remote Extension Registered but can't
	make/receive any call
Message-ID: <1376087525955-7593777.post@n2.nabble.com>

Hello every body,

i'm a new freepbx user and i'm trying to configured a freeswitch server
hosted in the cloud.
I can register my softphone but i can't do anything: when i try to make a
call i receive an error from the CLI that says INVALID_PROFILE.
Can someone help me?

Thanks in advice
crazygabry

P.S. that's what i see on CLI ( XXX.XXX.XXX is my local ip and YYY.YYY.YYY
is the freeswitch IP)

2013-08-09 22:28:10.070676 [NOTICE] switch_channel.c:1030 New Channel
sofia/external/1100 at YYY.YYY.YYY.YYY:9060
[dc15c333-fa2c-4055-a56e-f466279c8aae]
2013-08-09 22:28:10.070676 [DEBUG] switch_core_session.c:999 Send signal
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
2013-08-09 22:28:10.070676 [DEBUG] switch_core_session.c:999 Send signal
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
2013-08-09 22:28:10.070676 [DEBUG] switch_core_state_machine.c:415
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Running State Change CS_NEW
2013-08-09 22:28:10.070676 [DEBUG] switch_core_state_machine.c:433
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State NEW
2013-08-09 22:28:10.090709 [DEBUG] sofia.c:5727 Channel
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 entering state [received][100]
2013-08-09 22:28:10.090709 [DEBUG] sofia.c:5740 Remote SDP:
v=0
o=3cxVCE 303008025 298674930 IN IP4 XXX.XXX.XXX.XXX
s=3cxVCE Audio Call
c=IN IP4 XXX.XXX.XXX.XXX
t=0 0
m=audio 40046 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=video 40004 RTP/AVP 34
c=IN IP4 XXX.XXX.XXX.XXX
a=rtpmap:34 H263/90000
a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1

2013-08-09 22:28:10.090709 [DEBUG] sofia.c:5954
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State Change CS_NEW -> CS_INIT
2013-08-09 22:28:10.090709 [DEBUG] switch_core_session.c:1334 Send signal
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:415
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Running State Change CS_INIT
2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:454
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State INIT
2013-08-09 22:28:10.090709 [DEBUG] mod_sofia.c:87
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 SOFIA INIT
2013-08-09 22:28:10.090709 [DEBUG] mod_sofia.c:127
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State Change CS_INIT ->
CS_ROUTING
2013-08-09 22:28:10.090709 [DEBUG] switch_core_session.c:1334 Send signal
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:454
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State INIT going to sleep
2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:415
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Running State Change CS_ROUTING
2013-08-09 22:28:10.090709 [DEBUG] switch_channel.c:2111
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Callstate Change DOWN -> RINGING
2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:470
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State ROUTING
2013-08-09 22:28:10.090709 [DEBUG] mod_sofia.c:150
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 SOFIA ROUTING
2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:117
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Standard ROUTING
2013-08-09 22:28:10.090709 [INFO] mod_dialplan_xml.c:558 Processing Gwen
<1100>->1000 in context public
Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 parsing [public->unloop]
continue=false
Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Regex (PASS) [unloop]
${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Regex (FAIL) [unloop]
${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 parsing
[public->outside_call] continue=true
Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Absolute Condition
[outside_call]
Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Action
set(outside_call=true)
Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Action
export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 parsing
[public->call_debug] continue=true
Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Regex (FAIL) [call_debug]
${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 parsing
[public->public_extensions] continue=false
Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Regex (PASS)
[public_extensions] destination_number(1000) =~ /^(10[01][0-9])$/
break=on-false
Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Action
bridge(sofia/doublenat/1000%172.29.5.7 )
2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:167
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State Change CS_ROUTING ->
CS_EXECUTE
2013-08-09 22:28:10.090709 [DEBUG] switch_core_session.c:1334 Send signal
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:470
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State ROUTING going to sleep
2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:415
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Running State Change CS_EXECUTE
2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:477
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State EXECUTE
2013-08-09 22:28:10.090709 [DEBUG] mod_sofia.c:243
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 SOFIA EXECUTE
2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:209
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Standard EXECUTE
EXECUTE sofia/external/1100 at YYY.YYY.YYY.YYY:9060 set(outside_call=true)
2013-08-09 22:28:10.090709 [DEBUG] mod_dptools.c:1393
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 SET [outside_call]=[true]
EXECUTE sofia/external/1100 at YYY.YYY.YYY.YYY:9060 export(RFC2822_DATE=Fri, 09
Aug 2013 22:28:10 +0000)
2013-08-09 22:28:10.090709 [DEBUG] switch_channel.c:1222 EXPORT
(export_vars) [RFC2822_DATE]=[Fri, 09 Aug 2013 22:28:10 +0000]
EXECUTE sofia/external/1100 at YYY.YYY.YYY.YYY:9060
bridge(sofia/doublenat/1000%172.29.5.7 )
2013-08-09 22:28:10.090709 [DEBUG] switch_channel.c:1176
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 EXPORTING[export_vars]
[RFC2822_DATE]=[Fri, 09 Aug 2013 22:28:10 +0000] to event
2013-08-09 22:28:10.090709 [DEBUG] switch_ivr_originate.c:2060 Parsing
global variables
2013-08-09 22:28:10.090709 [ERR] mod_sofia.c:5024 Invalid Profile
2013-08-09 22:28:10.090709 [NOTICE] mod_sofia.c:5325 Close Channel N/A
[CS_NEW]
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:568 ()
Running State Change CS_DESTROY
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:578 (N/A)
State DESTROY
2013-08-09 22:28:10.110663 [DEBUG] mod_sofia.c:397 N/A SOFIA DESTROY
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:578 (N/A)
State DESTROY going to sleep
2013-08-09 22:28:10.110663 [NOTICE] switch_ivr_originate.c:2661 Cannot
create outgoing channel of type [sofia] cause: [INVALID_PROFILE]
2013-08-09 22:28:10.110663 [DEBUG] switch_ivr_originate.c:3632 Originate
Resulted in Error Cause: 611 [INVALID_PROFILE]
2013-08-09 22:28:10.110663 [INFO] mod_dptools.c:3192 Originate Failed. 
Cause: INVALID_PROFILE
2013-08-09 22:28:10.110663 [NOTICE] switch_channel.c:4573 Hangup
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [CS_EXECUTE] [INVALID_PROFILE]
2013-08-09 22:28:10.110663 [DEBUG] switch_channel.c:3130 Send signal
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [KILL]
2013-08-09 22:28:10.110663 [DEBUG] switch_core_session.c:1334 Send signal
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
2013-08-09 22:28:10.110663 [DEBUG] switch_core_session.c:2740
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:477
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State EXECUTE going to sleep
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:415
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Running State Change CS_HANGUP
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:678
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State HANGUP
2013-08-09 22:28:10.110663 [DEBUG] mod_sofia.c:504 Channel
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 hanging up, cause: INVALID_PROFILE
2013-08-09 22:28:10.110663 [DEBUG] mod_sofia.c:638 Responding to INVITE
with: 502
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:48
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Standard HANGUP, cause:
INVALID_PROFILE
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:678
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State HANGUP going to sleep
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:689
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Callstate Change RINGING ->
HANGUP
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:446
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State Change CS_HANGUP ->
CS_REPORTING
2013-08-09 22:28:10.110663 [DEBUG] switch_core_session.c:1334 Send signal
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:415
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Running State Change CS_REPORTING
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:761
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State REPORTING
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:92
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Standard REPORTING, cause:
INVALID_PROFILE
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:761
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State REPORTING going to sleep
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:440
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State Change CS_REPORTING ->
CS_DESTROY
2013-08-09 22:28:10.110663 [DEBUG] switch_core_session.c:1334 Send signal
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
2013-08-09 22:28:10.110663 [DEBUG] switch_core_session.c:1542 Session 26
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Locked, Waiting on external
entities
2013-08-09 22:28:10.110663 [NOTICE] switch_core_session.c:1560 Session 26
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Ended
2013-08-09 22:28:10.110663 [NOTICE] switch_core_session.c:1564 Close Channel
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [CS_DESTROY]
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:565
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Callstate Change HANGUP -> DOWN
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:568
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Running State Change CS_DESTROY
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:578
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State DESTROY
2013-08-09 22:28:10.110663 [DEBUG] mod_sofia.c:397
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 SOFIA DESTROY
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:99
sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Standard DESTROY
2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:578
(sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State DESTROY going to sleep






--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Remote-Extension-Registered-but-can-t-make-receive-any-call-tp7593777.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From markab242 at gmail.com  Sat Aug 10 10:40:32 2013
From: markab242 at gmail.com (Mark Berry)
Date: Sat, 10 Aug 2013 00:40:32 -0600
Subject: [Freeswitch-users] mod_callcenter - cc_warning_tone not working
In-Reply-To: <31A79B0B4414EB4B83C7EAF307ED57DA04D9FFAB@prod-exch01.corp.vseinc.com>
References: <31A79B0B4414EB4B83C7EAF307ED57DA04D9FB47@prod-exch01.corp.vseinc.com>
	<31A79B0B4414EB4B83C7EAF307ED57DA04D9FFAB@prod-exch01.corp.vseinc.com>
Message-ID: <CA+hLmVoXtdjKHHLX81bQvJV=pefrXYX8DFaZxcXJ2Ygg2V3kSg@mail.gmail.com>

I've had the same problem.  What I ended up having to do was have a
pre-bridge execute execute a script that displaces an audio beep to the
agent manually. Though it's not elegant , I'm able to achieve what i'm
needing with this method.


On Wed, Jul 24, 2013 at 7:42 AM, <bedgar at vseinc.com> wrote:

> All,****
>
> ** **
>
> Hopefully Marc Olivier Chouinard will catch this thread and have an answer
> for me?****
>
> ** **
>
> Brian****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian Edgar
> *Sent:* Monday, July 22, 2013 3:23 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] mod_callcenter - cc_warning_tone not working
> ****
>
> ** **
>
> Hello Experts,****
>
> ** **
>
> I have been working on setting up a call center in FS and am having a
> problem with the warning tone notification.  It does not play any tones
> before the caller is bridged to the agent.  I am using the latest stable
> build for 1.2 and I am using the example in the wiki: ****
>
> ** **
>
> https://wiki.freeswitch.org/wiki/Mod_callcenter#uuid-standby****
>
> ** **
>
> Snippet:****
>
> ** **
>
> <extension>****
>
>   <condition field="destination_number" expression="^(4099)$">****
>
>     <action application="set" data="transfer_after_bridge=4099"/> <!--
> Remove this if you just want to get a single call ****
>
>     <action application="sleep" data="300"/> <!-- Small delay for safety
> needs -->****
>
>     <action application="set" data="res=${callcenter_config(agent set uuid
> ${caller_id_number}@${domain_name} '${uuid}')}" />****
>
>     <action application="set" data="res=${callcenter_config(agent set type
> ${caller_id_number}@${domain_name} 'uuid-standby')}" />****
>
>     <action application="set" data="res=${callcenter_config(agent set
> status ${caller_id_number}@${domain_name} 'Available (On Demand)')}" />***
> *
>
>     <action application="set" data="res=${callcenter_config(agent set
> state ${caller_id_number}@${domain_name} 'Waiting')}" />****
>
>     <action application="set"
> data="cc_warning_tone=tone_stream://%(200,0,500,600,700)"/>****
>
>     <action application="answer" />****
>
>     <action application="playback" data="$${hold_music}"/>****
>
> ** **
>
>   </condition>****
>
> </extension>****
>
> ** **
>
> I have tried changing the tone values to see if making it repeat or play
> longer would produce a positive result.  No dice?****
>
> ** **
>
> On a second note, is there a way to play a file to the agent before the
> call is bridged?  My call center services multiple brands and would like to
> have the agent greet the caller appropriately.****
>
> ** **
>
> Thank you,****
>
> ** **
>
> Brian Edgar****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 khorsmann at gmail.com  Sat Aug 10 22:35:13 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Sat, 10 Aug 2013 20:35:13 +0200
Subject: [Freeswitch-users] Remote Extension Registered but can't
 make/receive any call
In-Reply-To: <1376087525955-7593777.post@n2.nabble.com>
References: <1376087525955-7593777.post@n2.nabble.com>
Message-ID: <CAFArqsajN6xhO4xxquX+JxocLq4Za8vZhC_A3Ag+HUh0BJeJfQ@mail.gmail.com>

Hi,

successful registered users only means that your user/password in the phone
are the same as in the directory.

It not means your dialplan is correct.

it seems that the bridge target is not there, the cli told you INVALID_PROFILE
after tried to use "doublenat" profile.

Check your sip_profile.

And please use http://pastebin.freeswitch.org/ for logs (username/password
show up in the login window).




2013/8/10 crazygabry <crazygabry at gmail.com>

> Hello every body,
>
> i'm a new freepbx user and i'm trying to configured a freeswitch server
> hosted in the cloud.
> I can register my softphone but i can't do anything: when i try to make a
> call i receive an error from the CLI that says INVALID_PROFILE.
> Can someone help me?
>
> Thanks in advice
> crazygabry
>
> P.S. that's what i see on CLI ( XXX.XXX.XXX is my local ip and YYY.YYY.YYY
> is the freeswitch IP)
>
> 2013-08-09 22:28:10.070676 [NOTICE] switch_channel.c:1030 New Channel
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060
> [dc15c333-fa2c-4055-a56e-f466279c8aae]
> 2013-08-09 22:28:10.070676 [DEBUG] switch_core_session.c:999 Send signal
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
> 2013-08-09 22:28:10.070676 [DEBUG] switch_core_session.c:999 Send signal
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
> 2013-08-09 22:28:10.070676 [DEBUG] switch_core_state_machine.c:415
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Running State Change CS_NEW
> 2013-08-09 22:28:10.070676 [DEBUG] switch_core_state_machine.c:433
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State NEW
> 2013-08-09 22:28:10.090709 [DEBUG] sofia.c:5727 Channel
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 entering state [received][100]
> 2013-08-09 22:28:10.090709 [DEBUG] sofia.c:5740 Remote SDP:
> v=0
> o=3cxVCE 303008025 298674930 IN IP4 XXX.XXX.XXX.XXX
> s=3cxVCE Audio Call
> c=IN IP4 XXX.XXX.XXX.XXX
> t=0 0
> m=audio 40046 RTP/AVP 3 0 8 101
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> m=video 40004 RTP/AVP 34
> c=IN IP4 XXX.XXX.XXX.XXX
> a=rtpmap:34 H263/90000
> a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
>
> 2013-08-09 22:28:10.090709 [DEBUG] sofia.c:5954
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State Change CS_NEW -> CS_INIT
> 2013-08-09 22:28:10.090709 [DEBUG] switch_core_session.c:1334 Send signal
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
> 2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:415
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Running State Change CS_INIT
> 2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:454
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State INIT
> 2013-08-09 22:28:10.090709 [DEBUG] mod_sofia.c:87
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 SOFIA INIT
> 2013-08-09 22:28:10.090709 [DEBUG] mod_sofia.c:127
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State Change CS_INIT ->
> CS_ROUTING
> 2013-08-09 22:28:10.090709 [DEBUG] switch_core_session.c:1334 Send signal
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
> 2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:454
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State INIT going to sleep
> 2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:415
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Running State Change CS_ROUTING
> 2013-08-09 22:28:10.090709 [DEBUG] switch_channel.c:2111
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Callstate Change DOWN ->
> RINGING
> 2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:470
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State ROUTING
> 2013-08-09 22:28:10.090709 [DEBUG] mod_sofia.c:150
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 SOFIA ROUTING
> 2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:117
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Standard ROUTING
> 2013-08-09 22:28:10.090709 [INFO] mod_dialplan_xml.c:558 Processing Gwen
> <1100>->1000 in context public
> Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 parsing
> [public->unloop]
> continue=false
> Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Regex (PASS) [unloop]
> ${unroll_loops}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Regex (FAIL) [unloop]
> ${sip_looped_call}() =~ /^true$/ break=on-false
> Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 parsing
> [public->outside_call] continue=true
> Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Absolute Condition
> [outside_call]
> Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Action
> set(outside_call=true)
> Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Action
> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)})
> Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 parsing
> [public->call_debug] continue=true
> Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Regex (FAIL)
> [call_debug]
> ${call_debug}(false) =~ /^true$/ break=never
> Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 parsing
> [public->public_extensions] continue=false
> Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Regex (PASS)
> [public_extensions] destination_number(1000) =~ /^(10[01][0-9])$/
> break=on-false
> Dialplan: sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Action
> bridge(sofia/doublenat/1000%172.29.5.7 )
> 2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:167
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State Change CS_ROUTING ->
> CS_EXECUTE
> 2013-08-09 22:28:10.090709 [DEBUG] switch_core_session.c:1334 Send signal
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
> 2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:470
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State ROUTING going to sleep
> 2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:415
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Running State Change CS_EXECUTE
> 2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:477
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State EXECUTE
> 2013-08-09 22:28:10.090709 [DEBUG] mod_sofia.c:243
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 SOFIA EXECUTE
> 2013-08-09 22:28:10.090709 [DEBUG] switch_core_state_machine.c:209
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Standard EXECUTE
> EXECUTE sofia/external/1100 at YYY.YYY.YYY.YYY:9060 set(outside_call=true)
> 2013-08-09 22:28:10.090709 [DEBUG] mod_dptools.c:1393
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 SET [outside_call]=[true]
> EXECUTE sofia/external/1100 at YYY.YYY.YYY.YYY:9060 export(RFC2822_DATE=Fri,
> 09
> Aug 2013 22:28:10 +0000)
> 2013-08-09 22:28:10.090709 [DEBUG] switch_channel.c:1222 EXPORT
> (export_vars) [RFC2822_DATE]=[Fri, 09 Aug 2013 22:28:10 +0000]
> EXECUTE sofia/external/1100 at YYY.YYY.YYY.YYY:9060
> bridge(sofia/doublenat/1000%172.29.5.7 )
> 2013-08-09 22:28:10.090709 [DEBUG] switch_channel.c:1176
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 EXPORTING[export_vars]
> [RFC2822_DATE]=[Fri, 09 Aug 2013 22:28:10 +0000] to event
> 2013-08-09 22:28:10.090709 [DEBUG] switch_ivr_originate.c:2060 Parsing
> global variables
> 2013-08-09 22:28:10.090709 [ERR] mod_sofia.c:5024 Invalid Profile
> 2013-08-09 22:28:10.090709 [NOTICE] mod_sofia.c:5325 Close Channel N/A
> [CS_NEW]
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:568 ()
> Running State Change CS_DESTROY
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:578 (N/A)
> State DESTROY
> 2013-08-09 22:28:10.110663 [DEBUG] mod_sofia.c:397 N/A SOFIA DESTROY
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:578 (N/A)
> State DESTROY going to sleep
> 2013-08-09 22:28:10.110663 [NOTICE] switch_ivr_originate.c:2661 Cannot
> create outgoing channel of type [sofia] cause: [INVALID_PROFILE]
> 2013-08-09 22:28:10.110663 [DEBUG] switch_ivr_originate.c:3632 Originate
> Resulted in Error Cause: 611 [INVALID_PROFILE]
> 2013-08-09 22:28:10.110663 [INFO] mod_dptools.c:3192 Originate Failed.
> Cause: INVALID_PROFILE
> 2013-08-09 22:28:10.110663 [NOTICE] switch_channel.c:4573 Hangup
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [CS_EXECUTE] [INVALID_PROFILE]
> 2013-08-09 22:28:10.110663 [DEBUG] switch_channel.c:3130 Send signal
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [KILL]
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_session.c:1334 Send signal
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_session.c:2740
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:477
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State EXECUTE going to sleep
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:415
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Running State Change CS_HANGUP
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:678
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State HANGUP
> 2013-08-09 22:28:10.110663 [DEBUG] mod_sofia.c:504 Channel
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 hanging up, cause:
> INVALID_PROFILE
> 2013-08-09 22:28:10.110663 [DEBUG] mod_sofia.c:638 Responding to INVITE
> with: 502
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:48
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Standard HANGUP, cause:
> INVALID_PROFILE
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:678
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State HANGUP going to sleep
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:689
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Callstate Change RINGING ->
> HANGUP
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:446
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State Change CS_HANGUP ->
> CS_REPORTING
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_session.c:1334 Send signal
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:415
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Running State Change
> CS_REPORTING
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:761
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State REPORTING
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:92
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Standard REPORTING, cause:
> INVALID_PROFILE
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:761
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State REPORTING going to sleep
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:440
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State Change CS_REPORTING ->
> CS_DESTROY
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_session.c:1334 Send signal
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [BREAK]
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_session.c:1542 Session 26
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Locked, Waiting on external
> entities
> 2013-08-09 22:28:10.110663 [NOTICE] switch_core_session.c:1560 Session 26
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Ended
> 2013-08-09 22:28:10.110663 [NOTICE] switch_core_session.c:1564 Close
> Channel
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 [CS_DESTROY]
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:565
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Callstate Change HANGUP -> DOWN
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:568
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) Running State Change CS_DESTROY
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:578
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State DESTROY
> 2013-08-09 22:28:10.110663 [DEBUG] mod_sofia.c:397
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 SOFIA DESTROY
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:99
> sofia/external/1100 at YYY.YYY.YYY.YYY:9060 Standard DESTROY
> 2013-08-09 22:28:10.110663 [DEBUG] switch_core_state_machine.c:578
> (sofia/external/1100 at YYY.YYY.YYY.YYY:9060) State DESTROY going to sleep
>
>
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Remote-Extension-Registered-but-can-t-make-receive-any-call-tp7593777.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From zhulizhong at live.com  Sun Aug 11 05:55:31 2013
From: zhulizhong at live.com (james.zhu)
Date: Sun, 11 Aug 2013 09:55:31 +0800
Subject: [Freeswitch-users] Wanpipe - Wanrouter help - freetdm and
	freeswitch
In-Reply-To: <1376051767.14626.YahooMailNeo@web160501.mail.bf1.yahoo.com>
References: <1376051767.14626.YahooMailNeo@web160501.mail.bf1.yahoo.com>
Message-ID: <BLU0-SMTP1209B5E5B36574684439CA8B55A0@phx.gbl>

hello:
you have to make sure the wanpipe file is there and with right data 
setting.

? 2013-8-9 20:36, Ravi ??:
> Hello Everyone !
>
> I am not sure if this error is relevant for this mailing list. My 
> apologies if this is not the forum. I am trying to configure the 
> Sangoma card, with the freetdm installation for Freeswitch. When I 
> complete the wanpipe installation, and try wanrouter status, and 
> wanrouter start I get the following:
>
> [root at bfree-server log]# wanrouter start
>
> Starting up device: wanpipe1
>
>
> wanconfig: WAN device wanpipe1 driver load failed !!
> : ioctl(wanpipe1,ROUTER_SETUP) failed:
> : 22 - Invalid argument
>
>
> Wanpipe driver did not load properly
> Please check /var/log/wanrouter and
> /var/log/messages for errors
>
> Configuring interfaces: w1g1 w1g1: unknown interface: No such device
>
> done.
> ___________________________________________________________________
>
> Can anyone help me understand why this is happening ? And how to 
> resolve it.
>
> Thanks.
> Ravi
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Best regards!
Connect with Sangoma in APAC
www.hiastar.com


-- 
Best regards! Connect with Sangoma in APAC www.hiastar.com
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From covici at ccs.covici.com  Sun Aug 11 07:22:50 2013
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sat, 10 Aug 2013 23:22:50 -0400
Subject: [Freeswitch-users] problem with menuexecapp
Message-ID: <10175.1376191370@ccs.covici.com>

Hi.  I am having a problem with the freeswitch ivr.  If someone types a
number of digits equal to the maximum length, and then types a #, the #
is saved and passed to wherever the transfer or whatever the menuexecapp
wants to go.  Now I have an app which responds to # by hanging up, so
this makes it difficult.  Is there a way that either the app could flush
the dtmf queue when it comes in, or could (and this would be much
better) the # be ignored when the maximum length is reached?    I
thought this was the way it was supposed to work, anyway.


-- 
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 a.afzali2003 at gmail.com  Sun Aug 11 08:49:06 2013
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sun, 11 Aug 2013 09:19:06 +0430
Subject: [Freeswitch-users] Problem with separating inline dialplan apps
In-Reply-To: <CAFiqYukbaptA3oNWeSzVRkJC-B7k2ob_nQ3yLsvndAwagjKmdA@mail.gmail.com>
References: <CABK0fC6iVUPx_QWNJSadO_Xixf_XB-hCo2Aw+6JA9O0sVK-fuQ@mail.gmail.com>
	<CAFiqYukbaptA3oNWeSzVRkJC-B7k2ob_nQ3yLsvndAwagjKmdA@mail.gmail.com>
Message-ID: <CABK0fC5n4dwwcGMTrJVzm4bRFbFFRemvYTt2FzaqA04uvyJ9nA@mail.gmail.com>

Hi Steven,

I've escaped it already but by just single ' \ ' (ha ha) as when I use in
CLI. It should be escaped with double '\\' to works.

Thank you so much,
Afshin


On Wed, Aug 7, 2013 at 6:12 PM, Steven Ayre <steveayre at gmail.com> wrote:

> You need to escape the comma - see example 4 on
> http://wiki.freeswitch.org/wiki/Loopback_endpoint
>
> As it stands now you are doing a forked dial via both the 'loopback'
> endpoint and the 'park' endpoint - except there is no such endpoint.
>
> -Steve
>
>
>
>
> On 7 August 2013 12:21, afshin afzali <a.afzali2003 at gmail.com> wrote:
>
>> Hi Guys,
>>
>> I use this command
>>   originate loopback/answer,park/context_1/inline &transfer(2001)
>> over a inbound event socket. Problem is that server takes just
>> loopback/answer part for new channel:
>> [NOTICE] switch_channel.c:1030 New Channel loopback/answer-a
>> [0eb8f17c-ff50-11e2-97ec-d950156f6aef]
>> and then try park for new channel!
>> [ERR] switch_core_session.c:496 Could not locate channel type park
>> When I submit same command over CLI (just added a \ before comma) it
>> works fine.
>>
>> appreciate all,
>> afshin
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 xyangni at gmail.com  Sun Aug 11 13:03:25 2013
From: xyangni at gmail.com (Jessica Wang)
Date: Sun, 11 Aug 2013 10:03:25 +0100
Subject: [Freeswitch-users] How to check who is attacking my fs server?
Message-ID: <CAO1DH1_vjzV4VxsD3TYP_+j0NiF=0HSB-Whgt8oNkBpU16ck8w@mail.gmail.com>

Dear all,

My FS server got attacked by someone. Unknown external user 107 is
frequently calling strange number on my FS server. From the log, I can not
find the real IP of this external user, the IP logged is my server's IP!
 How can check who is doing this and block any similar action?
 Part of the log is as below.

Thanks.



 2013-08-11 09:48:12.401286 [NOTICE] switch_channel.c:951 New Channel
sofia/external/107@(my fs server ip addr)
[c1499822-0262-11e3-af23-2d4b950c464a]
2013-08-11 09:48:12.421101 [INFO] mod_dialplan_xml.c:485 Processing 107
<107>->+970598610587 in context public
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From avi at avimarcus.net  Sun Aug 11 13:27:15 2013
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 11 Aug 2013 09:27:15 +0000
Subject: [Freeswitch-users] How to check who is attacking my fs server?
In-Reply-To: <CAO1DH1_vjzV4VxsD3TYP_+j0NiF=0HSB-Whgt8oNkBpU16ck8w@mail.gmail.com>
References: <CAO1DH1_vjzV4VxsD3TYP_+j0NiF=0HSB-Whgt8oNkBpU16ck8w@mail.gmail.com>
Message-ID: <000001406cb3784f-36838440-d3f0-4046-81a0-b8f0c56d0b9d-000000@email.amazonses.com>

CDRs say:
network_addr 85.195.107.50
For calls to 107. I guess I'm getting the same attack.
-Avi

On Sun, Aug 11, 2013 at 12:03 PM, Jessica Wang <xyangni at gmail.com> wrote:

> Dear all,
>
> My FS server got attacked by someone. Unknown external user 107 is
> frequently calling strange number on my FS server. From the log, I can not
> find the real IP of this external user, the IP logged is my server's IP!
>  How can check who is doing this and block any similar action?
>  Part of the log is as below.
>
> Thanks.
>
>
>
>  2013-08-11 09:48:12.401286 [NOTICE] switch_channel.c:951 New Channel
> sofia/external/107@(my fs server ip addr)
> [c1499822-0262-11e3-af23-2d4b950c464a]
> 2013-08-11 09:48:12.421101 [INFO] mod_dialplan_xml.c:485 Processing 107
> <107>->+970598610587 in context public
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 rajat.toshniwal at tekmindz.com  Sun Aug 11 14:46:24 2013
From: rajat.toshniwal at tekmindz.com (Rajat toshniwal)
Date: Sun, 11 Aug 2013 16:16:24 +0530
Subject: [Freeswitch-users] Freeswitch with Digium T316 timed out,
 T316 timed out, 
In-Reply-To: <52048B5D.8000207@tekmindz.com>
References: <5203774A.7040906@tekmindz.com> <52039DD4.9070109@tekmindz.com>
	<52048B5D.8000207@tekmindz.com>
Message-ID: <52076B80.8090103@tekmindz.com>

It seems like a bug in ftmod_libpri, as it is not able to handle the 
incoming Acknowledge packets. I have tried it with freeswitch 1.4.0 also 
it is still there. Digium TE121 + freeswitch 1.4.0 + Dahdi 2.7.0 + 
libpri 1.4.14 is not working. Channel State is still down where as 
Signaling is up.



On Friday 09 August 2013 11:55 AM, Rajat toshniwal wrote:
> Any Clues ??
>
> On Thursday 08 August 2013 07:02 PM, Rajat toshniwal wrote:
>> Forget to mention that I have already tested below mentioned 
>> configuration
>>
>>    1. Go to FS source directory and change dir to this
>>       path libs/freetdm/src/ftmod/ftmod_libpri
>>    2. change #define T316_TIMEOUT_MS_DEFAULT to 10000
>>    3. #define T316_TIMEOUT_MS_MAX to 300000
>>    4. #define T316_ATTEMPT_LIMIT_DEFAULT to 30 (max. channels in PRI
>>       lines)
>>
>> It didn't work
>>
>> Also we are using the latest version of Freeswitch 1.2.12 and 
>> libpri1.4.14
>>
>> Regards
>> Rajat Toshniwal
>>
>>
>>
>> On Thursday 08 August 2013 04:17 PM, Rajat toshniwal wrote:
>>> Hi
>>>
>>> I am trying to deploy freeswitch with Digium TE121 card for my office
>>> setup, but it is continuously showing Signaling is up and channels are
>>> down except D channel.
>>> Our Architecture is like
>>> We have freeswitch installed with libpri1.4 and Dahdi.
>>> I am from India and here we are having E1 trunk.
>>>
>>> Dahdi Configuration is
>>>
>>> cat system.conf
>>> # Autogenerated by /usr/sbin/dahdi_genconf on Wed Aug  7 19:39:07 2013
>>> # If you edit this file and execute /usr/sbin/dahdi_genconf again,
>>> # your manual changes will be LOST.
>>> # Dahdi Configuration File
>>> #
>>> # This file is parsed by the Dahdi Configurator, dahdi_cfg
>>> #
>>> # Span 1: WCT1/0 "Wildcard TE121 Card 0" (MASTER)
>>> span=1,1,0,ccs,hdb3,crc4
>>> # termtype: te
>>> bchan=1-15,17-31
>>> dchan=16
>>> echocanceller=mg2,1-15,17-31
>>>
>>> # Global data
>>>
>>> loadzone    = uk
>>> defaultzone    = uk
>>>
>>>
>>>
>>> cat modules
>>> # Autogenerated by /usr/sbin/dahdi_genconf (Dahdi::Config::Gen::Modules)
>>> on Wed Aug  7 19:37:48 2013
>>> # If you edit this file and execute /usr/sbin/dahdi_genconf again,
>>> # your manual changes will be LOST.
>>> wcte12xp
>>> # Xorcom Astribank Devices
>>> xpp_usb
>>>
>>>
>>> dahdi_hardware
>>> pci:0000:02:08.0     wcte12xp+    d161:8000 Wildcard TE121
>>>
>>> dahdi_scan
>>> [1]
>>> active=yes
>>> alarms=OK
>>> description=Wildcard TE121 Card 0
>>> name=WCT1/0
>>> manufacturer=Digium
>>> devicetype=Wildcard TE121 (VPMOCT032)
>>> location=PCI Bus 02 Slot 09
>>> basechan=1
>>> totchans=31
>>> irq=0
>>> type=digital-E1
>>> syncsrc=1
>>> lbo=0 db (CSU)/0-133 feet (DSX-1)
>>> coding_opts=AMI,HDB3
>>> framing_opts=CCS,CRC4
>>> coding=HDB3
>>> framing=CCS/CRC4
>>>
>>>
>>> Card is properly installed and recognized by Dahdi
>>>
>>> Freetdm is compiled with libpri and configuration is like
>>> cat /usr/local/freeswitch/conf/freetdm.conf
>>> [general]
>>> cpu_monitor =>  yes
>>> cpu_monitoring_interval =>  2000 ; monitor usage every 2 seconds
>>> cpu_set_alarm_threshold =>  90 ; whenever 90% of global CPU usage is
>>> reached, trigger the alarm.
>>> cpu_reset_alarm_threshold =>  80 ; when the CPU usage decreases at 80%,
>>> clear the alarm.
>>> cpu_alarm_action =>  reject,warn ; Start rejecting calls when the CPU
>>> alarm is triggered and also print warnings.
>>>
>>> [span zt myDAHDISpan]
>>> trunk_type =>  E1
>>> group =>  g1
>>> b-channel =>  1-15
>>> d-channel =>  16
>>> b-channel =>  17-31
>>>
>>>
>>> cat /usr/local/freeswitch/conf/autoload_configs/freetdm.conf.xml
>>> <configuration name="freetdm.conf" description="FreeTDM Configuration">
>>> <settings>
>>> <param name="debug" value="1"/>
>>> </settings>
>>> <libpri_spans>
>>> <span name="myDAHDISpan">
>>> <!-- Log Levels: none, alert, crit, err, warning, notice, info, debug -->
>>> <param name="switch" value="euroisdn"/>
>>> <param name="node" value="cpe"/>
>>> <param name="dialect" value="q931"/>
>>> <param name="debug" value="all"/>
>>> <param name="dialplan" value="XML"/>
>>> <param name="context" value="public"/>
>>> <param name="l1" value="alaw"/>
>>> </span>
>>> </libpri_spans>
>>> </configuration>
>>>
>>>
>>> Freeswitch logs are showing
>>>
>>> 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c10][1:10]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c11][1:11]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c12][1:12]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.613848 [WARNING] ftmod_libpri.c:1975 [s1c13][1:13]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c14][1:14]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c15][1:15]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c17][1:17]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c18][1:18]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c19][1:19]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c20][1:20]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c21][1:21]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c22][1:22]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c23][1:23]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c24][1:24]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c25][1:25]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c26][1:26]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c27][1:27]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c28][1:28]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c29][1:29]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c30][1:30]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:18.633847 [WARNING] ftmod_libpri.c:1975 [s1c31][1:31]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:23.673847 [WARNING] ftdm_io.c:3022 [s1c5][1:5] Channel
>>> not opened, proceeding anyway
>>> 2013-08-08 15:38:48.613848 [WARNING] ftmod_libpri.c:1975 [s1c6][1:6] --
>>> T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:48.613848 [WARNING] ftmod_libpri.c:1975 [s1c7][1:7] --
>>> T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c8][1:8] --
>>> T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c9][1:9] --
>>> T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c10][1:10]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c11][1:11]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c12][1:12]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c13][1:13]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c14][1:14]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c15][1:15]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:38:48.633850 [WARNING] ftmod_libpri.c:1975 [s1c17][1:17]
>>> -- T316 timed out, resending RESTART request
>>>
>>>
>>>
>>> Libpri logs are showing
>>>
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> DL-DATA request
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> Protocol Discriminator: Q.931 (8)  len=13
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> Message Type: RESTART (70)
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>> TEI=0 Transmitting N(S)=4, window is open V(A)=2 K=7
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> TEI: 0 State 7(Multi-frame established)
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> V(A)=2, V(S)=4, V(R)=6
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> T200_id=8192, N200=3, T203_id=0
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>  [
>>> 00 01 08 0c 08 02 00 00 46 18 03 a9 83 8d 79 01 80 ]
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> Informational frame:
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> SAPI: 00  C/R: 0 EA: 0
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> TEI: 000        EA: 1
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> N(S): 004   0: 0
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> N(R): 006   P: 0
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>  13
>>> bytes of data
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> Protocol Discriminator: Q.931 (8)  len=13
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> Message Type: RESTART (70)
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> [18 03 a9 83 8d]
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)  Spare: 0
>>> Exclusive  Dchan: 0
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>>   >                        ChanSel: As indicated in following octets
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>>   >                        Ext: 1  Coding: 0  Number Specified  Channel
>>> Type: 3
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>>   >                        Ext: 1  Channel: 13 Type: CPE]
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> [79 01 80]
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> Restart Indicator (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated
>>> Channel (0) ]
>>> 2013-08-08 15:41:18.633844 [DEBUG] lpwrap_pri.c:199 [s1c16][1:16] --
>>> Starting timer 0x2678478 with timeout 30000 ms
>>> 2013-08-08 15:41:18.633844 [WARNING] ftmod_libpri.c:1975 [s1c14][1:14]
>>> -- T316 timed out, resending RESTART request
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> DL-DATA request
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> Protocol Discriminator: Q.931 (8)  len=13
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> Message Type: RESTART (70)
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>> TEI=0 Transmitting N(S)=5, window is open V(A)=2 K=7
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> TEI: 0 State 7(Multi-frame established)
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> V(A)=2, V(S)=5, V(R)=6
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> K=7, RC=0, l3_initiated=0, reject_except=0, ack_pend=0
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> T200_id=8192, N200=3, T203_id=0
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>  [
>>> 00 01 0a 0c 08 02 00 00 46 18 03 a9 83 8e 79 01 80 ]
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> Informational frame:
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> SAPI: 00  C/R: 0 EA: 0
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> TEI: 000        EA: 1
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> N(S): 005   0: 0
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> N(R): 006   P: 0
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>  13
>>> bytes of data
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> Protocol Discriminator: Q.931 (8)  len=13
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> TEI=0 Call Ref: len= 2 (reference 0/0x0) (Sent from originator)
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> Message Type: RESTART (70)
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> [18 03 a9 83 8e]
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> Channel ID (len= 5) [ Ext: 1  IntID: Implicit  Other(PRI)  Spare: 0
>>> Exclusive  Dchan: 0
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>>   >                        ChanSel: As indicated in following octets
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>>   >                        Ext: 1  Coding: 0  Number Specified  Channel
>>> Type: 3
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]
>>>   >                        Ext: 1  Channel: 14 Type: CPE]
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> [79 01 80]
>>> 2013-08-08 15:41:18.633844 [DEBUG] ftmod_libpri.c:134 [s1c16][1:16]>
>>> Restart Indicator (len= 3) [ Ext: 1  Spare: 0  Resetting Indicated
>>> Channel (0) ]
>>> 2013-08-08 15:41:18.633844 [DEBUG] lpwrap_pri.c:199 [s1c16][1:16] --
>>> Starting timer 0x26784d8 with timeout 30000 ms
>>> 2013-08-08 15:41:18.633844 [WARNING] ftmod_libpri.c:1975 [s1c15][1:15]
>>> -- T316 timed out, resending RESTART request
>>>
>>>
>>> ftdm dump 1 n (where n is from 1 to 15 and  17 to 31) is showing state
>>> as down
>>>
>>> freeswitch at dst1>  ftdm dump 1 1
>>>
>>> span_id: 1
>>> chan_id: n
>>> physical_span_id: 1
>>> physical_chan_id: n
>>> physical_status: ok
>>> physical_status_red: 0
>>> physical_status_yellow: 0
>>> physical_status_rai: 0
>>> physical_status_blue: 0
>>> physical_status_ais: 0
>>> physical_status_general: 0
>>> signaling_status: UP
>>> type: B
>>> state: DOWN
>>> last_state: RESTART
>>> txgain: 0.00
>>> rxgain: 0.00
>>> cid_date:
>>> cid_name:
>>> cid_num:
>>> ani:
>>> aniII:
>>> dnis:
>>> rdnis:
>>> cause: NONE
>>> session: (none)
>>>
>>> I have tried many configurations, but I am not able to fix that issue ,
>>> If I put my Pri line into panasonic PBX, it is working properly.
>>>
>>> Kindly help me in solving this issue.
>>>
>>> Regards
>>> Rajat Toshniwal
>>> ----------------------------------------------------------------------------------
>>> Disclaimer: The information contained in this communication is confidential, private, proprietary, or otherwise privileged and is intended only for the use of the addressee. Unauthorized use, disclosure, distribution or copying is strictly prohibited and may be unlawful. If you have received this communication in error, please delete this message and notify the sender immediately - Samin TekMindz India Pvt. Ltd.
>>> ----------------------------------------------------------------------------------
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>>
>>
>> ------------------------------------------------------------------------
>> Disclaimer: The information contained in this communication is 
>> confidential, private, proprietary, or otherwise privileged and is 
>> intended only for the use of the addressee.Unauthorized use, 
>> disclosure, distribution or copying is strictly prohibited and may be 
>> unlawful. If you have received this communication in error, please 
>> delete this message and notify the sender immediately - Samin 
>> TekMindz India Pvt.Ltd.
>> ------------------------------------------------------------------------
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
>
>
> ------------------------------------------------------------------------
> Disclaimer: The information contained in this communication is 
> confidential, private, proprietary, or otherwise privileged and is 
> intended only for the use of the addressee.Unauthorized use, 
> disclosure, distribution or copying is strictly prohibited and may be 
> unlawful. If you have received this communication in error, please 
> delete this message and notify the sender immediately - Samin TekMindz 
> India Pvt.Ltd.
> ------------------------------------------------------------------------
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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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Disclaimer: The information contained in this communication is confidential, private, proprietary, or otherwise privileged and is intended only for the use of the addressee. Unauthorized use, disclosure, distribution or copying is strictly prohibited and may be unlawful. If you have received this communication in error, please delete this message and notify the sender immediately - Samin TekMindz India Pvt. Ltd.
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From covici at ccs.covici.com  Sun Aug 11 15:03:06 2013
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Sun, 11 Aug 2013 07:03:06 -0400
Subject: [Freeswitch-users] How to check who is attacking my fs server?
In-Reply-To: <000001406cb3784f-36838440-d3f0-4046-81a0-b8f0c56d0b9d-000000@email.amazonses.com>
References: <CAO1DH1_vjzV4VxsD3TYP_+j0NiF=0HSB-Whgt8oNkBpU16ck8w@mail.gmail.com>
	<000001406cb3784f-36838440-d3f0-4046-81a0-b8f0c56d0b9d-000000@email.amazonses.com>
Message-ID: <6481.1376218986@ccs.covici.com>

Me too, but fail2ban kills it nicely.

Avi Marcus <avi at avimarcus.net> wrote:

> CDRs say:
> network_addr 85.195.107.50
> For calls to 107. I guess I'm getting the same attack.
> -Avi
> 
> On Sun, Aug 11, 2013 at 12:03 PM, Jessica Wang <xyangni at gmail.com> wrote:
> 
> > Dear all,
> >
> > My FS server got attacked by someone. Unknown external user 107 is
> > frequently calling strange number on my FS server. From the log, I can not
> > find the real IP of this external user, the IP logged is my server's IP!
> >  How can check who is doing this and block any similar action?
> >  Part of the log is as below.
> >
> > Thanks.
> >
> >
> >
> >  2013-08-11 09:48:12.401286 [NOTICE] switch_channel.c:951 New Channel
> > sofia/external/107@(my fs server ip addr)
> > [c1499822-0262-11e3-af23-2d4b950c464a]
> > 2013-08-11 09:48:12.421101 [INFO] mod_dialplan_xml.c:485 Processing 107
> > <107>->+970598610587 in context public
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
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 jleung at v10networks.ca  Sun Aug 11 20:29:52 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Sun, 11 Aug 2013 09:29:52 -0700
Subject: [Freeswitch-users] How to check who is attacking my fs server?
In-Reply-To: <CAO1DH1_vjzV4VxsD3TYP_+j0NiF=0HSB-Whgt8oNkBpU16ck8w@mail.gmail.com>
References: <CAO1DH1_vjzV4VxsD3TYP_+j0NiF=0HSB-Whgt8oNkBpU16ck8w@mail.gmail.com>
Message-ID: <000301ce96b0$02b219c0$08164d40$@v10networks.ca>

You can use xml_cdr to log all calls going through your box. Xml_cdr tends
to capture almost everything including where the UDP packet came from IP
header wise.

 

You can also try Brain West's blacklist here:


http://daffy.bkw.org/blacklist.txt

 

That black list blocks IPs that are known to be attacking boxes and so far,
it's pretty freaking effective against them ;)

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jessica
Wang
Sent: Sunday, August 11, 2013 2:03 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] How to check who is attacking my fs server?

 

Dear all,

 

My FS server got attacked by someone. Unknown external user 107 is
frequently calling strange number on my FS server. From the log, I can not
find the real IP of this external user, the IP logged is my server's IP!

 How can check who is doing this and block any similar action?

 Part of the log is as below. 

 

Thanks. 

 

 

 

 2013-08-11 09:48:12.401286 [NOTICE] switch_channel.c:951 New Channel
sofia/external/107@(my fs server ip addr)
[c1499822-0262-11e3-af23-2d4b950c464a]

2013-08-11 09:48:12.421101 [INFO] mod_dialplan_xml.c:485 Processing 107
<107>->+970598610587 in context public

 

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From andretodd at verizon.net  Sun Aug 11 20:35:04 2013
From: andretodd at verizon.net (Andre)
Date: Sun, 11 Aug 2013 12:35:04 -0400
Subject: [Freeswitch-users] Tech Prefix with Gateway
Message-ID: <10e101ce96b0$bc29f580$347de080$@verizon.net>

Hi, I'm trying to pass a Tech Prefix while using a gateway. When I add the
prefix in front of the phone number I get a routing error back. 

sofia/gateway/myGateway/prefix + Phone Number

sofia/gateway/myGateway/211713015551212

So, what is the proper way to pass the Tech Prefix?
Thanks
Andre



From avi at avimarcus.net  Sun Aug 11 20:47:32 2013
From: avi at avimarcus.net (Avi Marcus)
Date: Sun, 11 Aug 2013 16:47:32 +0000
Subject: [Freeswitch-users] Tech Prefix with Gateway
In-Reply-To: <10e101ce96b0$bc29f580$347de080$@verizon.net>
References: <10e101ce96b0$bc29f580$347de080$@verizon.net>
Message-ID: <000001406e468968-e1f11e73-c81f-4ddd-989e-f7d4200ae3dc-000000@email.amazonses.com>

You're giving an example of calling with prefix 2117 to USA number
13015551212?
That looks right. I'd follow up with the carrier...

-Avi

On Sun, Aug 11, 2013 at 7:35 PM, Andre <andretodd at verizon.net> wrote:

> Hi, I'm trying to pass a Tech Prefix while using a gateway. When I add the
> prefix in front of the phone number I get a routing error back.
>
> sofia/gateway/myGateway/prefix + Phone Number
>
> sofia/gateway/myGateway/211713015551212
>
> So, what is the proper way to pass the Tech Prefix?
> Thanks
> Andre
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ga at steadfasttelecom.com  Sun Aug 11 20:57:00 2013
From: ga at steadfasttelecom.com (Gilad Abada)
Date: Sun, 11 Aug 2013 12:57:00 -0400
Subject: [Freeswitch-users] Tech Prefix with Gateway
In-Reply-To: <000001406e468968-e1f11e73-c81f-4ddd-989e-f7d4200ae3dc-000000@email.amazonses.com>
References: <10e101ce96b0$bc29f580$347de080$@verizon.net>
	<000001406e468968-e1f11e73-c81f-4ddd-989e-f7d4200ae3dc-000000@email.amazonses.com>
Message-ID: <-3125850653500742935@unknownmsgid>

You may need a * after your tech prefix.

Sent from my mobile device.

On Aug 11, 2013, at 12:54, Avi Marcus <avi at avimarcus.net> wrote:

You're giving an example of calling with prefix 2117 to USA number
13015551212?
That looks right. I'd follow up with the carrier...

-Avi

On Sun, Aug 11, 2013 at 7:35 PM, Andre <andretodd at verizon.net> wrote:

> Hi, I'm trying to pass a Tech Prefix while using a gateway. When I add the
> prefix in front of the phone number I get a routing error back.
>
> sofia/gateway/myGateway/prefix + Phone Number
>
> sofia/gateway/myGateway/211713015551212
>
> So, what is the proper way to pass the Tech Prefix?
> Thanks
> Andre
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 andretodd at verizon.net  Sun Aug 11 20:57:37 2013
From: andretodd at verizon.net (Andre)
Date: Sun, 11 Aug 2013 12:57:37 -0400
Subject: [Freeswitch-users] Tech Prefix with Gateway
In-Reply-To: <000001406e468968-e1f11e73-c81f-4ddd-989e-f7d4200ae3dc-000000@email.amazonses.com>
References: <10e101ce96b0$bc29f580$347de080$@verizon.net>
	<000001406e468968-e1f11e73-c81f-4ddd-989e-f7d4200ae3dc-000000@email.amazonses.com>
Message-ID: <10ee01ce96b3$e3130530$a9390f90$@verizon.net>

Yes, odd after I closed everything and restarted it worked as the example
below. Hmm..

Thanks for confirming that I had it right.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi
Marcus
Sent: Sunday, August 11, 2013 12:48 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Tech Prefix with Gateway

 

You're giving an example of calling with prefix 2117 to USA number
13015551212?
That looks right. I'd follow up with the carrier...

-Avi

 

On Sun, Aug 11, 2013 at 7:35 PM, Andre <andretodd at verizon.net
<mailto:andretodd at verizon.net> > wrote:

Hi, I'm trying to pass a Tech Prefix while using a gateway. When I add the
prefix in front of the phone number I get a routing error back.

sofia/gateway/myGateway/prefix + Phone Number

sofia/gateway/myGateway/211713015551212

So, what is the proper way to pass the Tech Prefix?
Thanks
Andre


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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

 

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From khorsmann at gmail.com  Sun Aug 11 20:58:32 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Sun, 11 Aug 2013 18:58:32 +0200
Subject: [Freeswitch-users] WebRTC sipml5 client hangs up after 120
 seconds on bridged calls to sip phone
In-Reply-To: <CAFArqsY-sBqFYu1xAw_K48tTrNHKxxAYy7ZYj3pLF_wm7p+WUA@mail.gmail.com>
References: <CAFArqsY-sBqFYu1xAw_K48tTrNHKxxAYy7ZYj3pLF_wm7p+WUA@mail.gmail.com>
Message-ID: <CAFArqsYggJoB2xcuVkoDxC7K7SQXuFGMUGjei+Q3YonDqWgAkw@mail.gmail.com>

Hello List,

answering myself to that problem.

With deactivated session-timers *) it seems to work. Iam not sure if i
broke then something other.
Maybe the snom firmware is buggy. I will try an upgrade to latest and try
another hardphone too.

*) in the sip-profile:
<param name="enable-timer" value="false"/>



2013/8/10 Karsten Horsmann <khorsmann at gmail.com>

> Hello,
>
> i play around with freeswitch master on centos 6.4 and webrtc to
> sip-phones.
> if i call from the browser (chrome) to a snom, the call ends after 120
> seconds.
> I see that the sipml5 client is sending then a bye message.
>
> The other direction works fine. From snom to the webrtc phone, no hangup
> after 120 seconds.
> And from webrtc to the moh extension works fine too.
>
> Any special settings to resolve that?
>
> I made an sofia trace, both phones are on the same sip-profile.
> http://pastebin.freeswitch.org/21291
>
>
> my git version is d1268e81036ce8ce00de8ee22f387cdbf43a7203
>
> --
> Kind Regards
> Mit freundlichen Gr??en
> *Karsten Horsmann*
>



-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From drk at drkngs.net  Mon Aug 12 09:37:55 2013
From: drk at drkngs.net (Dave R. Kompel)
Date: Sun, 11 Aug 2013 22:37:55 -0700
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <52049391.4080105@quentustech.com>
Message-ID: <20130812053755.96624167@mail.tritonwest.net>

William,  
   
The business case is for those of us that use FreeSWITCH as a main switch at a CLEC, where we have our own prefix assignments. In order to do SMS on those numbers we need to interconnect with a SMS backbone provider. SMPP is you're only option for that...  
   
--Dave
      _____  

  From: William King [mailto:william.king at quentustech.com]
To: freeswitch-users at lists.freeswitch.org
Sent: Fri, 09 Aug 2013 00:00:33 -0700
Subject: Re: [Freeswitch-users] Potential FreeSWITCH support for SMPP

>From what I've seen, retail level SMS carriers(not sms aggregators)
usually offer a mix of HTTP, XMPP, SIP SIMPLE, or SMPP interfaces. I've
seen the higher volume SMS connections prefer SMPP or HTTP.

I'm exploring to see if there is a business case for FreeSWITCH support
for SMPP(either direct support, or interface support through an
independent application) that is not already covered by the current FS
feature set.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main: (877) 211-9337
Office: (206) 388-4772
Cell: (253) 686-5518
william.king at quentustech.com

On 08/08/2013 04:23 AM, Raymond Chandler wrote:
> So, to the best of my knowledge, SMPP is strictly for SMS so you can
> route SMS to your clients via FS to SIMPLE / dingaling / etc clients
> 
> -Ray
> 
> On Aug 7, 2013 11:05 PM, "Victor Chukalovskiy"
> <victor.chukalovskiy at gmail.com <mailto:victor.chukalovskiy at gmail.com>>
> wrote:
> 
> So am I as well as another ClueCon attendee I spoke to.
> 
> What is the usage scenario you are looking at?
> 
> On 13-08-07 06:46 PM, Dmitry Lysenko wrote:
>> Yes. I am interested.
>>
>> Dmitry.
>>
>>
>> 2013/8/8 William King <william.king at quentustech.com
>> <mailto:william.king at quentustech.com>>
>>
>> Is anyone on this list interested in FreeSWITCH support for
>> SMPP for SMS
>> messages?
>>
>> For more information about the SMPP protocol checkout:
>> http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
>>
>> If so feel free to contact me on or off this list.
>> --
>> William King
>> Senior Engineer
>> Quentus Technologies, INC
>> 1037 NE 65th St Suite 273
>> Seattle, WA 98115
>> Main: (877) 211-9337 <tel:%28877%29%20211-9337>
>> Office: (206) 388-4772 <tel:%28206%29%20388-4772>
>> Cell: (253) 686-5518 <tel:%28253%29%20686-5518>
>> william.king at quentustech.com <mailto:william.king at quentustech.com>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 julien.terrasson at gmail.com  Mon Aug 12 11:01:37 2013
From: julien.terrasson at gmail.com (julien terrasson)
Date: Mon, 12 Aug 2013 09:01:37 +0200
Subject: [Freeswitch-users] How to create a single outbound ESL socket at
	freeswitch startup.
Message-ID: <CAFi=dz9iXE6ow0pHg4tw+d9cP12we5Aadh7nv6TLgqCPntzhJw@mail.gmail.com>

Hi,

I would like to create a single outbound ESL socket that would establish
the connection with the ESL server at freeswitch startup.
This connexion would remains up and would used to transfer events from
calls going through this server.
I managed to do this with an inbound ESL but i would prefer doing it with
an outbound ESL (it is important for me that freeswitch controls the
establishement of the connection).
Does anybody know if that's possible to start a common ESL connection (that
would push event from all call) at FS startup ? If yes, where should it be
provisionned (from the dialplan? in a startup script ?)

Hope somebody can help..

Regards,

Julien
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From treitinger at as-infodienste.de  Mon Aug 12 12:33:36 2013
From: treitinger at as-infodienste.de (Melanie Treitinger)
Date: Mon, 12 Aug 2013 10:33:36 +0200
Subject: [Freeswitch-users]  uuid_displace: change loudness level of audio
References: 52010853.2020603@as-infodienste.de
Message-ID: <52089DE0.2000203@as-infodienste.de>

Hi Michael,

I think I'll try sox, thanks.


Melanie
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From pablo.platt at gmail.com  Mon Aug 12 11:40:17 2013
From: pablo.platt at gmail.com (pablo platt)
Date: Mon, 12 Aug 2013 10:40:17 +0300
Subject: [Freeswitch-users] Conference for Flash and WebRTC users
Message-ID: <CANdLC8VAzwRcFBHog5RT_M8=AZqTdZNdeNAcFDAzPU0AFDgHkQ@mail.gmail.com>

Hi,

I'm currently using a RTMP server for audio conference.
There are no performance or latency issues because there is no encryption,
transcoding or mixing involved. A 1GB VPS server can handle more than 100
users without a problem.

I'm trying to evaluate FreeSWITCH so I'll be able to support both Flash and
WebRTC users in a conference.

My use case is two types of conferences:
- A conference with up to 5 participants all speaking.
- A conference with 20-30 participants with 2-3 speakers.

What part require more CPU?
- Encryption (DTLS-SRTP)
- Transcoding
- Mixing

Does FS mix a channel separately for each participant or can it reuse the
same mix?
For example, if I have 1 speaker and 20 listeners, 10 with speex and 10
with opus, does FS produce 2 streams or 20?

Does FS exclude silent participants from the mix to save CPU?

Thanks
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From itsme.kunnu at gmail.com  Mon Aug 12 12:51:45 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Mon, 12 Aug 2013 14:21:45 +0530
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
Message-ID: <CAMvhjJtBONsXnrhRoDdoMTUPhHb1NjVRSJOsdbt5u+0tjqaBug@mail.gmail.com>

I reinstalled freeswitch but still i am getting the same error :
fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
mod_socket is also loaded...i have also flushed the firewall rules...pls
help
On Aug 6, 2013 2:30 AM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:

> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i
> have installed freeswitch) it gives me the following error :
> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>
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From mike at jerris.com  Mon Aug 12 18:14:45 2013
From: mike at jerris.com (Michael Jerris)
Date: Mon, 12 Aug 2013 10:14:45 -0400
Subject: [Freeswitch-users] garbled audio with G726-32,
	other codecs are fine
In-Reply-To: <51FDEC5B.2010307@c3i.bg>
References: <51FD0E8B.3060702@c3i.bg>
	<CAHgk4BVNA+NOg5ChyYxSddDxhMxgSmWVe1tQ7NiCPRKo6M5BCw@mail.gmail.com>
	<003901ce90b9$749f0090$5ddd01b0$@v10networks.ca>
	<51FDEC5B.2010307@c3i.bg>
Message-ID: <065260E6-8DAB-4273-8C5A-0D0A48BD175B@jerris.com>

There is always bit packing, but there are 2 different ways to do the bit packing.  A lot of devices get it wrong so its worth looking at that.

Mike

On Aug 4, 2013, at 1:53 AM, Ivan Mitev <imitev at c3i.bg> wrote:

> Thanks for the suggestion but I'm testing with G726-32, not AAL2-G726-32 
> ; so bitpacking shouldn't be used. By the way when I tested with AAL2 to 
> the linphone client I only got cracks and whitenoise, I've forgot to 
> mention that in my post.
> 
> That said I've tried to uncomment and set <param name="bitpacking" 
> value="none"/> in internal.xml ("none" is a wild guess - I couldn't find 
> any doc on values accepted by this parameter), but that doesn't help.
> 
> Speex: the ATAs don't support it. And being stubborn I'd like to 
> understand what's wrong with G726 :)
> 
> 
> On 08/04/2013 05:22 AM, Jeff Leung wrote:
>> 
>> You can turn off G726 AAC bit-packing in spandsp.conf.xml.
>> 
>> By the way, there are other codecs out there you can try. SPEEX comes 
>> to mind if all your endpoints don?t deal with the PSTN.
>> 
>> *From:*freeswitch-users-bounces at lists.freeswitch.org 
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
>> *Brian Foster
>> *Sent:* Saturday, August 3, 2013 2:37 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] garbled audio with G726-32, other 
>> codecs are fine
>> 
>> AAC bitpacking by any chance? I thought I had a similar issue, 
>> happened so long ago I cant remember what I did.
>> 
>> Thank you,
>> 
>> Brian Foster
>> Project Manager/Owner's Rep.
>> Davri Investments, Inc.
>> O: 317-787-2686 x2102
>> M: 317-600-9753
>> E: bdfoster at davri.com <mailto:bdfoster at davri.com>
>> Indianapolis, Indiana
>> 
>> Sent from a mobile device.
>> 
>> On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev at c3i.bg 
>> <mailto:imitev at c3i.bg>> wrote:
>> 
>> Hello
>> 
>> I'm migrating an office setup from asterisk to FS and in the process I
>> was considering using G726-32 for some bandwidth starved remote
>> endpoints. However I only get metallic/garbled audio with that codec
>> even when simply playing moh to the endpoint, while other codecs work
>> fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds
>> marginally better but still garbled and really worse than G711.
>> 
>> The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest
>> (centos6 64bit host). But please don't shoot ! :) - I know about virtual
>> environment limitations but for these tests the host is only lightly
>> loaded, there aren't any calls to the FS instance except my tests, and
>> the fact that it works with other codecs makes me think that
>> virtualization is not the issue here. I may be wrong though.
>> 
>> Is there any guide for debugging that kind of problem before reverting
>> to a fresh install on bare-metal with the latest HEAD ? Until now I've
>> tried:
>> 
>> - improving timers ; but the default soft timer (which I guess uses
>> timerd) works best. The time interval between sent packets on a tcpdump
>> trace looks identical to the output of "timer_test", so that doesn't
>> seem to be a network/jitter problem. And there's no problem with other
>> codecs, but maybe G726-XX is specific. For info the guest's clocksource
>> is kvm_clock, while the host uses tsc.
>> 
>> - using different endpoints: the production ones are Linksys PAP2
>> ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the
>> same thing happens with linphone on a fedora 19 laptop.
>> 
>> A call with rtp media going through FS without transcoding - G726-32 to
>> G726-32 - works perfectly (I can't hear the difference with G711). The
>> problem is only when there's transcoding to G726 (from wav for moh, or
>> from any other codec when bridging). I've looked at the wiki, posts,
>> changelogs, jira, ..., but am a bit at a loss now.
>> 
>> Any pointers ?
>> 
>> Except that little problem, FS rocks, and I'm happy I can finally ditch
>> asterisk. Kudos to the core devs and contributors.
>> 
>> Ivan
>> 



From steveayre at gmail.com  Mon Aug 12 18:44:38 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 12 Aug 2013 15:44:38 +0100
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAMvhjJtBONsXnrhRoDdoMTUPhHb1NjVRSJOsdbt5u+0tjqaBug@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<CAMvhjJtBONsXnrhRoDdoMTUPhHb1NjVRSJOsdbt5u+0tjqaBug@mail.gmail.com>
Message-ID: <CAFiqYuneYGGujBH-a_hanzEQ72Rdiks5kfwntwz2QGp1YC9A7A@mail.gmail.com>

Have you checked the logfile and netstat?


On 12 August 2013 09:51, Ashish Mishra <itsme.kunnu at gmail.com> wrote:

> I reinstalled freeswitch but still i am getting the same error :
> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
> mod_socket is also loaded...i have also flushed the firewall rules...pls
> help
> On Aug 6, 2013 2:30 AM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>
>> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i
>> have installed freeswitch) it gives me the following error :
>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mike at jerris.com  Mon Aug 12 18:45:46 2013
From: mike at jerris.com (Michael Jerris)
Date: Mon, 12 Aug 2013 10:45:46 -0400
Subject: [Freeswitch-users] Issue with JSSIP + Freeswitch
In-Reply-To: <CAA7Zpyo5rGpo5jdXjK35-kuH52wJ_10k9KFn=OudDpHvBNqbvw@mail.gmail.com>
References: <CAA7Zpyo5rGpo5jdXjK35-kuH52wJ_10k9KFn=OudDpHvBNqbvw@mail.gmail.com>
Message-ID: <830B6C4C-0B66-44C6-8649-D9DDFFDEFD02@jerris.com>

We only use non rfc-1918 ip's by default.  If you want to use 1918 ip's you need to tweak acls

Mike


On Aug 5, 2013, at 8:24 AM, Shahrzad A. <shahrzad.aziminia at gmail.com> wrote:

> Hi  everyone
> 
> I'm having latest version of Freeswitch installed on Ubuntu 12.04.2 LTS  with the latest version of Openssl ('OpenSSL 1.0.1e 11 Feb 2013')
> I'm using the default configuration and just uncommentated the ' <param name="ws-binding"  value=":5066"/> ' in internal.xml in order to have the support for webrtc. 
> As the client I'm having JSSIP, the latest version with the adjustment to have (DtlsSrtpKeyAgreement:true). Both my clients and server located in the same network without having any firewalls therefore I'm not using any STUN server. I can make call but there is no sound! and after setting debig_level 10 and dial from extension 1003 to 5000 I'm having the following Alert in console and no sound:
> 
> [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!)
> 
> its the same case if I dial from extension 1002 to 1005! (Both are JSSIP clients) 
> If I call from extension 1000 which is a SIP client set on my iPhone and call 1003 I'm having the following alerts but voice on both JSSIP client and SIP client:
> 
> 2013-08-05 14:17:05.222446 [ALERT] switch_rtp.c:4563 sofia/internal/1000 at 10.0.14.16:5060 timer while HOT
> 2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/1000 at 10.0.14.16:5060 Hot Hit 1
> 
> And from extension 1003 (JSSIP) to extension 1000 ( SIP on Iphone) No voice and I'm getting the following alerts:
> 
> 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4546 sofia/internal/sip:1000 at 10.0.14.182:5065 Hot Hit 4
> 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/sip:1000 at 10.0.14.182:5065 timer while HOT
> 2013-08-05 14:19:57.132441 [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!)
> 
> If I use bypass media or proxy media I will have voice on both JSSIP clients but cant ring any SIP clients and again no voice if I call 5000!
> 
> Wonder to know if there is any special setting required on FreeSwitch or its and issue from JSSIP? 
> 
> Thanks in advanced!
> 
> Sherry

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From mike at jerris.com  Mon Aug 12 18:50:31 2013
From: mike at jerris.com (Michael Jerris)
Date: Mon, 12 Aug 2013 10:50:31 -0400
Subject: [Freeswitch-users] TLS/Freeswitch self signed certs
In-Reply-To: <CABkS=xqE=3D_PuSVPLWXM8gFzf_7+RPYjj4PTNAUUTU=F1iyog@mail.gmail.com>
References: <CABkS=xockpPROOhnbSaYJxt0uw2bUtKbyYus_CWqEsWz=sjB6Q@mail.gmail.com>
	<CAC0j6rfkNbJ7KMKMo7VVfGRG0uox3xZjqaw8heX2=HE09TcrEg@mail.gmail.com>
	<CABkS=xqg_Nw=X97szavNccZOT+cHA5PZ2sQ0-Tg=7DW+cqZTrg@mail.gmail.com>
	<CABkS=xqE=3D_PuSVPLWXM8gFzf_7+RPYjj4PTNAUUTU=F1iyog@mail.gmail.com>
Message-ID: <FC8B9B69-D238-428C-9799-C3EFBA64DDB5@jerris.com>

This sounds like it should be in the script for everyone.  Can you open a bug on jira.freeswitch.org for this issue.

Thanks
Mike

On Aug 6, 2013, at 2:16 AM, Peter <eidevm5 at gmail.com> wrote:

> Finally figured out the issue was related to the gentls_cert script was generating an openssl template that didn't have the required x509v3 extensions set.
> 
> I modified the script where it generates config.tpl to add
> 
>                         x509_extensions = v3_ca
> 
> to the [req] section, then I added the section:
> 
>                         [ v3_ca ]
>                         subjectKeyIdentifier=hash
>                         authorityKeyIdentifier=keyid:always,issuer
>                         basicConstraints=CA:TRUE
> 
> Now when you issue:
> 
> openssl x509 -noout -inform pem -text -in cafile.pem
> 
> you'll see the following section:
> 
>         X509v3 extensions:
>             X509v3 Subject Key Identifier:
>                 02:0A:A8:D0:5C:23:7C:8B:C4:EF:79:11:C7:0C:A8:86:71:15:59:D5
>             X509v3 Authority Key Identifier:
>                 keyid:02:0A:A8:D0:5C:23:7C:8B:C4:EF:79:11:C7:0C:A8:86:71:15:59:D5
> 
>             X509v3 Basic Constraints:
>                 CA:TRUE
> 
> If these are present, then Android will treat the cert as a standard user cert.
> 
> Then it was a simple matter of copying cafile.pem to cafile.crt on the sdcard on the Android device and using the "install from device storage" option.
> 
> When the cert installer dialog comes up, it will now detect cafile.crt as a CA cert and not user cert.
> 
> Hope this helps other people, as cert management on Android is a right pain in the $#%^.
> 
> Peter
> 
> 
> 
> On Tue, Aug 6, 2013 at 2:31 PM, Peter <eidevm5 at gmail.com> wrote:
> The reason I put it on a webserver is mostly for convenience to make it easier to install.
> 
> I tried copying cafile.pem to /sdcard on a Galaxy Note II, but when I try the "Install from device storage" option, it just comes back with:
> 
> "No certificate file found on SD card"
> 
> 
> 
> On Mon, Aug 5, 2013 at 5:51 PM, Mehroz Ashraf <mehroz.ashraf85 at gmail.com> wrote:
> Why do you want to place the cert on webserver and point android browser? If you are doing this to download cert into android then that is probably not the right approach.
> 
> I used cafile.pem (without converting it into .der format) and placed the file in  SD card or phone memory, and point out linphone to get the CA from the path. You may search in libraries where it need to tell the path. 
> 
> 
> On Mon, Aug 5, 2013 at 12:15 PM, Peter <eidevm5 at gmail.com> wrote:
> Has anyone managed to get TLS working between Android Linphone and Freeswitch?
> 
> I've done the basic TLS setup as per https://wiki.freeswitch.org/wiki/Tls
> 
> I then convert the CA cert from PEM to DER format with:
> 
> openssl x509  -inform PEM -outform der -in cafile.pem -out fs.crt
> 
> I place fs.crt on a webserver and point my Android browser to it.
> 
> When I click on fs.crt, I get the default Android Certificate installer popup, but it always says:
> 
> "Package contains: one user certificate"
> 
> ie: it thinks it is a user cert rather than a CA cert.
> 
> Android appears to be a real pain to add a CA to its trusted credential store.
> 
> Really interested if anyone has managed to get Android to import the CA cert.
> 
> Thanks
> 
> Peter

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From ccachor at gmail.com  Mon Aug 12 19:06:26 2013
From: ccachor at gmail.com (Chris Cachor)
Date: Mon, 12 Aug 2013 10:06:26 -0500
Subject: [Freeswitch-users] How to create a single outbound ESL socket at
	freeswitch startup.
In-Reply-To: <mailman.22448.1376318780.576.freeswitch-users@lists.freeswitch.org>
References: <mailman.22448.1376318780.576.freeswitch-users@lists.freeswitch.org>
Message-ID: <713D6F14-A41F-4809-8998-9D1A7AD45897@gmail.com>

Julien,

Unfortunately I've run into the same problem. Having an outbound ESL connection to push event data to a ESL server is ideal, especially in a multi-server environment (simple config setup). However, if you read the documentation, the intended behavior of the outbound ESL socket is to give control of the call to the ESL server/application. If that's your implementation, then you could simply add the socket application to your dial plan. Otherwise, an inbound socket connection is what you're looking for. 

- Chris

On Aug 12, 2013, at 9:46 AM, freeswitch-users-request at lists.freeswitch.org wrote:

> Send FreeSWITCH-users mailing list submissions to
> 	freeswitch-users at lists.freeswitch.org
> 
> To subscribe or unsubscribe via the World Wide Web, visit
> 	http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> or, via email, send a message with subject or body 'help' to
> 	freeswitch-users-request at lists.freeswitch.org
> 
> You can reach the person managing the list at
> 	freeswitch-users-owner at lists.freeswitch.org
> 
> When replying, please edit your Subject line so it is more specific
> than "Re: Contents of FreeSWITCH-users digest..."
> Today's Topics:
> 
>   1. How to create a single outbound ESL socket at	freeswitch
>      startup. (julien terrasson)
>   2.  uuid_displace: change loudness level of audio
>      (Melanie Treitinger)
>   3. Conference for Flash and WebRTC users (pablo platt)
>   4. Re: Error in launching fs_cli (Ashish Mishra)
>   5. Re: garbled audio with G726-32,	other codecs are fine
>      (Michael Jerris)
>   6. Re: Error in launching fs_cli (Steven Ayre)
>   7. Re: Issue with JSSIP + Freeswitch (Michael Jerris)
> 
> From: julien terrasson <julien.terrasson at gmail.com>
> Subject: [Freeswitch-users] How to create a single outbound ESL socket at freeswitch startup.
> Date: August 12, 2013 2:01:37 AM CDT
> To: freeswitch-users at lists.freeswitch.org
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Hi,
> 
> I would like to create a single outbound ESL socket that would establish the connection with the ESL server at freeswitch startup.
> This connexion would remains up and would used to transfer events from calls going through this server.
> I managed to do this with an inbound ESL but i would prefer doing it with an outbound ESL (it is important for me that freeswitch controls the establishement of the connection).
> Does anybody know if that's possible to start a common ESL connection (that would push event from all call) at FS startup ? If yes, where should it be provisionned (from the dialplan? in a startup script ?)
> 
> Hope somebody can help..
> 
> Regards,
> 
> Julien
> 
> 
> 
> 
> 
> 
> 
> 
> From: Melanie Treitinger <treitinger at as-infodienste.de>
> Subject: [Freeswitch-users] uuid_displace: change loudness level of audio
> Date: August 12, 2013 3:33:36 AM CDT
> To: freeswitch-users at lists.freeswitch.org
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Hi Michael,
> 
> I think I'll try sox, thanks.
> 
> 
> Melanie
> 
> 
> 
> From: pablo platt <pablo.platt at gmail.com>
> Subject: [Freeswitch-users] Conference for Flash and WebRTC users
> Date: August 12, 2013 2:40:17 AM CDT
> To: freeswitch-users at lists.freeswitch.org
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Hi,
> 
> I'm currently using a RTMP server for audio conference.
> There are no performance or latency issues because there is no encryption, transcoding or mixing involved. A 1GB VPS server can handle more than 100 users without a problem.
> 
> I'm trying to evaluate FreeSWITCH so I'll be able to support both Flash and WebRTC users in a conference.
> 
> My use case is two types of conferences:
> - A conference with up to 5 participants all speaking.
> - A conference with 20-30 participants with 2-3 speakers.
> 
> What part require more CPU?
> - Encryption (DTLS-SRTP)
> - Transcoding
> - Mixing
> 
> Does FS mix a channel separately for each participant or can it reuse the same mix?
> For example, if I have 1 speaker and 20 listeners, 10 with speex and 10 with opus, does FS produce 2 streams or 20?
> 
> Does FS exclude silent participants from the mix to save CPU?
> 
> Thanks
> 
> 
> 
> 
> 
> 
> From: Ashish Mishra <itsme.kunnu at gmail.com>
> Subject: Re: [Freeswitch-users] Error in launching fs_cli
> Date: August 12, 2013 3:51:45 AM CDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> I reinstalled freeswitch but still i am getting the same error :
> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] mod_socket is also loaded...i have also flushed the firewall rules...pls help
> 
> On Aug 6, 2013 2:30 AM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i have installed freeswitch) it gives me the following error :
> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] 
> 
> 
> 
> From: Michael Jerris <mike at jerris.com>
> Subject: Re: [Freeswitch-users] garbled audio with G726-32, other codecs are fine
> Date: August 12, 2013 9:14:45 AM CDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> There is always bit packing, but there are 2 different ways to do the bit packing.  A lot of devices get it wrong so its worth looking at that.
> 
> Mike
> 
> On Aug 4, 2013, at 1:53 AM, Ivan Mitev <imitev at c3i.bg> wrote:
> 
>> Thanks for the suggestion but I'm testing with G726-32, not AAL2-G726-32 
>> ; so bitpacking shouldn't be used. By the way when I tested with AAL2 to 
>> the linphone client I only got cracks and whitenoise, I've forgot to 
>> mention that in my post.
>> 
>> That said I've tried to uncomment and set <param name="bitpacking" 
>> value="none"/> in internal.xml ("none" is a wild guess - I couldn't find 
>> any doc on values accepted by this parameter), but that doesn't help.
>> 
>> Speex: the ATAs don't support it. And being stubborn I'd like to 
>> understand what's wrong with G726 :)
>> 
>> 
>> On 08/04/2013 05:22 AM, Jeff Leung wrote:
>>> 
>>> You can turn off G726 AAC bit-packing in spandsp.conf.xml.
>>> 
>>> By the way, there are other codecs out there you can try. SPEEX comes 
>>> to mind if all your endpoints don?t deal with the PSTN.
>>> 
>>> *From:*freeswitch-users-bounces at lists.freeswitch.org 
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
>>> *Brian Foster
>>> *Sent:* Saturday, August 3, 2013 2:37 PM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] garbled audio with G726-32, other 
>>> codecs are fine
>>> 
>>> AAC bitpacking by any chance? I thought I had a similar issue, 
>>> happened so long ago I cant remember what I did.
>>> 
>>> Thank you,
>>> 
>>> Brian Foster
>>> Project Manager/Owner's Rep.
>>> Davri Investments, Inc.
>>> O: 317-787-2686 x2102
>>> M: 317-600-9753
>>> E: bdfoster at davri.com <mailto:bdfoster at davri.com>
>>> Indianapolis, Indiana
>>> 
>>> Sent from a mobile device.
>>> 
>>> On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev at c3i.bg 
>>> <mailto:imitev at c3i.bg>> wrote:
>>> 
>>> Hello
>>> 
>>> I'm migrating an office setup from asterisk to FS and in the process I
>>> was considering using G726-32 for some bandwidth starved remote
>>> endpoints. However I only get metallic/garbled audio with that codec
>>> even when simply playing moh to the endpoint, while other codecs work
>>> fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds
>>> marginally better but still garbled and really worse than G711.
>>> 
>>> The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest
>>> (centos6 64bit host). But please don't shoot ! :) - I know about virtual
>>> environment limitations but for these tests the host is only lightly
>>> loaded, there aren't any calls to the FS instance except my tests, and
>>> the fact that it works with other codecs makes me think that
>>> virtualization is not the issue here. I may be wrong though.
>>> 
>>> Is there any guide for debugging that kind of problem before reverting
>>> to a fresh install on bare-metal with the latest HEAD ? Until now I've
>>> tried:
>>> 
>>> - improving timers ; but the default soft timer (which I guess uses
>>> timerd) works best. The time interval between sent packets on a tcpdump
>>> trace looks identical to the output of "timer_test", so that doesn't
>>> seem to be a network/jitter problem. And there's no problem with other
>>> codecs, but maybe G726-XX is specific. For info the guest's clocksource
>>> is kvm_clock, while the host uses tsc.
>>> 
>>> - using different endpoints: the production ones are Linksys PAP2
>>> ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the
>>> same thing happens with linphone on a fedora 19 laptop.
>>> 
>>> A call with rtp media going through FS without transcoding - G726-32 to
>>> G726-32 - works perfectly (I can't hear the difference with G711). The
>>> problem is only when there's transcoding to G726 (from wav for moh, or
>>> from any other codec when bridging). I've looked at the wiki, posts,
>>> changelogs, jira, ..., but am a bit at a loss now.
>>> 
>>> Any pointers ?
>>> 
>>> Except that little problem, FS rocks, and I'm happy I can finally ditch
>>> asterisk. Kudos to the core devs and contributors.
>>> 
>>> Ivan
>>> 
> 
> 
> 
> 
> 
> 
> From: Steven Ayre <steveayre at gmail.com>
> Subject: Re: [Freeswitch-users] Error in launching fs_cli
> Date: August 12, 2013 9:44:38 AM CDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> Have you checked the logfile and netstat?
> 
> 
> On 12 August 2013 09:51, Ashish Mishra <itsme.kunnu at gmail.com> wrote:
> I reinstalled freeswitch but still i am getting the same error :
> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] mod_socket is also loaded...i have also flushed the firewall rules...pls help
> 
> On Aug 6, 2013 2:30 AM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i have installed freeswitch) it gives me the following error :
> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ] 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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: Michael Jerris <mike at jerris.com>
> Subject: Re: [Freeswitch-users] Issue with JSSIP + Freeswitch
> Date: August 12, 2013 9:45:46 AM CDT
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Reply-To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> 
> 
> We only use non rfc-1918 ip's by default.  If you want to use 1918 ip's you need to tweak acls
> 
> Mike
> 
> 
> On Aug 5, 2013, at 8:24 AM, Shahrzad A. <shahrzad.aziminia at gmail.com> wrote:
> 
>> Hi  everyone
>> 
>> I'm having latest version of Freeswitch installed on Ubuntu 12.04.2 LTS  with the latest version of Openssl ('OpenSSL 1.0.1e 11 Feb 2013')
>> I'm using the default configuration and just uncommentated the ' <param name="ws-binding"  value=":5066"/> ' in internal.xml in order to have the support for webrtc. 
>> As the client I'm having JSSIP, the latest version with the adjustment to have (DtlsSrtpKeyAgreement:true). Both my clients and server located in the same network without having any firewalls therefore I'm not using any STUN server. I can make call but there is no sound! and after setting debig_level 10 and dial from extension 1003 to 5000 I'm having the following Alert in console and no sound:
>> 
>> [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!)
>> 
>> its the same case if I dial from extension 1002 to 1005! (Both are JSSIP clients) 
>> If I call from extension 1000 which is a SIP client set on my iPhone and call 1003 I'm having the following alerts but voice on both JSSIP client and SIP client:
>> 
>> 2013-08-05 14:17:05.222446 [ALERT] switch_rtp.c:4563 sofia/internal/1000 at 10.0.14.16:5060 timer while HOT
>> 2013-08-05 14:17:05.242449 [ALERT] switch_rtp.c:4546 sofia/internal/1000 at 10.0.14.16:5060 Hot Hit 1
>> 
>> And from extension 1003 (JSSIP) to extension 1000 ( SIP on Iphone) No voice and I'm getting the following alerts:
>> 
>> 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4546 sofia/internal/sip:1000 at 10.0.14.182:5065 Hot Hit 4
>> 2013-08-05 14:19:57.122450 [ALERT] switch_rtp.c:4563 sofia/internal/sip:1000 at 10.0.14.182:5065 timer while HOT
>> 2013-08-05 14:19:57.132441 [ALERT] switch_rtp.c:5672 Skip sending audio packet 172 bytes (dtls not ready!)
>> 
>> If I use bypass media or proxy media I will have voice on both JSSIP clients but cant ring any SIP clients and again no voice if I call 5000!
>> 
>> Wonder to know if there is any special setting required on FreeSwitch or its and issue from JSSIP? 
>> 
>> Thanks in advanced!
>> 
>> Sherry
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Mon Aug 12 19:21:22 2013
From: mike at jerris.com (Michael Jerris)
Date: Mon, 12 Aug 2013 11:21:22 -0400
Subject: [Freeswitch-users] FreeSwitch and Yealink
In-Reply-To: <20130810010116.3D80657E003@mail.mydcs.ca>
References: <20130807044439.9A5F8F4002@mail.mydcs.ca>
	<CAAQCs31MW_pFNiD=p=udTp4YmDueu=9rY8uDhozrYTU5JGy-uw@mail.gmail.com>
	<20130810010116.3D80657E003@mail.mydcs.ca>
Message-ID: <9A5E874A-D248-4FCF-84DE-C3AE553F416A@jerris.com>

What phone model and yealink firmware are you using?

On Aug 9, 2013, at 9:01 PM, Paul <pasha at prosperity4ever.com> wrote:

> I'm gonna try all of your guys' suggestions. 
> 
> Ivan, I do not think its codec related, I made /log 11 and siptrace on all profiles (internal/external) as well as pcap capture of the phone itself. For some strange reason the moment the PBX send a packe tot ye Yealink phone that the remote party picked up and the call should be bridges, the Yealink phone replies with a BYE packet, I'm thinking it might be a bug in the Yealink firmware, so needless to say I have gotten a hold of their support and opened a ticket, see if something comes of that, but meanwhile going to try some of the suggestions in this thread and see if I can get any close to work around.
> 
> Paul
> 
> On Fri, 9 Aug, 2013 at 10:59 AM, Nikolay Rogoshchenkov <nickolayr at gmail.com> wrote:
>> Check RTP Packet Size too.
>> 
>> --
>> Rogoshchenkov Nikolay
>> 
>> 
>> On Wed, Aug 7, 2013 at 12:44 AM, Paul <pasha at prosperity4ever.com> wrote:
>> Hi guys,
>> 
>> Has anyone had any issues using FreeSwitch with Yealink phones? My phones connect to FS via openvpn tunnel. All incoming calls work no problem, call comes through phones ring everyone can hear each other no issues, having a very strange issue though on the outgoing calls. As soon as the destination party picks up (this external calls) the call hangs up.
>> 
>> short FS LOG:
>> 
>> switch_ivr_bridge.c:475 Channel [sofia/internal/105 at 10.0.0.34] has been answered
>> sofia.c:6528 Hangup sofia/internal/105 at 10.0.0.34 [CS_EXECUTE] [ORIGINATOR_CANCEL]
>> switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 
>> 
>> So FS thinks the phone sent a BYE packet (which I can see with siptrace) but the phone's timer keeps going as if it thinks the call is supposed to keep going.
>> 
>> Internal extension to extension works fine (even if the extensions are at a different physical location and subnet).
>> 
>> I setup a second account to one of my asterisk servers and outgoing/incoming work just fine, so it seems this strange combination of FS and Yealink ... does it on 2 models T32G and T38G (only phones I have).
>> 
>> I have updated firwmare to their latest version (which in the comments say freeswitch ready)
>> 
>> Wondering if anyone else had any experience with these, or has some thoughts?
>> 
>> Thanks
>> 
>> Paul
>> 
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From mike at jerris.com  Mon Aug 12 19:24:32 2013
From: mike at jerris.com (Michael Jerris)
Date: Mon, 12 Aug 2013 11:24:32 -0400
Subject: [Freeswitch-users] Real-time billing application for the
	FreeSWITCH (mod_lua, mod_perl or ESL)
In-Reply-To: <002101ce952e$6e3fc3f0$4abf4bd0$@verizon.net>
References: <002101ce952e$6e3fc3f0$4abf4bd0$@verizon.net>
Message-ID: <EAF2323F-AFF1-42C7-9CD5-E805B56F6CF5@jerris.com>

If your happy with C, go for it.  You can take a look at mod_xml_cdr and mod_nibblebill for some hints on hooks for billing.  I might use the state hooks for reporting instead and the data you get from that one as you are less time constrained to handle the data in that state.

Mike

On Aug 9, 2013, at 2:29 PM, Sami Montour <smontour at verizon.net> wrote:

> Hello Everyone,
> I am a long time user of OpenSIPS and have recently been tasked with developing a call control/real-time billing application for the FreeSWITCH. I would like to develop the application in either C/C++, Perl or lua (totally new to lua) in this order.
>  
> I have used ESL to develop some basic functionality of the application using Perl. For instance, when the call comes in, I pass over control from FreeSWITCH to the call control application using outbound TCP socket. Then the call control application listens for CHANNEL_ANSWER event to start billing process and CHANNEL_HANGUP event to stop billing. The application does some database work as well to retrieve and check user status and balance and all that stuff. My main concern is with load testing because of the TCP connection between the FreeSWITCH and call control application and also TCP connection to database to retrieve subscriber info. The basic functionality seems to work fine but haven?t tested it with high call volume, which I am about to do.
>  
> The last couple of days I have been experimenting with mod_lua but can?t find a method under ?Session? to get information related to CHANNEL_ANSWER and CHANNEL_HANGUP events. The ?session:geVariable? does not have variables related to session answer time or hangup time, something similar to ?$e->getHeader("Caller-Channel-Answered-Time")?  in ESL. I must be missing something very simple here. Again, my experience with mod_lua is only a couple of days old.
>  
> I am new Freeswitch and thought I would seek the advice of all the experts and people with more experience with Freeswitch. For this type of application, am I better off with mod_lua, or mod_perl, or shall I stick with ESL using Perl? How about ESL with C/C++? Will I gain some performance gain there?
>  
> Any feedback would be appreciated. Thanks.
>  
> Sam
> 
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From mike at jerris.com  Mon Aug 12 19:28:30 2013
From: mike at jerris.com (Michael Jerris)
Date: Mon, 12 Aug 2013 11:28:30 -0400
Subject: [Freeswitch-users] problem with menuexecapp
In-Reply-To: <10175.1376191370@ccs.covici.com>
References: <10175.1376191370@ccs.covici.com>
Message-ID: <58C685B3-6E78-4A42-9F2D-8559DB8F7228@jerris.com>

The problem here is its already out of the dtmf listener right after it gets to max digits.  You would have to flush at the start of your app, and even then its a race condition depending on how slow they type.


On Aug 10, 2013, at 11:22 PM, covici at ccs.covici.com wrote:

> Hi.  I am having a problem with the freeswitch ivr.  If someone types a
> number of digits equal to the maximum length, and then types a #, the #
> is saved and passed to wherever the transfer or whatever the menuexecapp
> wants to go.  Now I have an app which responds to # by hanging up, so
> this makes it difficult.  Is there a way that either the app could flush
> the dtmf queue when it comes in, or could (and this would be much
> better) the # be ignored when the maximum length is reached?    I
> thought this was the way it was supposed to work, anyway.



From mike at jerris.com  Mon Aug 12 19:33:37 2013
From: mike at jerris.com (Michael Jerris)
Date: Mon, 12 Aug 2013 11:33:37 -0400
Subject: [Freeswitch-users] How to create a single outbound ESL socket
	at freeswitch startup.
In-Reply-To: <CAFi=dz9iXE6ow0pHg4tw+d9cP12we5Aadh7nv6TLgqCPntzhJw@mail.gmail.com>
References: <CAFi=dz9iXE6ow0pHg4tw+d9cP12we5Aadh7nv6TLgqCPntzhJw@mail.gmail.com>
Message-ID: <1721DDE8-9502-4793-8F61-0095DAF72D0F@jerris.com>

Outbound is only in the context of a call.. This would require additions to mod_event_socket.  I'd be happy to review a patch to add this feature.

Mike

On Aug 12, 2013, at 3:01 AM, julien terrasson <julien.terrasson at gmail.com> wrote:

> Hi,
> 
> I would like to create a single outbound ESL socket that would establish the connection with the ESL server at freeswitch startup.
> This connexion would remains up and would used to transfer events from calls going through this server.
> I managed to do this with an inbound ESL but i would prefer doing it with an outbound ESL (it is important for me that freeswitch controls the establishement of the connection).
> Does anybody know if that's possible to start a common ESL connection (that would push event from all call) at FS startup ? If yes, where should it be provisionned (from the dialplan? in a startup script ?)
> 



From mike at jerris.com  Mon Aug 12 19:37:56 2013
From: mike at jerris.com (Michael Jerris)
Date: Mon, 12 Aug 2013 11:37:56 -0400
Subject: [Freeswitch-users] Conference for Flash and WebRTC users
In-Reply-To: <CANdLC8VAzwRcFBHog5RT_M8=AZqTdZNdeNAcFDAzPU0AFDgHkQ@mail.gmail.com>
References: <CANdLC8VAzwRcFBHog5RT_M8=AZqTdZNdeNAcFDAzPU0AFDgHkQ@mail.gmail.com>
Message-ID: <96083A3E-7C60-4E4B-BF42-5D1BBC651710@jerris.com>


On Aug 12, 2013, at 3:40 AM, pablo platt <pablo.platt at gmail.com> wrote:

> Hi,
> 
> I'm currently using a RTMP server for audio conference.
> There are no performance or latency issues because there is no encryption, transcoding or mixing involved. A 1GB VPS server can handle more than 100 users without a problem.
> 
> I'm trying to evaluate FreeSWITCH so I'll be able to support both Flash and WebRTC users in a conference.
> 
> My use case is two types of conferences:
> - A conference with up to 5 participants all speaking.
> - A conference with 20-30 participants with 2-3 speakers.
> 
> What part require more CPU?
> - Encryption (DTLS-SRTP)
> - Transcoding
> - Mixing
> 

I'm pretty sure it will be a toss up between srtp (the dtls-srtp negotiation will be minimal in comparison) and transcoding depending on the codec.  g711 will be for sure less than srtp.  g729 will be more than srtp.  The mixing is typically pretty minor on cpu in comparison depending on the number of talking participants, which is typically 0 or 1.

> Does FS mix a channel separately for each participant or can it reuse the same mix?

It does one mix, but for transcoding it does a separate encode for each participant.

> For example, if I have 1 speaker and 20 listeners, 10 with speex and 10 with opus, does FS produce 2 streams or 20?
> 
> Does FS exclude silent participants from the mix to save CPU?

Yes.  This does require us to decode all incoming streams not marked as muted.




From covici at ccs.covici.com  Mon Aug 12 19:57:26 2013
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 12 Aug 2013 11:57:26 -0400
Subject: [Freeswitch-users] problem with menuexecapp
In-Reply-To: <58C685B3-6E78-4A42-9F2D-8559DB8F7228@jerris.com>
References: <10175.1376191370@ccs.covici.com>
	<58C685B3-6E78-4A42-9F2D-8559DB8F7228@jerris.com>
Message-ID: <8352.1376323046@ccs.covici.com>

Or, make the max lengthlarger and never use that many digits -- I
thought it used to work differently, but may be not.

Michael Jerris <mike at jerris.com> wrote:

> The problem here is its already out of the dtmf listener right after it gets to max digits.  You would have to flush at the start of your app, and even then its a race condition depending on how slow they type.
> 
> 
> On Aug 10, 2013, at 11:22 PM, covici at ccs.covici.com wrote:
> 
> > Hi.  I am having a problem with the freeswitch ivr.  If someone types a
> > number of digits equal to the maximum length, and then types a #, the #
> > is saved and passed to wherever the transfer or whatever the menuexecapp
> > wants to go.  Now I have an app which responds to # by hanging up, so
> > this makes it difficult.  Is there a way that either the app could flush
> > the dtmf queue when it comes in, or could (and this would be much
> > better) the # be ignored when the maximum length is reached?    I
> > thought this was the way it was supposed to work, anyway.
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
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 nreis at wavecom.pt  Mon Aug 12 23:29:07 2013
From: nreis at wavecom.pt (Nuno Reis)
Date: Mon, 12 Aug 2013 20:29:07 +0100
Subject: [Freeswitch-users] freeswitch HANGING (process exists but stops
	responding)
Message-ID: <CAE2+g+JvWmz2fSF1q_Woaz7uiqYPeO7macC2s-pnvqmOabrkfA@mail.gmail.com>

Hello all.

I've been experiencing a problem with freeswitch where it stops responding
(doesn't answer any SIP REQUESTS) and produces the following output in the
log file every time the symptom happens:

2013-08-12 15:18:31.488906 [WARNING] switch_event.c:552 Create event
dispatch thread 1
2013-08-12 15:18:31.578906 [WARNING] switch_event.c:554 Create additional
event dispatch thread 2
2013-08-12 15:18:31.598907 [WARNING] switch_event.c:554 Create additional
event dispatch thread 3
2013-08-12 15:18:31.648904 [WARNING] switch_event.c:554 Create additional
event dispatch thread 4
2013-08-12 15:18:31.728905 [WARNING] switch_event.c:554 Create additional
event dispatch thread 5
2013-08-12 15:19:04.348914 [CONSOLE] sofia.c:1602 MSG Thread 4 Started

After this, i can see that the process exists and i can get to the fs_cli
prompt, although none of the normal command like: show calls work. The
command simply hangs without returning back a response.
I think this could be memory related, although can't find a reasonable
explanation to that. I had 4GB of physical RAM available + 512MB for SWAP
on a CentOS x86_64 machine and the OS was using like about 800MB but if i
considered  the cached amount of memory, it was like a total of 3.8GB which
from what i understand is normal in linux, since the kernel tries to cache
all the available memory over time.
Nevertheless i've decided to give the machine more 4GB RAM and SWAP is now
16GB, so now i've 8GB total for physical RAM and 16GB SWAP which solved the
problem at least temporarily and made freeswitch stop getting back into the
symptom described above.
I can see that the cached memory is again growing over time. I still don't
know if I'll end up with the same symptom again when the total used memory
(used+ cached) gets near the 8GB.

Anyone has had this symptom before too?
Some feedback is appreciated.

Thanks.

--

*Nuno Miguel Reis* | *Unified Communication** Systems*
M. +351 913907481 | nreis at wavecom.pt
WAVECOM-Solu??es R?dio, S.A.
Cacia Park | Rua do Progresso, Lote 15
3800-639 AVEIRO | Portugal
T. +351 309 700 225 | F. +351 234 919 191
*GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
www.wavecom.pt** <http://www.wavecom.pt/>*

[image: Description: Description:
WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php>

[image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>
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From nreis at wavecom.pt  Mon Aug 12 23:35:44 2013
From: nreis at wavecom.pt (Nuno Reis)
Date: Mon, 12 Aug 2013 20:35:44 +0100
Subject: [Freeswitch-users] freeswitch HANGING (process exists but stops
	responding)
In-Reply-To: <CAE2+g+JvWmz2fSF1q_Woaz7uiqYPeO7macC2s-pnvqmOabrkfA@mail.gmail.com>
References: <CAE2+g+JvWmz2fSF1q_Woaz7uiqYPeO7macC2s-pnvqmOabrkfA@mail.gmail.com>
Message-ID: <CAE2+g+KR-kZkrhxH3NekjsCk5+CiKG2j6Q5UTiR6TWa-QpGU9Q@mail.gmail.com>

Another eventually important tip:

This is happening at least with freeswitch versions 1.2.3 and 1.2.8.

Thanks.

--

*Nuno Miguel Reis* | *Unified Communication** Systems*
M. +351 913907481 | nreis at wavecom.pt
WAVECOM-Solu??es R?dio, S.A.
Cacia Park | Rua do Progresso, Lote 15
3800-639 AVEIRO | Portugal
T. +351 309 700 225 | F. +351 234 919 191
*GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
www.wavecom.pt** <http://www.wavecom.pt/>*

[image: Description: Description:
WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php>

[image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>




On Mon, Aug 12, 2013 at 8:29 PM, Nuno Reis <nreis at wavecom.pt> wrote:

> Hello all.
>
> I've been experiencing a problem with freeswitch where it stops responding
> (doesn't answer any SIP REQUESTS) and produces the following output in the
> log file every time the symptom happens:
>
> 2013-08-12 15:18:31.488906 [WARNING] switch_event.c:552 Create event
> dispatch thread 1
> 2013-08-12 15:18:31.578906 [WARNING] switch_event.c:554 Create additional
> event dispatch thread 2
> 2013-08-12 15:18:31.598907 [WARNING] switch_event.c:554 Create additional
> event dispatch thread 3
> 2013-08-12 15:18:31.648904 [WARNING] switch_event.c:554 Create additional
> event dispatch thread 4
> 2013-08-12 15:18:31.728905 [WARNING] switch_event.c:554 Create additional
> event dispatch thread 5
> 2013-08-12 15:19:04.348914 [CONSOLE] sofia.c:1602 MSG Thread 4 Started
>
> After this, i can see that the process exists and i can get to the fs_cli
> prompt, although none of the normal command like: show calls work. The
> command simply hangs without returning back a response.
> I think this could be memory related, although can't find a reasonable
> explanation to that. I had 4GB of physical RAM available + 512MB for SWAP
> on a CentOS x86_64 machine and the OS was using like about 800MB but if i
> considered  the cached amount of memory, it was like a total of 3.8GB which
> from what i understand is normal in linux, since the kernel tries to cache
> all the available memory over time.
> Nevertheless i've decided to give the machine more 4GB RAM and SWAP is now
> 16GB, so now i've 8GB total for physical RAM and 16GB SWAP which solved the
> problem at least temporarily and made freeswitch stop getting back into the
> symptom described above.
> I can see that the cached memory is again growing over time. I still don't
> know if I'll end up with the same symptom again when the total used memory
> (used+ cached) gets near the 8GB.
>
> Anyone has had this symptom before too?
> Some feedback is appreciated.
>
> Thanks.
>
> --
>
> *Nuno Miguel Reis* | *Unified Communication** Systems*
> M. +351 913907481 | nreis at wavecom.pt
> WAVECOM-Solu??es R?dio, S.A.
> Cacia Park | Rua do Progresso, Lote 15
> 3800-639 AVEIRO | Portugal
> T. +351 309 700 225 | F. +351 234 919 191
> *GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
> www.wavecom.pt** <http://www.wavecom.pt/>*
>
> [image: Description: Description: WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php>
>
> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>
>
>
>
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From krice at freeswitch.org  Mon Aug 12 23:38:49 2013
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 12 Aug 2013 14:38:49 -0500
Subject: [Freeswitch-users] freeswitch HANGING (process exists but stops
 responding)
In-Reply-To: <CAE2+g+JvWmz2fSF1q_Woaz7uiqYPeO7macC2s-pnvqmOabrkfA@mail.gmail.com>
Message-ID: <CE2EA3F9.78CDB%krice@freeswitch.org>

If this happens again please check jira to see if there is another instance
of this happening. If there is not open a jira, and use gcore to get a core
file, and then follow the standard info on the wiki for obtaining a stack
trace and attach the resultant output to the ticket.

Sounds like you might be getting hammered by something like sipvicious


On 8/12/13 2:29 PM, "Nuno Reis" <nreis at wavecom.pt> wrote:

> Hello all.
> 
> I've been experiencing a problem with freeswitch where it stops responding
> (doesn't answer any SIP REQUESTS) and produces the following output in the log
> file every time the symptom happens:
> 
> 2013-08-12 15:18:31.488906 [WARNING] switch_event.c:552 Create event dispatch
> thread 1
> 2013-08-12 15:18:31.578906 [WARNING] switch_event.c:554 Create additional
> event dispatch thread 2
> 2013-08-12 15:18:31.598907 [WARNING] switch_event.c:554 Create additional
> event dispatch thread 3
> 2013-08-12 15:18:31.648904 [WARNING] switch_event.c:554 Create additional
> event dispatch thread 4
> 2013-08-12 15:18:31.728905 [WARNING] switch_event.c:554 Create additional
> event dispatch thread 5
> 2013-08-12 15:19:04.348914 [CONSOLE] sofia.c:1602 MSG Thread 4 Started
> 
> After this, i can see that the process exists and i can get to the fs_cli
> prompt, although none of the normal command like: show calls work. The command
> simply hangs without returning back a response.
> I think this could be memory related, although can't find a reasonable
> explanation to that. I had 4GB of physical RAM available + 512MB for SWAP on a
> CentOS x86_64 machine and the OS was using like about 800MB but if i
> considered? the cached amount of memory, it was like a total of 3.8GB which
> from what i understand is normal in linux, since the kernel tries to cache all
> the available memory over time.
> Nevertheless i've decided to give the machine more 4GB RAM and SWAP is now
> 16GB, so now i've 8GB total for physical RAM and 16GB SWAP which solved the
> problem at least temporarily and made freeswitch stop getting back into the
> symptom described above.
> I can see that the cached memory is again growing over time. I still don't
> know if I'll end up with the same symptom again when the total used memory
> (used+ cached) gets near the 8GB.
> 
> Anyone has had this symptom before too?
> Some feedback is appreciated.
> 
> Thanks.
> --
> 
> Nuno Miguel Reis | Unified Communication Systems
> M. +351 913907481 | nreis at wavecom.pt <mailto:nreis at wavecom.pt>
> WAVECOM-Solu??es R?dio, S.A.
> Cacia Park | Rua do Progresso, Lote 15
> 3800-639 AVEIRO | Portugal
> T. +351 309 700 225 | F. +351 234 919 191
> GPS 
> <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a61
> 44f0df88>  | www.wavecom.pt <http://www.wavecom.pt/>  <http://www.wavecom.pt/>
> 
>  <http://www.wavecom.pt/pt/wavecom/premios.php>
> 
>  <http://www.wavecom.pt/pt/mail_eventos.php>
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From rafal.gwizdala at gmail.com  Mon Aug 12 23:39:26 2013
From: rafal.gwizdala at gmail.com (Rafal Gwizdala)
Date: Mon, 12 Aug 2013 21:39:26 +0200
Subject: [Freeswitch-users] How to create a single outbound ESL socket
 at freeswitch startup.
In-Reply-To: <1721DDE8-9502-4793-8F61-0095DAF72D0F@jerris.com>
References: <CAFi=dz9iXE6ow0pHg4tw+d9cP12we5Aadh7nv6TLgqCPntzhJw@mail.gmail.com>
	<1721DDE8-9502-4793-8F61-0095DAF72D0F@jerris.com>
Message-ID: <CALtno0985RhEjPOrWQsZKJ+hFKVzfA5Jr0qR+6Ti8raWsHT6xQ@mail.gmail.com>

Maybe it would be enough if FS sent some signal to your program on startup
(eg by doing an http request or some sort of ping) so you would know when
to start the inbound ESL connection?


On Mon, Aug 12, 2013 at 5:33 PM, Michael Jerris <mike at jerris.com> wrote:

> Outbound is only in the context of a call.. This would require additions
> to mod_event_socket.  I'd be happy to review a patch to add this feature.
>
> Mike
>
> On Aug 12, 2013, at 3:01 AM, julien terrasson <julien.terrasson at gmail.com>
> wrote:
>
> > Hi,
> >
> > I would like to create a single outbound ESL socket that would establish
> the connection with the ESL server at freeswitch startup.
> > This connexion would remains up and would used to transfer events from
> calls going through this server.
> > I managed to do this with an inbound ESL but i would prefer doing it
> with an outbound ESL (it is important for me that freeswitch controls the
> establishement of the connection).
> > Does anybody know if that's possible to start a common ESL connection
> (that would push event from all call) at FS startup ? If yes, where should
> it be provisionned (from the dialplan? in a startup script ?)
> >
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Mon Aug 12 23:40:38 2013
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 12 Aug 2013 14:40:38 -0500
Subject: [Freeswitch-users] freeswitch HANGING (process exists but stops
 responding)
In-Reply-To: <CAE2+g+KR-kZkrhxH3NekjsCk5+CiKG2j6Q5UTiR6TWa-QpGU9Q@mail.gmail.com>
Message-ID: <CE2EA466.78CDE%krice@freeswitch.org>

You might want to update to lastest and test then it could have already been
resolved... 


On 8/12/13 2:35 PM, "Nuno Reis" <nreis at wavecom.pt> wrote:

> Another eventually important tip:
> 
> This is happening at least with freeswitch versions 1.2.3 and 1.2.8.
> 
> Thanks.
> --
> 
> Nuno Miguel Reis | Unified Communication Systems
> M. +351 913907481 | nreis at wavecom.pt <mailto:nreis at wavecom.pt>
> WAVECOM-Solu??es R?dio, S.A.
> Cacia Park | Rua do Progresso, Lote 15
> 3800-639 AVEIRO | Portugal
> T. +351 309 700 225 | F. +351 234 919 191
> GPS 
> <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a61
> 44f0df88>  | www.wavecom.pt <http://www.wavecom.pt/>  <http://www.wavecom.pt/>
> 
>  <http://www.wavecom.pt/pt/wavecom/premios.php>
> 
>  <http://www.wavecom.pt/pt/mail_eventos.php>
> 
> 
> 
> 
> On Mon, Aug 12, 2013 at 8:29 PM, Nuno Reis <nreis at wavecom.pt> wrote:
>> Hello all.
>> 
>> I've been experiencing a problem with freeswitch where it stops responding
>> (doesn't answer any SIP REQUESTS) and produces the following output in the
>> log file every time the symptom happens:
>> 
>> 2013-08-12 15:18:31.488906 [WARNING] switch_event.c:552 Create event dispatch
>> thread 1
>> 2013-08-12 15:18:31.578906 [WARNING] switch_event.c:554 Create additional
>> event dispatch thread 2
>> 2013-08-12 15:18:31.598907 [WARNING] switch_event.c:554 Create additional
>> event dispatch thread 3
>> 2013-08-12 15:18:31.648904 [WARNING] switch_event.c:554 Create additional
>> event dispatch thread 4
>> 2013-08-12 15:18:31.728905 [WARNING] switch_event.c:554 Create additional
>> event dispatch thread 5
>> 2013-08-12 15:19:04.348914 [CONSOLE] sofia.c:1602 MSG Thread 4 Started
>> 
>> After this, i can see that the process exists and i can get to the fs_cli
>> prompt, although none of the normal command like: show calls work. The
>> command simply hangs without returning back a response.
>> I think this could be memory related, although can't find a reasonable
>> explanation to that. I had 4GB of physical RAM available + 512MB for SWAP on
>> a CentOS x86_64 machine and the OS was using like about 800MB but if i
>> considered? the cached amount of memory, it was like a total of 3.8GB which
>> from what i understand is normal in linux, since the kernel tries to cache
>> all the available memory over time.
>> Nevertheless i've decided to give the machine more 4GB RAM and SWAP is now
>> 16GB, so now i've 8GB total for physical RAM and 16GB SWAP which solved the
>> problem at least temporarily and made freeswitch stop getting back into the
>> symptom described above.
>> I can see that the cached memory is again growing over time. I still don't
>> know if I'll end up with the same symptom again when the total used memory
>> (used+ cached) gets near the 8GB.
>> 
>> Anyone has had this symptom before too?
>> Some feedback is appreciated.
>> 
>> Thanks.
>> --
>> 
>> Nuno Miguel Reis | Unified Communication Systems
>> M. +351 913907481 | nreis at wavecom.pt <mailto:nreis at wavecom.pt>
>> WAVECOM-Solu??es R?dio, S.A.
>> Cacia Park | Rua do Progresso, Lote 15
>> 3800-639 AVEIRO | Portugal
>> T. +351 309 700 225 | F. +351 234 919 191
>> GPS 
>> <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6
>> 144f0df88>  | www.wavecom.pt <http://www.wavecom.pt/>
>> <http://www.wavecom.pt/>
>> 
>>  <http://www.wavecom.pt/pt/wavecom/premios.php>
>> 
>>  <http://www.wavecom.pt/pt/mail_eventos.php>
>> 
>> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From krice at freeswitch.org  Mon Aug 12 23:48:07 2013
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 12 Aug 2013 14:48:07 -0500
Subject: [Freeswitch-users] Weekly FreeSWITCH conference call
Message-ID: <CE2EA627.78CE1%krice@freeswitch.org>

Hey Guys,

Well Last week, we took a week off for ClueCon 2013 and it was great! If you
missed it, theres things popping up all over social media.

However we now have about 50 weekly Episodes of the FreeSWITCH Community
conference call to fill!
Want to come on and talk about your project? Email Me!
Have a suggestion for someone to get on the call to talk about their
project? Email Me! 
Do you use FreeSWITCH at your job and you want to show us what cool thing
you have done? Email Me!

K


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From karl at xtronics.com  Tue Aug 13 00:02:06 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Mon, 12 Aug 2013 15:02:06 -0500
Subject: [Freeswitch-users] blocking IP addresses and fail2ban setup
In-Reply-To: <58C685B3-6E78-4A42-9F2D-8559DB8F7228@jerris.com>
References: <10175.1376191370@ccs.covici.com>
	<58C685B3-6E78-4A42-9F2D-8559DB8F7228@jerris.com>
Message-ID: <52093F3E.6040506@xtronics.com>

First, the wiki page about fail2ban

http://wiki.freeswitch.org/wiki/Fail2ban

has this bit of tantalizing, but cryptic advice:

"Enable "log-auth-failures" on each Sofia profile to monitor -- this requires a high enough loglevel 
on your logs to save these messages. "

What does "high enough" mean?

( It looks like the default in autoload_configs/syslog.conf.xml is warning )

Why isn't log-auth-failures the default?

,.,

Looking at simple ways to block lists of IP addresses - there are WRONG ways to do this.

Huge lists blocked as dynamic will reduce performance of your firewall.

Looks like the best way is via ipsets:

http://www.shorewall.net/ipsets.html

So if you want to use Brain West?s blacklist  ( http://daffy.bkw.org/blacklist.txt ) best to read up 
on ipsets.

,.,

The other bit is that it is best to block at the firewall if possible - this looks like it can be 
done by setting up fail2ban on the freeswitch box and setting up the action to use the ban command 
over ssh.

actionban = ssh user at firewall.com shorewall drop <ip>

actionunban = ssh user at firewall.com shorewall allow <ip>


fail2ban using shorewall uses the dynamic method (appropriately due to the smaller number of IPs) If 
you want to see a list of what is currently blocked:

$ shorewall show dynamic

Will dump out a list of the currently banned IP addresses.



--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

The society that puts equality before freedom will
end up with neither. The society that puts freedom
before equality will end up with a great measure of both.
- Milton Freidman
--------------------------------------------------------------------------------


From pasha at prosperity4ever.com  Tue Aug 13 01:20:18 2013
From: pasha at prosperity4ever.com (Paul)
Date: Mon, 12 Aug 2013 21:13:18 -0007
Subject: [Freeswitch-users] FreeSwitch and Yealink
In-Reply-To: <9A5E874A-D248-4FCF-84DE-C3AE553F416A@jerris.com>
References: <20130807044439.9A5F8F4002@mail.mydcs.ca>
	<CAAQCs31MW_pFNiD=p=udTp4YmDueu=9rY8uDhozrYTU5JGy-uw@mail.gmail.com>
	<20130810010116.3D80657E003@mail.mydcs.ca>
	<9A5E874A-D248-4FCF-84DE-C3AE553F416A@jerris.com>
Message-ID: <20130812212027.B410A86E9C2@mail.mydcs.ca>

This is happening on both T38Gs and T32Gs, I have physical access to 
the T32, firmware 32.70.0.132

Paul

On Mon, 12 Aug, 2013 at 8:21 AM, Michael Jerris <mike at jerris.com> wrote:
> What phone model and yealink firmware are you using?
> 
> On Aug 9, 2013, at 9:01 PM, Paul <pasha at prosperity4ever.com> wrote:
> 
>> I'm gonna try all of your guys' suggestions.?
>> 
>> Ivan, I do not think its codec related, I made /log 11 and siptrace 
>> on all profiles (internal/external) as well as pcap capture of the 
>> phone itself. For some strange reason the moment the PBX send a 
>> packe tot ye Yealink phone that the remote party picked up and the 
>> call should be bridges, the Yealink phone replies with a BYE packet, 
>> I'm thinking it might be a bug in the Yealink firmware, so needless 
>> to say I have gotten a hold of their support and opened a ticket, 
>> see if something comes of that, but meanwhile going to try some of 
>> the suggestions in this thread and see if I can get any close to 
>> work around.
>> 
>> Paul
>> 
>> On Fri, 9 Aug, 2013 at 10:59 AM, Nikolay Rogoshchenkov 
>> <nickolayr at gmail.com> wrote:
>>> Check?RTP Packet Size too.
>>> 
>>> --
>>> Rogoshchenkov Nikolay
>>> 
>>> 
>>> On Wed, Aug 7, 2013 at 12:44 AM, Paul <pasha at prosperity4ever.com> 
>>> wrote:
>>>> Hi guys,
>>>> 
>>>> Has anyone had any issues using FreeSwitch with Yealink phones? My 
>>>> phones connect to FS via openvpn tunnel. All incoming calls work 
>>>> no problem, call comes through phones ring everyone can hear each 
>>>> other no issues, having a very strange issue though on the 
>>>> outgoing calls. As soon as the destination party picks up (this 
>>>> external calls) the call hangs up.
>>>> 
>>>> short FS LOG:
>>>> 
>>>> switch_ivr_bridge.c:475 Channel [sofia/internal/105 at 10.0.0.34] has 
>>>> been answered
>>>> sofia.c:6528 Hangup sofia/internal/105 at 10.0.0.34 [CS_EXECUTE] 
>>>> [ORIGINATOR_CANCEL]
>>>> switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 
>>>> [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]?
>>>> 
>>>> So FS thinks the phone sent a BYE packet (which I can see with 
>>>> siptrace) but the phone's timer keeps going as if it thinks the 
>>>> call is supposed to keep going.
>>>> 
>>>> Internal extension to extension works fine (even if the extensions 
>>>> are at a different physical location and subnet).
>>>> 
>>>> I setup a second account to one of my asterisk servers and 
>>>> outgoing/incoming work just fine, so it seems this strange 
>>>> combination of FS and Yealink ... does it on 2 models T32G and 
>>>> T38G (only phones I have).
>>>> 
>>>> I have updated firwmare to their latest version (which in the 
>>>> comments say freeswitch ready)
>>>> 
>>>> Wondering if anyone else had any experience with these, or has 
>>>> some thoughts?
>>>> 
>>>> Thanks
>>>> 
>>>> Paul
>>>> 
>>>> 
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From mishehu at freeswitch.org  Tue Aug 13 01:45:00 2013
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Mon, 12 Aug 2013 16:45:00 -0500
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAFiqYuneYGGujBH-a_hanzEQ72Rdiks5kfwntwz2QGp1YC9A7A@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<CAMvhjJtBONsXnrhRoDdoMTUPhHb1NjVRSJOsdbt5u+0tjqaBug@mail.gmail.com>
	<CAFiqYuneYGGujBH-a_hanzEQ72Rdiks5kfwntwz2QGp1YC9A7A@mail.gmail.com>
Message-ID: <5209575C.9030203@freeswitch.org>

I have the nagging feeling that Ashish isn't telling the whole story here...

Ashish -  what happens if you are on the machine that is running 
FreeSWITCH and you execute `telnet localhost 8021` ?  Do you see 
something like this:

# telnet localhost 8021
Trying 127.0.0.1...
Connected to localhost.
Escape character is '^]'.
Content-Type: auth/request

If you are running with the standard configurations this should work in 
this fashion.  If you have changed the 
conf/autoload_configs/event_socket.conf.xml , then paste those 
configurations.  (If too lengthy, please post to 
http://pastebin.freeswitch.org )

-Yossi

On 08/12/2013 09:44 AM, Steven Ayre wrote:
> Have you checked the logfile and netstat?
>
>
> On 12 August 2013 09:51, Ashish Mishra <itsme.kunnu at gmail.com 
> <mailto:itsme.kunnu at gmail.com>> wrote:
>
>     I reinstalled freeswitch but still i am getting the same error :
>     fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>     mod_socket is also loaded...i have also flushed the firewall
>     rules...pls help
>
>     On Aug 6, 2013 2:30 AM, "Ashish Mishra" <itsme.kunnu at gmail.com
>     <mailto:itsme.kunnu at gmail.com>> wrote:
>
>         When i am trying to launch fs_cli on my ubuntu 12.04 machine
>         (on which i have installed freeswitch) it gives me the
>         following error :
>         fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 marketing at cluecon.com  Tue Aug 13 02:40:28 2013
From: marketing at cluecon.com (Michael Collins)
Date: Mon, 12 Aug 2013 15:40:28 -0700
Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes - ClueCon 2013
	Edition
Message-ID: <CAKzWOxXJyaQuJUxScKexZdPzyhVNsDUFSeVgE9s=03NT8sZddQ@mail.gmail.com>

Hello all!

We've survived another ClueCon <http://www.cluecon.com/> season - and
ClueCon 2013 was brilliant! We had 29 talks, 5 lightning talks, and a
security BoF that featured Philip Zimmermann. Ken Rice has posted some
pictures here<https://plus.google.com/115131822208898699412/posts/WjwVZxShGK5>.
We are currently gathering all of the presentation files and YouTube links.
Once they are collected they will be available from the presentations
pages. Just click on the title of a presentation to see the download links.

This week Ken and I, along with any other ClueCon attendees, will be doing
a recap of some of the things that took place at ClueCon 2013. If you were
at ClueCon 2013 and have an experience you'd like to share with the group
then please join us on this week's conference
call<https://wiki.freeswitch.org/wiki/FS_weekly_2013_08_14>
.

Thanks to everyone for making this year's event the best one yet. We look
forward to 2014!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From bob.mccarthy at experient.com  Tue Aug 13 03:02:13 2013
From: bob.mccarthy at experient.com (Bob McCarthy)
Date: Mon, 12 Aug 2013 17:02:13 -0600
Subject: [Freeswitch-users] Invalid application oreka_record
Message-ID: <009c01ce97af$fc724020$f556c060$@experient.com>

I am trying to test/use mod_oreka, I have configured via
http://wiki.freeswitch.org/wiki/Mod_oreka but in the dialplan 

I get 2013-08-12 16:35:09.231636 [ERR] switch_core_session.c:2557 Invalid
Application oreka_record

 

Is there something more I need to do to compile/inlude the oreka module ?
mod_oreka.so is present in the mod directory

 

Bob

 

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From krice at freeswitch.org  Tue Aug 13 04:01:44 2013
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 12 Aug 2013 19:01:44 -0500
Subject: [Freeswitch-users] Invalid application oreka_record
In-Reply-To: <009c01ce97af$fc724020$f556c060$@experient.com>
References: <009c01ce97af$fc724020$f556c060$@experient.com>
Message-ID: <D0BD8814-3F42-4F6F-B7BD-5185D8382A58@freeswitch.org>

you need to add it to your modules.conf.xml so it loads at freeswitch start time, for a 1 time load, just "load mod_oreka" from fs_cli.

Ken
Sent from my iPad

On Aug 12, 2013, at 18:02, "Bob McCarthy" <bob.mccarthy at experient.com> wrote:

> I am trying to test/use mod_oreka, I have configured via http://wiki.freeswitch.org/wiki/Mod_oreka but in the dialplan
> I get 2013-08-12 16:35:09.231636 [ERR] switch_core_session.c:2557 Invalid Application oreka_record
>  
> Is there something more I need to do to compile/inlude the oreka module ?  mod_oreka.so is present in the mod directory
>  
> Bob
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Tue Aug 13 04:18:48 2013
From: eidevm5 at gmail.com (Peter)
Date: Tue, 13 Aug 2013 10:18:48 +1000
Subject: [Freeswitch-users] TLS/Freeswitch self signed certs
In-Reply-To: <FC8B9B69-D238-428C-9799-C3EFBA64DDB5@jerris.com>
References: <CABkS=xockpPROOhnbSaYJxt0uw2bUtKbyYus_CWqEsWz=sjB6Q@mail.gmail.com>
	<CAC0j6rfkNbJ7KMKMo7VVfGRG0uox3xZjqaw8heX2=HE09TcrEg@mail.gmail.com>
	<CABkS=xqg_Nw=X97szavNccZOT+cHA5PZ2sQ0-Tg=7DW+cqZTrg@mail.gmail.com>
	<CABkS=xqE=3D_PuSVPLWXM8gFzf_7+RPYjj4PTNAUUTU=F1iyog@mail.gmail.com>
	<FC8B9B69-D238-428C-9799-C3EFBA64DDB5@jerris.com>
Message-ID: <CABkS=xpqry45O7XX7s4R5ib-uwD-wyW7TqAZtTVRujr0XpzaBg@mail.gmail.com>

Yes, I'll open a jira ticket.

Also, just wanted to correct something I wrote in my last email.

Where I said:

"If these are present, then Android will treat the cert as a standard user
cert."

I meant

"If these are NOT present, then Android will treat the cert as a standard
user cert."


On Tue, Aug 13, 2013 at 12:50 AM, Michael Jerris <mike at jerris.com> wrote:

> This sounds like it should be in the script for everyone.  Can you open a
> bug on jira.freeswitch.org for this issue.
>
> Thanks
> Mike
>
> On Aug 6, 2013, at 2:16 AM, Peter <eidevm5 at gmail.com> wrote:
>
> Finally figured out the issue was related to the gentls_cert script was
> generating an openssl template that didn't have the required x509v3
> extensions set.
>
> I modified the script where it generates config.tpl to add
>
>                         x509_extensions = v3_ca
>
> to the [req] section, then I added the section:
>
>                         [ v3_ca ]
>                         subjectKeyIdentifier=hash
>                         authorityKeyIdentifier=keyid:always,issuer
>                         basicConstraints=CA:TRUE
>
> Now when you issue:
>
> openssl x509 -noout -inform pem -text -in cafile.pem
>
> you'll see the following section:
>
>         X509v3 extensions:
>             X509v3 Subject Key Identifier:
>                 02:0A:A8:D0:5C:23:7C:8B:C4:EF:79:11:C7:0C:A8:86:71:15:59:D5
>             X509v3 Authority Key Identifier:
>
> keyid:02:0A:A8:D0:5C:23:7C:8B:C4:EF:79:11:C7:0C:A8:86:71:15:59:D5
>
>             X509v3 Basic Constraints:
>                 CA:TRUE
>
> If these are present, then Android will treat the cert as a standard user
> cert.
>
> Then it was a simple matter of copying cafile.pem to cafile.crt on the
> sdcard on the Android device and using the "install from device storage"
> option.
>
> When the cert installer dialog comes up, it will now detect cafile.crt as
> a CA cert and not user cert.
>
> Hope this helps other people, as cert management on Android is a right
> pain in the $#%^.
>
> Peter
>
>
>
> On Tue, Aug 6, 2013 at 2:31 PM, Peter <eidevm5 at gmail.com> wrote:
>
>> The reason I put it on a webserver is mostly for convenience to make it
>> easier to install.
>>
>> I tried copying cafile.pem to /sdcard on a Galaxy Note II, but when I try
>> the "Install from device storage" option, it just comes back with:
>>
>> "No certificate file found on SD card"
>>
>>
>>
>> On Mon, Aug 5, 2013 at 5:51 PM, Mehroz Ashraf <mehroz.ashraf85 at gmail.com>wrote:
>>
>>> Why do you want to place the cert on webserver and point android
>>> browser? If you are doing this to download cert into android then that is
>>> probably not the right approach.
>>>
>>> I used cafile.pem (without converting it into .der format) and placed
>>> the file in  SD card or phone memory, and point out linphone to get the CA
>>> from the path. You may search in libraries where it need to tell the path.
>>>
>>>
>>> On Mon, Aug 5, 2013 at 12:15 PM, Peter <eidevm5 at gmail.com> wrote:
>>>
>>>> Has anyone managed to get TLS working between Android Linphone and
>>>> Freeswitch?
>>>>
>>>> I've done the basic TLS setup as per
>>>> https://wiki.freeswitch.org/wiki/Tls
>>>>
>>>> I then convert the CA cert from PEM to DER format with:
>>>>
>>>> openssl x509  -inform PEM -outform der -in cafile.pem -out fs.crt
>>>>
>>>> I place fs.crt on a webserver and point my Android browser to it.
>>>>
>>>> When I click on fs.crt, I get the default Android Certificate installer
>>>> popup, but it always says:
>>>>
>>>> "Package contains: one user certificate"
>>>>
>>>> ie: it thinks it is a user cert rather than a CA cert.
>>>>
>>>> Android appears to be a real pain to add a CA to its trusted credential
>>>> store.
>>>>
>>>> Really interested if anyone has managed to get Android to import the CA
>>>> cert.
>>>>
>>>> Thanks
>>>>
>>>> Peter
>>>>
>>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nreis at wavecom.pt  Tue Aug 13 04:52:45 2013
From: nreis at wavecom.pt (Nuno Reis)
Date: Tue, 13 Aug 2013 01:52:45 +0100
Subject: [Freeswitch-users] Recommended ULIMIT settings
Message-ID: <CAE2+g++iHtJH4q5jkVoL5y_6W6kxqDsoB848XC9Xg7_fLqyDtQ@mail.gmail.com>

Hi guys.

I'm currently using the recommended ulimit settings as defined here:

http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations

Although there's at least one parameter that doesn't seem to be right when
i do a status at fs_cli:

UP 0 years, 0 days, 10 hours, 25 minutes, 14 seconds, 617 milliseconds, 626
microseconds
FreeSWITCH is ready
55390 session(s) since startup
0 session(s) 0/100
5000 session(s) max
min idle cpu 0.00/98.00
*Current Stack Size/Max 240K/240K*

The stack size should be 8192 as defined, but FS shows otherwise.

I'm using centOS x86_64 and a 64 bit version of FS.
Is this normal?
Thanks.

--

*Nuno Miguel Reis* | *Unified Communication** Systems*
M. +351 913907481 | nreis at wavecom.pt
WAVECOM-Solu??es R?dio, S.A.
Cacia Park | Rua do Progresso, Lote 15
3800-639 AVEIRO | Portugal
T. +351 309 700 225 | F. +351 234 919 191
*GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
www.wavecom.pt** <http://www.wavecom.pt/>*

[image: Description: Description:
WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php>

[image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>
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From krice at freeswitch.org  Tue Aug 13 05:18:14 2013
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 12 Aug 2013 20:18:14 -0500
Subject: [Freeswitch-users] Recommended ULIMIT settings
In-Reply-To: <CAE2+g++iHtJH4q5jkVoL5y_6W6kxqDsoB848XC9Xg7_fLqyDtQ@mail.gmail.com>
References: <CAE2+g++iHtJH4q5jkVoL5y_6W6kxqDsoB848XC9Xg7_fLqyDtQ@mail.gmail.com>
Message-ID: <C958FB5F-04D1-433A-815B-AA2D6D2AEA51@freeswitch.org>

yes this is normal, if fs can auto adjust it to 240 it will... you dont need > that for the thread stack anyway

Ken
Sent from my iPad

On Aug 12, 2013, at 19:52, Nuno Reis <nreis at wavecom.pt> wrote:

> Hi guys.
> 
> I'm currently using the recommended ulimit settings as defined here:
> 
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
> 
> Although there's at least one parameter that doesn't seem to be right when i do a status at fs_cli:
> 
> UP 0 years, 0 days, 10 hours, 25 minutes, 14 seconds, 617 milliseconds, 626 microseconds
> FreeSWITCH is ready
> 55390 session(s) since startup
> 0 session(s) 0/100
> 5000 session(s) max
> min idle cpu 0.00/98.00
> Current Stack Size/Max 240K/240K
> 
> The stack size should be 8192 as defined, but FS shows otherwise.
> 
> I'm using centOS x86_64 and a 64 bit version of FS.
> Is this normal?
> Thanks.
> 
> --
> 
> Nuno Miguel Reis | Unified Communication Systems
> M. +351 913907481 | nreis at wavecom.pt
> 
> WAVECOM-Solu??es R?dio, S.A.
> Cacia Park | Rua do Progresso, Lote 15
> 3800-639 AVEIRO | Portugal
> T. +351 309 700 225 | F. +351 234 919 191
> GPS | www.wavecom.pt
> 
> <image001.png>
> 
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jaybinks at gmail.com  Tue Aug 13 06:13:35 2013
From: jaybinks at gmail.com (jay binks)
Date: Tue, 13 Aug 2013 12:13:35 +1000
Subject: [Freeswitch-users] Recommended ULIMIT settings
In-Reply-To: <C958FB5F-04D1-433A-815B-AA2D6D2AEA51@freeswitch.org>
References: <CAE2+g++iHtJH4q5jkVoL5y_6W6kxqDsoB848XC9Xg7_fLqyDtQ@mail.gmail.com>
	<C958FB5F-04D1-433A-815B-AA2D6D2AEA51@freeswitch.org>
Message-ID: <CAO47HpffL-JkE=K8k73H3C-bRWtkope+cy53V+GoGRq5xqVbwA@mail.gmail.com>

So Ken,

we should change
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
to be :

ulimit -s 240


Just want to be 100% sure before changing it .

Jay





On 13 August 2013 11:18, Ken Rice <krice at freeswitch.org> wrote:

> yes this is normal, if fs can auto adjust it to 240 it will... you dont
> need > that for the thread stack anyway
>
> Ken
> Sent from my iPad
>
> On Aug 12, 2013, at 19:52, Nuno Reis <nreis at wavecom.pt> wrote:
>
> Hi guys.
>
> I'm currently using the recommended ulimit settings as defined here:
>
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
>
> Although there's at least one parameter that doesn't seem to be right when
> i do a status at fs_cli:
>
> UP 0 years, 0 days, 10 hours, 25 minutes, 14 seconds, 617 milliseconds,
> 626 microseconds
> FreeSWITCH is ready
> 55390 session(s) since startup
> 0 session(s) 0/100
> 5000 session(s) max
> min idle cpu 0.00/98.00
> *Current Stack Size/Max 240K/240K*
>
> The stack size should be 8192 as defined, but FS shows otherwise.
>
> I'm using centOS x86_64 and a 64 bit version of FS.
> Is this normal?
> Thanks.
>
> --
>
> *Nuno Miguel Reis* | *Unified Communication** Systems*
> M. +351 913907481 | nreis at wavecom.pt
> WAVECOM-Solu??es R?dio, S.A.
> Cacia Park | Rua do Progresso, Lote 15
> 3800-639 AVEIRO | Portugal
> T. +351 309 700 225 | F. +351 234 919 191
> *GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
> www.wavecom.pt** <http://www.wavecom.pt/>*
>
> <image001.png> <http://www.wavecom.pt/pt/wavecom/premios.php>
>
> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Sincerely

Jay
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From eidevm5 at gmail.com  Tue Aug 13 06:26:58 2013
From: eidevm5 at gmail.com (Peter)
Date: Tue, 13 Aug 2013 12:26:58 +1000
Subject: [Freeswitch-users] Establishing SRTP from SBC to endpoint
Message-ID: <CABkS=xqD0mukebja+w3aurrhcUJ4Gp=DcOpkh+BDnCpOKr32CA@mail.gmail.com>

In my environment, I have the following (simplified) setup:

FS1  ----  FS SBC ---  FS2

Phones registered to FS1 (100x) use TLS/SRTP and phones registered to FS2
(200x) use SIP/RTP

FS1 has inbound-bypass-media set to true to allow SRTP peer to peer and
direct to the SBC.

If I make an inbound call (eg: 1000 to 2000), SRTP is correctly established
between the phone and SBC with RTP on the other side of the SBC to the
internal phone.

However, when I try it the other way, I can't get SRTP established from the
SBC to the external phone.

I've been using https://wiki.freeswitch.org/wiki/Secure_RTP as a guide.

I've even tried explicitly setting sip_secure_media to true on the SBC and
FS1.

The dialplan on the SBC has:

  <extension name="outgoing">
        <condition field="destination_number" expression="^(10[0-9][0-9])$">
            <action application="set" data="sip_secure_media=true"/>
            <action application="bridge" data="sofia/external/${
destination_number}@10.1.1.204"/>
        </condition>
  </extension>


And on FS1, the dialplan has:

   <extension name="Local-Numbers">
      <condition field="destination_number" expression="^(10[01][0-9])$">
        <action application="export" data="dialed_extension=$1"/>
        <action application="set" data="sip_secure_media=true"/>
        <action application="bridge" data="user/${dialed_extension}@
${domain_name}"/>
      </condition>
    </extension>


Note that I've been testing this against two phones with SRTP enabled, but
only one that is using TLS.  I get the same result calling each phone.

On a related point, what it the step required for a TLS connection from the
SBC to the phone?   I'm assume the phone just needs the CA cert from the
SBC.  Correct?

Any information as to where I'm going wrong will be gratefully accepted.

Thanks

Peter
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From krice at freeswitch.org  Tue Aug 13 06:28:05 2013
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 12 Aug 2013 21:28:05 -0500
Subject: [Freeswitch-users] Recommended ULIMIT settings
In-Reply-To: <CAO47HpffL-JkE=K8k73H3C-bRWtkope+cy53V+GoGRq5xqVbwA@mail.gmail.com>
References: <CAE2+g++iHtJH4q5jkVoL5y_6W6kxqDsoB848XC9Xg7_fLqyDtQ@mail.gmail.com>
	<C958FB5F-04D1-433A-815B-AA2D6D2AEA51@freeswitch.org>
	<CAO47HpffL-JkE=K8k73H3C-bRWtkope+cy53V+GoGRq5xqVbwA@mail.gmail.com>
Message-ID: <EFDA627B-703B-4FD2-91AC-68ED5F343FE0@freeswitch.org>

well if you look at the wiki history there seems to be a fight over what to set it to, i always set it to 240 cause it makes that error msg anout stack > 240 at startup go away

Ken
Sent from my iPad

On Aug 12, 2013, at 21:13, jay binks <jaybinks at gmail.com> wrote:

> So Ken,
> 
> we should change http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
> to be :
> ulimit -s 240    
> 
> Just want to be 100% sure before changing it .
> 
> Jay
> 
> 
> 
> 
> 
> On 13 August 2013 11:18, Ken Rice <krice at freeswitch.org> wrote:
>> yes this is normal, if fs can auto adjust it to 240 it will... you dont need > that for the thread stack anyway
>> 
>> Ken
>> Sent from my iPad
>> 
>> On Aug 12, 2013, at 19:52, Nuno Reis <nreis at wavecom.pt> wrote:
>> 
>>> Hi guys.
>>> 
>>> I'm currently using the recommended ulimit settings as defined here:
>>> 
>>> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
>>> 
>>> Although there's at least one parameter that doesn't seem to be right when i do a status at fs_cli:
>>> 
>>> UP 0 years, 0 days, 10 hours, 25 minutes, 14 seconds, 617 milliseconds, 626 microseconds
>>> FreeSWITCH is ready
>>> 55390 session(s) since startup
>>> 0 session(s) 0/100
>>> 5000 session(s) max
>>> min idle cpu 0.00/98.00
>>> Current Stack Size/Max 240K/240K
>>> 
>>> The stack size should be 8192 as defined, but FS shows otherwise.
>>> 
>>> I'm using centOS x86_64 and a 64 bit version of FS.
>>> Is this normal?
>>> Thanks.
>>> 
>>> --
>>> 
>>> Nuno Miguel Reis | Unified Communication Systems
>>> M. +351 913907481 | nreis at wavecom.pt
>>> 
>>> WAVECOM-Solu??es R?dio, S.A.
>>> Cacia Park | Rua do Progresso, Lote 15
>>> 3800-639 AVEIRO | Portugal
>>> T. +351 309 700 225 | F. +351 234 919 191
>>> GPS | www.wavecom.pt
>>> 
>>> <image001.png>
>>> 
>>> 
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> 
> 
> -- 
> Sincerely
> 
> Jay
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 zhulizhong at live.com  Tue Aug 13 06:42:49 2013
From: zhulizhong at live.com (James zhu)
Date: Tue, 13 Aug 2013 02:42:49 +0000
Subject: [Freeswitch-users] Can not make out going call from sangoma A200
In-Reply-To: <BLU171-W755EFA24D8B38183EE150AB55E0@phx.gbl>
References: <BLU171-W3672C8625546F122FFC013B5510@phx.gbl>,
	, , <CAHgk4BXX1nArMXv-T5CNzLbjh_wyn0fjzYNOUCaAPsUE14KNdA@mail.gmail.com>,
	, , <BLU171-W58FF5C141AECC333E5BD0FB5520@phx.gbl>,
	, , <CAHgk4BVdu6Dk3P=rrRBKquiydLUUGBUxSHauTv73tx=XPq=pjA@mail.gmail.com>,
	, <BLU171-W488ED06FE9240BE8CD3149B55D0@phx.gbl>,
	<BLU171-W755EFA24D8B38183EE150AB55E0@phx.gbl>
Message-ID: <BLU171-W1BA48FE2C8874BEBCE339B5440@phx.gbl>

hello:I made a test with Asterisk, wanpipe and dahdi. the both of outgoing and incoming calls worked. Only the problem is existing by using freeTDM and FreeSWITCH. Under freeTDMand FreeSWITCH, we can not make outing calls. the system logs show:2013-08-02 14:36:42.003531 [NOTICE] mod_freetdm.c:2273 Hangup FreeTDM/2:1/13811737925 [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION]freeswitch at internal> -------------------------------------------------------I think it should be a bug with freeTDM. 
Best regards,

James.zhu

website: www.hiastar.com

From: zhulizhong at live.com
To: freeswitch-users at lists.freeswitch.org
Date: Wed, 7 Aug 2013 01:52:15 +0000
Subject: Re: [Freeswitch-users] Can not make out going call from sangoma A200




thanks for the support. this is log:013-08-02 14:36:36.943647 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/1008 at 192.168.0.173) State ROUTING going to sleepfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/1008 at 192.168.0.173) Running State Change CS_EXECUTEfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/1008 at 192.168.0.173) State EXECUTEfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] mod_sofia.c:230 sofia/internal/1008 at 192.168.0.173 SOFIA EXECUTEfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] switch_core_state_machine.c:209 sofia/internal/1008 at 192.168.0.173 Standard EXECUTEfreeswitch at internal> 
EXECUTE sofia/internal/1008 at 192.168.0.173 set(open=true)freeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] mod_dptools.c:1393 sofia/internal/1008 at 192.168.0.173 SET [open]=[true]freeswitch at internal> 
EXECUTE sofia/internal/1008 at 192.168.0.173 bridge(freetdm/2/1/13811737925)freeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] switch_ivr_originate.c:2060 Parsing global variablesfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] mod_freetdm.c:403 Set codec PCMA 20msfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] mod_freetdm.c:1242 Connect outbound channel FreeTDM/2:1/13811737925freeswitch at internal> 
2013-08-02 14:36:36.943647 [NOTICE] switch_channel.c:1030 New Channel FreeTDM/2:1/13811737925 [db274fb7-a395-4f29-8ff1-5a6634f01c56]freeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] mod_freetdm.c:1251 (FreeTDM/2:1/13811737925) State Change CS_NEW -> CS_INITfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]freeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] mod_freetdm.c:1270 Attached session db274fb7-a395-4f29-8ff1-5a6634f01c56 to channel 2:1freeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] ftmod_analog.c:62 [s2c1][1:3] Changed state from DOWN to DIALINGfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] ftmod_analog.c:437 [s2c1][1:3] ANALOG CHANNEL thread starting.freeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] ftdm_io.c:3381 [s2c1][1:3] Enabled software DTMF detectorfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] ftmod_analog.c:457 [s2c1][1:3] Initialized DTMF detectionfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] ftmod_analog.c:638 [s2c1][1:3] Completed state change from DOWN to DIALING in 1 msfreeswitch at internal> 
2013-08-02 14:36:36.943647 [DEBUG] ftmod_analog.c:644 [s2c1][1:3] Executing state handler on 2:1 for DIALINGfreeswitch at internal> 
2013-08-02 14:36:36.963647 [DEBUG] ftmod_wanpipe.c:1002 [s2c1][1:3] First packet read stats: Rx queue len: 0, Rx queue size: 10freeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925) Running State Change CS_INITfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:455 (FreeTDM/2:1/13811737925) State INITfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] mod_freetdm.c:430 (FreeTDM/2:1/13811737925) State Change CS_INIT -> CS_ROUTINGfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]freeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:455 (FreeTDM/2:1/13811737925) State INIT going to sleepfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925) Running State Change CS_ROUTINGfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:471 (FreeTDM/2:1/13811737925) State ROUTINGfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] mod_freetdm.c:453 FreeTDM/2:1/13811737925 CHANNEL ROUTINGfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_ivr_originate.c:67 (FreeTDM/2:1/13811737925) State Change CS_ROUTING -> CS_CONSUME_MEDIAfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]freeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:471 (FreeTDM/2:1/13811737925) State ROUTING going to sleepfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925) Running State Change CS_CONSUME_MEDIAfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:490 (FreeTDM/2:1/13811737925) State CONSUME_MEDIAfreeswitch at internal> 
2013-08-02 14:36:37.443634 [DEBUG] switch_core_state_machine.c:490 (FreeTDM/2:1/13811737925) State CONSUME_MEDIA going to sleepfreeswitch at internal> 
2013-08-02 14:36:41.963531 [DEBUG] ftmod_analog.c:503 [s2c1][1:3] Changed state from DIALING to BUSYfreeswitch at internal> 
2013-08-02 14:36:41.983901 [DEBUG] ftmod_analog.c:638 [s2c1][1:3] Completed state change from DIALING to BUSY in 20 msfreeswitch at internal> 
2013-08-02 14:36:41.983901 [DEBUG] ftmod_analog.c:644 [s2c1][1:3] Executing state handler on 2:1 for BUSYfreeswitch at internal> 
2013-08-02 14:36:41.983901 [DEBUG] ftmod_analog.c:803 [s2c1][1:3] Changed state from BUSY to DOWNfreeswitch at internal> 
2013-08-02 14:36:42.003531 [DEBUG] ftmod_analog.c:638 [s2c1][1:3] Completed state change from BUSY to DOWN in 20 msfreeswitch at internal> 
2013-08-02 14:36:42.003531 [DEBUG] ftmod_analog.c:644 [s2c1][1:3] Executing state handler on 2:1 for DOWNfreeswitch at internal> 
2013-08-02 14:36:42.003531 [DEBUG] mod_freetdm.c:2253 got FXO sig 2:1 [STOP]freeswitch at internal> 
2013-08-02 14:36:42.003531 [NOTICE] mod_freetdm.c:2273 Hangup FreeTDM/2:1/13811737925 [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION]freeswitch at internal> 
2013-08-02 14:36:42.003531 [DEBUG] switch_channel.c:3135 Send signal FreeTDM/2:1/13811737925 [KILL]freeswitch at internal> 
2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]freeswitch at internal> 
freeswitch at internal> -----------------------------------------------------------------------the port is working for inbound calls. 

Best regards,

James.zhu

website: www.hiastar.com

From: zhulizhong at live.com
To: freeswitch-users at lists.freeswitch.org
Date: Tue, 6 Aug 2013 06:42:14 +0000
Subject: Re: [Freeswitch-users] Can not make out going call from sangoma A200




hello:I contacted Sangoma, but I do not have any feedback yet. I think it might be a bugfor analog card with freeSWITCH.  freetdm  partially works with sangoma A200. There are two issues:1) Can not get callerid from IO:  only shows this:Initializing cid data! the callerid is emptyeven I set to CN.2) Can not make outgoing calls by the port that I used to make outgoing calls. even I replaced a FXO module, the problem is same. I use same syntax as follow:http://blog.hiastar.com/?p=276 Hope someone from the FreeSWITCH community to clarify that. 
Best regards,

James.zhu

website: www.hiastar.com

Date: Fri, 2 Aug 2013 23:03:47 -0400
From: bdfoster at davri.com
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Can not make out going call from sangoma A200

I apologize, I misread your email. Hopefully someone who knows FreeTDM will come around. Have you contacted Sangoma? They are the maintainers of FreeTDM as far as I know. You might try that route. They are very familiar with freeswitch im sure :).

Thank you,
Brian Foster

Project Manager/Owner's Rep.

Davri Investments, Inc.

O: 317-787-2686 x2102

M: 317-600-9753

E: bdfoster at davri.com

Indianapolis, Indiana
Sent from a mobile device.
On Aug 2, 2013 10:39 PM, "James zhu" <zhulizhong at live.com> wrote:




thanks, Brian.  actually the 2:1 is idle because the port can accept incoming call. afterI reload freetdm or restart FS, I still can not make outgoing calls. If the port physicallyfailed, how come I can make incoming call use the same port. confused. thanks again.

Best regards,

James.zhu

website: www.hiastar.com

Date: Fri, 2 Aug 2013 14:22:29 -0400
From: bdfoster at davri.com
To: freeswitch-users at lists.freeswitch.org

Subject: Re: [Freeswitch-users] Can not make out going call from sangoma A200

That's the expected behavior if 2:1 is in use. So that means you use another port to place your outbound call. Tips on how to do that, however, I can't really produce. Hopefully someone with more experience with FreeTDM can chime in.


Thank you,
Brian Foster

Project Manager/Owner's Rep.

Davri Investments, Inc.

O: 317-787-2686 x2102

M: 317-600-9753

E: bdfoster at davri.com

Indianapolis, Indiana
Sent from a mobile device.
On Aug 2, 2013 3:09 AM, "James zhu" <zhulizhong at live.com> wrote:




hello:I installed freeswitch and sangoma A200 with 2 FXOs. i set the tone group to CNand opermode=china. the A200 can make accept incoming calls from FreeTDM/2:1 , but I use same port to make out going port, the shows the port is CONGESTION.

--------------------------system log-----------------------------------------------2013-08-02 14:36:42.003531 [DEBUG] mod_freetdm.c:640 [2:1] FreeTDM/2:1/13811737925 CHANNEL HANGUP EXIT

2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:48 FreeTDM/2:1/13811737925 Standard HANGUP, cause: NORMAL_CIRCUIT_CONGESTION2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:679 (FreeTDM/2:1/13811737925) State HANGUP going to sleep

2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:692 (FreeTDM/2:1/13811737925) Callstate Change DOWN -> HANGUP2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:447 (FreeTDM/2:1/13811737925) State Change CS_HANGUP -> CS_REPORTING

2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:416 (FreeTDM/2:1/13811737925) Running State Change CS_REPORTING

2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State REPORTING2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:92 FreeTDM/2:1/13811737925 Standard REPORTING, cause: NORMAL_CIRCUIT_CONGESTION

2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:764 (FreeTDM/2:1/13811737925) State REPORTING going to sleep2013-08-02 14:36:42.003531 [DEBUG] switch_core_state_machine.c:441 (FreeTDM/2:1/13811737925) State Change CS_REPORTING -> CS_DESTROY

2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1341 Send signal FreeTDM/2:1/13811737925 [BREAK]2013-08-02 14:36:42.003531 [DEBUG] switch_core_session.c:1549 Session 37 (FreeTDM/2:1/13811737925) Locked, Waiting on external entities

2013-08-02 14:36:42.023531 [DEBUG] switch_ivr_originate.c:3632 Originate Resulted in Error Cause: 34 [NORMAL_CIRCUIT_CONGESTION]2013-08-02 14:36:42.023531 [INFO] mod_dptools.c:3192 Originate Failed.  Cause: NORMAL_CIRCUIT_CONGESTION

2013-08-02 14:36:42.023531 [NOTICE] switch_channel.c:4593 Hangup sofia/internal/1008 at 192.168.0.173 [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION]2013-08-02 14:36:42.023531 [DEBUG] switch_channel.c:3135 Send signal sofia/internal/1008 at 192.168.0.173 [KILL]

2013-08-02 14:36:42.023531 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal---------------------------------------------the FS is download from git and wanpipe is latest version.  please give a help for that issue.


Best regards,

James.zhu

website: www.hiastar.com 		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.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




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 		 	   		  

_________________________________________________________________________

Professional FreeSWITCH Consulting Services:

consulting at freeswitch.org

http://www.freeswitchsolutions.com









Official FreeSWITCH Sites

http://www.freeswitch.org

http://wiki.freeswitch.org

http://www.cluecon.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




_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 		 	   		  

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 		 	   		  

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

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From jaybinks at gmail.com  Tue Aug 13 06:53:01 2013
From: jaybinks at gmail.com (jay binks)
Date: Tue, 13 Aug 2013 12:53:01 +1000
Subject: [Freeswitch-users] Recommended ULIMIT settings
In-Reply-To: <EFDA627B-703B-4FD2-91AC-68ED5F343FE0@freeswitch.org>
References: <CAE2+g++iHtJH4q5jkVoL5y_6W6kxqDsoB848XC9Xg7_fLqyDtQ@mail.gmail.com>
	<C958FB5F-04D1-433A-815B-AA2D6D2AEA51@freeswitch.org>
	<CAO47HpffL-JkE=K8k73H3C-bRWtkope+cy53V+GoGRq5xqVbwA@mail.gmail.com>
	<EFDA627B-703B-4FD2-91AC-68ED5F343FE0@freeswitch.org>
Message-ID: <CAO47HpejFCiJzA5oawdv5OmgGuAMAp0n3+-J=+Y5p8_fyZgMiA@mail.gmail.com>

can we just get a concensus from the people who know ?? :)

Anthm, MikeJ, BWK ..

it sounds like ulimit -s 240 is the way to go, right ?




On 13 August 2013 12:28, Ken Rice <krice at freeswitch.org> wrote:

> well if you look at the wiki history there seems to be a fight over what
> to set it to, i always set it to 240 cause it makes that error msg anout
> stack > 240 at startup go away
>
>
> Ken
> Sent from my iPad
>
> On Aug 12, 2013, at 21:13, jay binks <jaybinks at gmail.com> wrote:
>
> So Ken,
>
> we should change
> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
>  to be :
>
> ulimit -s 240
>
>
> Just want to be 100% sure before changing it .
>
> Jay
>
>
>
>
>
> On 13 August 2013 11:18, Ken Rice <krice at freeswitch.org> wrote:
>
>> yes this is normal, if fs can auto adjust it to 240 it will... you dont
>> need > that for the thread stack anyway
>>
>> Ken
>> Sent from my iPad
>>
>> On Aug 12, 2013, at 19:52, Nuno Reis <nreis at wavecom.pt> wrote:
>>
>> Hi guys.
>>
>> I'm currently using the recommended ulimit settings as defined here:
>>
>> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
>>
>> Although there's at least one parameter that doesn't seem to be right
>> when i do a status at fs_cli:
>>
>> UP 0 years, 0 days, 10 hours, 25 minutes, 14 seconds, 617 milliseconds,
>> 626 microseconds
>> FreeSWITCH is ready
>> 55390 session(s) since startup
>> 0 session(s) 0/100
>> 5000 session(s) max
>> min idle cpu 0.00/98.00
>> *Current Stack Size/Max 240K/240K*
>>
>> The stack size should be 8192 as defined, but FS shows otherwise.
>>
>> I'm using centOS x86_64 and a 64 bit version of FS.
>> Is this normal?
>> Thanks.
>>
>> --
>>
>> *Nuno Miguel Reis* | *Unified Communication** Systems*
>> M. +351 913907481 | nreis at wavecom.pt
>> WAVECOM-Solu??es R?dio, S.A.
>> Cacia Park | Rua do Progresso, Lote 15
>> 3800-639 AVEIRO | Portugal
>> T. +351 309 700 225 | F. +351 234 919 191
>> *GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
>> www.wavecom.pt** <http://www.wavecom.pt/>*
>>
>> <image001.png> <http://www.wavecom.pt/pt/wavecom/premios.php>
>>
>> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Sincerely
>
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Sincerely

Jay
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From k4kaleem at gmail.com  Tue Aug 13 10:07:06 2013
From: k4kaleem at gmail.com (Kaleem)
Date: Tue, 13 Aug 2013 07:07:06 +0100
Subject: [Freeswitch-users] Securing FS
In-Reply-To: <mailman.22524.1376353160.576.freeswitch-users@lists.freeswitch.org>
References: <mailman.22524.1376353160.576.freeswitch-users@lists.freeswitch.org>
Message-ID: <2A78AF6C-7D80-416A-BEF1-E76463B82F28@gmail.com>

Hi all,

Just a quick question

I need to secure my FS, I can think of 2 ways. Please advise of best option and also how to setup 2nd for learning

1: have FS behind IPTables FW and use access list on which ip can talk to FS and only allow providers.

2: setup another FS as SBC and have it that only internal FS IP can generate calls (no external users, if Ever get any, probably get them to VPN in.

Thanks in advance
Regards,
K
Sent from my iPhone

From ivan at c3i.bg  Tue Aug 13 10:37:20 2013
From: ivan at c3i.bg (Ivan)
Date: Tue, 13 Aug 2013 09:37:20 +0300
Subject: [Freeswitch-users] blocking IP addresses and fail2ban setup
In-Reply-To: <52093F3E.6040506@xtronics.com>
References: <10175.1376191370@ccs.covici.com>
	<58C685B3-6E78-4A42-9F2D-8559DB8F7228@jerris.com>
	<52093F3E.6040506@xtronics.com>
Message-ID: <5209D420.1070706@c3i.bg>

> Huge lists blocked as dynamic will reduce performance of your firewall.

FWIW, one of my firewalls (a rather low-end machine by today's 
standards) sees a hundred of Go of daily traffic, most of it hitting a 
standard iptables chain with 4000+ blacklist rules before being 
accepted, and the load average / CPU usage are close to 0. The rules are 
a mix of source ips/nets, protocol and packet size checks, etc.

In comparison, Brian's blacklist is only 142 lines so ipset is not 
really needed for performance (but of course, if it's packaged for your 
distro why not use it - in my case it was not and I was lazy 
compiling/maintaining it).

BTW a way to flush/repopulate a chain *very* quickly is to create a file 
in the iptables-restore format and restore it:

LIST=/path/to/blacklist
RULES=/path/to/some/rules
echo "*filter" > $RULES
echo ":blacklist - [0:0]" >> $RULES
for i in $(cat "$LIST"); do
	echo "-A blacklist -s $i -j DROP" >>  $RULES
done
echo "COMMIT" >> $RULES
/sbin/iptables-restore --noflush $RULES




From ivan at c3i.bg  Tue Aug 13 10:57:50 2013
From: ivan at c3i.bg (Ivan)
Date: Tue, 13 Aug 2013 09:57:50 +0300
Subject: [Freeswitch-users] garbled audio with G726-32,
 other codecs are fine
In-Reply-To: <065260E6-8DAB-4273-8C5A-0D0A48BD175B@jerris.com>
References: <51FD0E8B.3060702@c3i.bg>
	<CAHgk4BVNA+NOg5ChyYxSddDxhMxgSmWVe1tQ7NiCPRKo6M5BCw@mail.gmail.com>
	<003901ce90b9$749f0090$5ddd01b0$@v10networks.ca>
	<51FDEC5B.2010307@c3i.bg>
	<065260E6-8DAB-4273-8C5A-0D0A48BD175B@jerris.com>
Message-ID: <5209D8EE.4090903@c3i.bg>

Ah, I didn't know there was always bit packing - I guess I should read 
how G726 works, thanks for pointing that.

The thing is, both linksys PAPs and linphone have the same problem, and 
they are not "exotic" endpoints so that makes me think the cause is my 
freeswitch setup. I'll try to test with Xlite when I have a chance, and 
also see if wireshark shows enough codec detail to find out if the 
endpoints get it wrong. I'm of course interested if you have other ideas 
on how to debug that.

ivan


On 08/12/2013 05:14 PM, Michael Jerris wrote:
> There is always bit packing, but there are 2 different ways to do the bit packing.  A lot of devices get it wrong so its worth looking at that.
>
> Mike
>
> On Aug 4, 2013, at 1:53 AM, Ivan Mitev <imitev at c3i.bg> wrote:
>
>> Thanks for the suggestion but I'm testing with G726-32, not AAL2-G726-32
>> ; so bitpacking shouldn't be used. By the way when I tested with AAL2 to
>> the linphone client I only got cracks and whitenoise, I've forgot to
>> mention that in my post.
>>
>> That said I've tried to uncomment and set <param name="bitpacking"
>> value="none"/> in internal.xml ("none" is a wild guess - I couldn't find
>> any doc on values accepted by this parameter), but that doesn't help.
>>
>> Speex: the ATAs don't support it. And being stubborn I'd like to
>> understand what's wrong with G726 :)
>>
>>
>> On 08/04/2013 05:22 AM, Jeff Leung wrote:
>>>
>>> You can turn off G726 AAC bit-packing in spandsp.conf.xml.
>>>
>>> By the way, there are other codecs out there you can try. SPEEX comes
>>> to mind if all your endpoints don?t deal with the PSTN.
>>>
>>> *From:*freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of
>>> *Brian Foster
>>> *Sent:* Saturday, August 3, 2013 2:37 PM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] garbled audio with G726-32, other
>>> codecs are fine
>>>
>>> AAC bitpacking by any chance? I thought I had a similar issue,
>>> happened so long ago I cant remember what I did.
>>>
>>> Thank you,
>>>
>>> Brian Foster
>>> Project Manager/Owner's Rep.
>>> Davri Investments, Inc.
>>> O: 317-787-2686 x2102
>>> M: 317-600-9753
>>> E: bdfoster at davri.com <mailto:bdfoster at davri.com>
>>> Indianapolis, Indiana
>>>
>>> Sent from a mobile device.
>>>
>>> On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev at c3i.bg
>>> <mailto:imitev at c3i.bg>> wrote:
>>>
>>> Hello
>>>
>>> I'm migrating an office setup from asterisk to FS and in the process I
>>> was considering using G726-32 for some bandwidth starved remote
>>> endpoints. However I only get metallic/garbled audio with that codec
>>> even when simply playing moh to the endpoint, while other codecs work
>>> fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds
>>> marginally better but still garbled and really worse than G711.
>>>
>>> The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest
>>> (centos6 64bit host). But please don't shoot ! :) - I know about virtual
>>> environment limitations but for these tests the host is only lightly
>>> loaded, there aren't any calls to the FS instance except my tests, and
>>> the fact that it works with other codecs makes me think that
>>> virtualization is not the issue here. I may be wrong though.
>>>
>>> Is there any guide for debugging that kind of problem before reverting
>>> to a fresh install on bare-metal with the latest HEAD ? Until now I've
>>> tried:
>>>
>>> - improving timers ; but the default soft timer (which I guess uses
>>> timerd) works best. The time interval between sent packets on a tcpdump
>>> trace looks identical to the output of "timer_test", so that doesn't
>>> seem to be a network/jitter problem. And there's no problem with other
>>> codecs, but maybe G726-XX is specific. For info the guest's clocksource
>>> is kvm_clock, while the host uses tsc.
>>>
>>> - using different endpoints: the production ones are Linksys PAP2
>>> ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the
>>> same thing happens with linphone on a fedora 19 laptop.
>>>
>>> A call with rtp media going through FS without transcoding - G726-32 to
>>> G726-32 - works perfectly (I can't hear the difference with G711). The
>>> problem is only when there's transcoding to G726 (from wav for moh, or
>>> from any other codec when bridging). I've looked at the wiki, posts,
>>> changelogs, jira, ..., but am a bit at a loss now.
>>>
>>> Any pointers ?
>>>
>>> Except that little problem, FS rocks, and I'm happy I can finally ditch
>>> asterisk. Kudos to the core devs and contributors.
>>>
>>> Ivan
>>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Tue Aug 13 11:13:20 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Tue, 13 Aug 2013 12:43:20 +0530
Subject: [Freeswitch-users] Softphone not getting connected
Message-ID: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>

I installed freeswitch on my ubuntu 12.04 machine...but when i am trying to
connect the softphone installed on my windows 8 pc with ubuntu machine the
softphone gives me an error that account not enabled...i have used a
network cable to connect the two machines...also the firewall in both the
machines is disabled...
Kindly help..

Ashish Mishra
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From itsme.kunnu at gmail.com  Tue Aug 13 11:16:29 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Tue, 13 Aug 2013 12:46:29 +0530
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <5209575C.9030203@freeswitch.org>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<CAMvhjJtBONsXnrhRoDdoMTUPhHb1NjVRSJOsdbt5u+0tjqaBug@mail.gmail.com>
	<CAFiqYuneYGGujBH-a_hanzEQ72Rdiks5kfwntwz2QGp1YC9A7A@mail.gmail.com>
	<5209575C.9030203@freeswitch.org>
Message-ID: <CAMvhjJuZqjssQ896TBxAFKXUrkrNb=60EJ9kKdkOfWK4VD3Rsg@mail.gmail.com>

Thank you for your reply...i am now able to run fs_cli on my ubuntu machine
on which freeswitch is installed...:-)
On Aug 13, 2013 3:17 AM, "I put the Who? in Mishehu" <mishehu at freeswitch.org>
wrote:

>  I have the nagging feeling that Ashish isn't telling the whole story
> here...
>
> Ashish -  what happens if you are on the machine that is running
> FreeSWITCH and you execute `telnet localhost 8021` ?  Do you see something
> like this:
>
> # telnet localhost 8021
> Trying 127.0.0.1...
> Connected to localhost.
> Escape character is '^]'.
> Content-Type: auth/request
>
> If you are running with the standard configurations this should work in
> this fashion.  If you have changed the
> conf/autoload_configs/event_socket.conf.xml , then paste those
> configurations.  (If too lengthy, please post to
> http://pastebin.freeswitch.org )
>
> -Yossi
>
> On 08/12/2013 09:44 AM, Steven Ayre wrote:
>
> Have you checked the logfile and netstat?
>
>
> On 12 August 2013 09:51, Ashish Mishra <itsme.kunnu at gmail.com> wrote:
>
>> I reinstalled freeswitch but still i am getting the same error :
>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>> mod_socket is also loaded...i have also flushed the firewall rules...pls
>> help
>>  On Aug 6, 2013 2:30 AM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>>
>>> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which i
>>> have installed freeswitch) it gives me the following error :
>>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Tue Aug 13 11:19:38 2013
From: eidevm5 at gmail.com (Peter)
Date: Tue, 13 Aug 2013 17:19:38 +1000
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
Message-ID: <CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>

Have you turned on debugging from the FS cli and seen if any registration
requests come in?



On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> I installed freeswitch on my ubuntu 12.04 machine...but when i am trying
> to connect the softphone installed on my windows 8 pc with ubuntu machine
> the softphone gives me an error that account not enabled...i have used a
> network cable to connect the two machines...also the firewall in both the
> machines is disabled...
> Kindly help..
>
> Ashish Mishra
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Tue Aug 13 11:34:49 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Tue, 13 Aug 2013 13:04:49 +0530
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
Message-ID: <CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>

Thank you Peter...you mean to say that i should first turn on the fs_cli
and then retry to connect the softphone...??? I would also like to remind
you that fs_cli and freeswitch are running on the same machine and i am
trying to connect thru a network cable my windows 8 pc that has the
softphone on it...
Regards
Ashish Mishra
On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:

> Have you turned on debugging from the FS cli and seen if any registration
> requests come in?
>
>
>
> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> I installed freeswitch on my ubuntu 12.04 machine...but when i am trying
>> to connect the softphone installed on my windows 8 pc with ubuntu machine
>> the softphone gives me an error that account not enabled...i have used a
>> network cable to connect the two machines...also the firewall in both the
>> machines is disabled...
>> Kindly help..
>>
>> Ashish Mishra
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Tue Aug 13 11:50:39 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Tue, 13 Aug 2013 13:20:39 +0530
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
Message-ID: <CAMvhjJu8H0Y25Gd1qtJ0kEb9eiu_D56+rZVJy_zbHV7mEzMLcA@mail.gmail.com>

Hello Peter...i set the fs cli to debugging by typing /log 7 and then
retried to connect...but still the same result...
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From steveayre at gmail.com  Tue Aug 13 12:17:27 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 13 Aug 2013 09:17:27 +0100
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAMvhjJuZqjssQ896TBxAFKXUrkrNb=60EJ9kKdkOfWK4VD3Rsg@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<CAMvhjJtBONsXnrhRoDdoMTUPhHb1NjVRSJOsdbt5u+0tjqaBug@mail.gmail.com>
	<CAFiqYuneYGGujBH-a_hanzEQ72Rdiks5kfwntwz2QGp1YC9A7A@mail.gmail.com>
	<5209575C.9030203@freeswitch.org>
	<CAMvhjJuZqjssQ896TBxAFKXUrkrNb=60EJ9kKdkOfWK4VD3Rsg@mail.gmail.com>
Message-ID: <CAFiqYumwEGA7zYmJPpQ51w6q_1WsQARht5Z2Gyutoqa4KuRt2w@mail.gmail.com>

What was the issue?


On 13 August 2013 08:16, Ashish Mishra <itsme.kunnu at gmail.com> wrote:

> Thank you for your reply...i am now able to run fs_cli on my ubuntu
> machine on which freeswitch is installed...:-)
> On Aug 13, 2013 3:17 AM, "I put the Who? in Mishehu" <
> mishehu at freeswitch.org> wrote:
>
>>  I have the nagging feeling that Ashish isn't telling the whole story
>> here...
>>
>> Ashish -  what happens if you are on the machine that is running
>> FreeSWITCH and you execute `telnet localhost 8021` ?  Do you see something
>> like this:
>>
>> # telnet localhost 8021
>> Trying 127.0.0.1...
>> Connected to localhost.
>> Escape character is '^]'.
>> Content-Type: auth/request
>>
>> If you are running with the standard configurations this should work in
>> this fashion.  If you have changed the
>> conf/autoload_configs/event_socket.conf.xml , then paste those
>> configurations.  (If too lengthy, please post to
>> http://pastebin.freeswitch.org )
>>
>> -Yossi
>>
>> On 08/12/2013 09:44 AM, Steven Ayre wrote:
>>
>> Have you checked the logfile and netstat?
>>
>>
>> On 12 August 2013 09:51, Ashish Mishra <itsme.kunnu at gmail.com> wrote:
>>
>>> I reinstalled freeswitch but still i am getting the same error :
>>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>> mod_socket is also loaded...i have also flushed the firewall rules...pls
>>> help
>>>  On Aug 6, 2013 2:30 AM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>>>
>>>> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which
>>>> i have installed freeswitch) it gives me the following error :
>>>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jaybinks at gmail.com  Tue Aug 13 13:35:48 2013
From: jaybinks at gmail.com (jay binks)
Date: Tue, 13 Aug 2013 19:35:48 +1000
Subject: [Freeswitch-users] freeswitch and mediaproxy-ng
Message-ID: <CAO47HpeAk4Mbo6dGRfDbvwxcoHfGkn_-uRFxLuQ9ZD35HPeqZA@mail.gmail.com>

So.. when I stumbled across this : https://github.com/sipwise/mediaproxy-ng

I was quite interested in the kernel based RTP proxying and the efficiency
gained by reducing interrupts.

has anyone ever investigated the integration of this with Freeswitch ?
Im not even sure if FS's RTP stack is modular, but I thought id ask the
questions.


-- 
Sincerely

Jay
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From denis.gasparin at edistar.com  Tue Aug 13 14:01:07 2013
From: denis.gasparin at edistar.com (Denis Gasparin)
Date: Tue, 13 Aug 2013 12:01:07 +0200 (CEST)
Subject: [Freeswitch-users] Httapi playback: input-timeout
In-Reply-To: <264401995.3487.1376385647375.JavaMail.root@mailserver.edistar.com>
Message-ID: <1759312119.3559.1376388067659.JavaMail.root@mailserver.edistar.com>




Hi. 


I setup an ivr using httapi and I've got a strange behaviour with input-timeout and digit-timeout attributes. 


The ivr xml is: 


<?xml version="1.0"?> 
<document type="text/freeswitch-httapi"> 
<work> 
<playback file="ivr/ivr-please_enter_the_number_where_we_can_reach_you.wav" name="FluentIvrDigitInput" terminators="#" input-timeout="10000" digit-timeout="5000"> 
<bind action="--my_bind_action_url--">~^\d+#</bind> 
</playback> 
<continue action="--my_continue_action--"/> 
</work> 
</document> 


With no input, t he timeout triggers after 10s but I was expecting 5s according to digit-timeout. 
The relevant log of the call is: 



2013-08-13 11:46:10.315627 [DEBUG] mod_httapi.c:1217 Process Tag: [playback] 
2013-08-13 11:46:10.315627 [INFO] switch_ivr_async.c:212 Digit parser HTTAPI: Setting realm to 'default' 
2013-08-13 11:46:10.315627 [DEBUG] switch_ivr_async.c:318 Digit parser HTTAPI: binding ^\d+#/default/0 callback: 0x7fa1e8486bd0 data: 0x880478 
2013-08-13 11:46:10.315627 [INFO] switch_ivr_async.c:212 Digit parser HTTAPI: Setting realm to 'default' 
2013-08-13 11:46:10.315627 [INFO] switch_ivr_async.c:201 Digit parser HTTAPI: Setting terminators for realm 'default' to '#' 
2013-08-13 11:46:10.315627 [DEBUG] switch_ivr_play_say.c:1315 Codec Activated L16 at 8000hz 1 channels 20ms 
2013-08-13 11:46:12.275614 [DEBUG] switch_ivr_play_say.c:1703 done playing file /u/freeswitch/sounds/en/us/callie/ivr/ivr-please_enter_the_number_where_we_can_reach_you.wav 
2013-08-13 11:46:22.275617 [DEBUG] mod_httapi.c:1217 Process Tag: [continue] 




With a single digit input, the timeout triggers after 5s: i was expecting 10s (according to input-timeout). 
The log follows: 



2013-08-13 11:48:34.335616 [DEBUG] mod_httapi.c:1217 Process Tag: [playback] 
2013-08-13 11:48:34.335616 [INFO] switch_ivr_async.c:212 Digit parser HTTAPI: Setting realm to 'default' 
2013-08-13 11:48:34.335616 [DEBUG] switch_ivr_async.c:318 Digit parser HTTAPI: binding ^\d+#/default/0 callback: 0x7fa1e8486bd0 data: 0x7fa1ec0b7718 
2013-08-13 11:48:34.335616 [INFO] switch_ivr_async.c:212 Digit parser HTTAPI: Setting realm to 'default' 
2013-08-13 11:48:34.335616 [INFO] switch_ivr_async.c:201 Digit parser HTTAPI: Setting terminators for realm 'default' to '#' 
2013-08-13 11:48:34.335616 [DEBUG] switch_ivr_play_say.c:1315 Codec Activated L16 at 8000hz 1 channels 20ms 
2013-08-13 11:48:36.295615 [DEBUG] switch_ivr_play_say.c:1703 done playing file /u/freeswitch/sounds/en/us/callie/ivr/ivr-please_enter_the_number_where_we_can_reach_you.wav 
2013-08-13 11:48:37.375645 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 1:880 
2013-08-13 11:48:37.375645 [DEBUG] switch_channel.c:471 RECV DTMF 1:880 

2013-08-13 11:48:42.395644 [DEBUG] mod_httapi.c:1217 Process Tag: [continue] 


With a multi-digit input, the timeout triggers after 10s from the end of playback (according to input-timeout). I was expecting a 10s timeout after each digit. 



2013-08-13 11:51:44.715652 [DEBUG] mod_httapi.c:1217 Process Tag: [playback] 
2013-08-13 11:51:44.715652 [INFO] switch_ivr_async.c:212 Digit parser HTTAPI: Setting realm to 'default' 
2013-08-13 11:51:44.715652 [DEBUG] switch_ivr_async.c:318 Digit parser HTTAPI: binding ^\d+#/default/0 callback: 0x7fa1e8486bd0 data: 0x7fa1dc3c0468 
2013-08-13 11:51:44.715652 [INFO] switch_ivr_async.c:212 Digit parser HTTAPI: Setting realm to 'default' 
2013-08-13 11:51:44.715652 [INFO] switch_ivr_async.c:201 Digit parser HTTAPI: Setting terminators for realm 'default' to '#' 
2013-08-13 11:51:44.715652 [DEBUG] switch_ivr_play_say.c:1315 Codec Activated L16 at 8000hz 1 channels 20ms 
2013-08-13 11:51:46.675626 [DEBUG] switch_ivr_play_say.c:1703 done playing file /u/freeswitch/sounds/en/us/callie/ivr/ivr-please_enter_the_number_where_we_can_reach_you.wav 
2013-08-13 11:51:47.615624 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 1:800 
2013-08-13 11:51:47.615624 [DEBUG] switch_channel.c:471 RECV DTMF 1:800 
2013-08-13 11:51:48.255615 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 2:800 
2013-08-13 11:51:48.255615 [DEBUG] switch_channel.c:471 RECV DTMF 2:800 
2013-08-13 11:51:48.975626 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 3:800 
2013-08-13 11:51:48.975626 [DEBUG] switch_channel.c:471 RECV DTMF 3:800 
2013-08-13 11:51:49.735623 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 4:800 
2013-08-13 11:51:49.735623 [DEBUG] switch_channel.c:471 RECV DTMF 4:800 
2013-08-13 11:51:50.355645 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 5:800 
2013-08-13 11:51:50.355645 [DEBUG] switch_channel.c:471 RECV DTMF 5:800 
2013-08-13 11:51:51.035623 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 6:800 
2013-08-13 11:51:51.035623 [DEBUG] switch_channel.c:471 RECV DTMF 6:800 
2013-08-13 11:51:51.755623 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 7:800 
2013-08-13 11:51:51.755623 [DEBUG] switch_channel.c:471 RECV DTMF 7:800 
2013-08-13 11:51:52.455625 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 8:800 
2013-08-13 11:51:52.455625 [DEBUG] switch_channel.c:471 RECV DTMF 8:800 
2013-08-13 11:51:53.235631 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 9:880 
2013-08-13 11:51:53.235631 [DEBUG] switch_channel.c:471 RECV DTMF 9:880 
2013-08-13 11:51:53.995624 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 0:880 
2013-08-13 11:51:53.995624 [DEBUG] switch_channel.c:471 RECV DTMF 0:880 
2013-08-13 11:51:54.695615 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 7:800 
2013-08-13 11:51:54.695615 [DEBUG] switch_channel.c:471 RECV DTMF 7:800 
2013-08-13 11:51:55.575615 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 8:800 
2013-08-13 11:51:55.575615 [DEBUG] switch_channel.c:471 RECV DTMF 8:800 
2013-08-13 11:51:56.675645 [DEBUG] mod_httapi.c:1217 Process Tag: [continue] 




According to Freeswitch wiki, t hese are the definitions of input-timeout and digit-timeout attributes: 
digit-timeout      ?: Pause to wait for digits after file plays (when input bindings are present)
input-timeout      ?: Pause to wait for more digits in a multi-digit input 


However the input-timeout acts as the maximum time the user has to enter the total input digits after the end of playback while the digit-timeout seems to be "ignored" when input-timeout is present. 


What am I missing? 


I'm using the latest stable version of freeswitch (1.2.12). 


Thank you for your help. 

Denis Gasparin 





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From jackal at cybershroud.net  Tue Aug 13 17:06:49 2013
From: jackal at cybershroud.net (Carlos Flor)
Date: Tue, 13 Aug 2013 09:06:49 -0400
Subject: [Freeswitch-users] Establishing SRTP from SBC to endpoint
In-Reply-To: <CABkS=xqD0mukebja+w3aurrhcUJ4Gp=DcOpkh+BDnCpOKr32CA@mail.gmail.com>
References: <CABkS=xqD0mukebja+w3aurrhcUJ4Gp=DcOpkh+BDnCpOKr32CA@mail.gmail.com>
Message-ID: <CAAYhoXLzsB-o=5VNBaRsR9xdS8bm=j5zKxe8ideg5-jmKU=fKw@mail.gmail.com>

Try using rtp_secure_media=true instead of sip_secure_media.  If you are
trying to set it on the b-leg, you probably want to use export instead of
set, or use nolocal:rtp_secure_media.

Hope that helps.


On Mon, Aug 12, 2013 at 10:26 PM, Peter <eidevm5 at gmail.com> wrote:

> In my environment, I have the following (simplified) setup:
>
> FS1  ----  FS SBC ---  FS2
>
> Phones registered to FS1 (100x) use TLS/SRTP and phones registered to FS2
> (200x) use SIP/RTP
>
> FS1 has inbound-bypass-media set to true to allow SRTP peer to peer and
> direct to the SBC.
>
> If I make an inbound call (eg: 1000 to 2000), SRTP is correctly
> established between the phone and SBC with RTP on the other side of the SBC
> to the internal phone.
>
> However, when I try it the other way, I can't get SRTP established from
> the SBC to the external phone.
>
> I've been using https://wiki.freeswitch.org/wiki/Secure_RTP as a guide.
>
> I've even tried explicitly setting sip_secure_media to true on the SBC and
> FS1.
>
> The dialplan on the SBC has:
>
>   <extension name="outgoing">
>         <condition field="destination_number"
> expression="^(10[0-9][0-9])$">
>             <action application="set" data="sip_secure_media=true"/>
>             <action application="bridge" data="sofia/external/${
> destination_number}@10.1.1.204"/>
>         </condition>
>   </extension>
>
>
> And on FS1, the dialplan has:
>
>    <extension name="Local-Numbers">
>       <condition field="destination_number" expression="^(10[01][0-9])$">
>         <action application="export" data="dialed_extension=$1"/>
>         <action application="set" data="sip_secure_media=true"/>
>         <action application="bridge" data="user/${dialed_extension}@
> ${domain_name}"/>
>       </condition>
>     </extension>
>
>
> Note that I've been testing this against two phones with SRTP enabled, but
> only one that is using TLS.  I get the same result calling each phone.
>
> On a related point, what it the step required for a TLS connection from
> the SBC to the phone?   I'm assume the phone just needs the CA cert from
> the SBC.  Correct?
>
> Any information as to where I'm going wrong will be gratefully accepted.
>
> Thanks
>
> Peter
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bob.mccarthy at experient.com  Tue Aug 13 17:09:36 2013
From: bob.mccarthy at experient.com (Bob McCarthy)
Date: Tue, 13 Aug 2013 07:09:36 -0600
Subject: [Freeswitch-users] Invalid application oreka_record
In-Reply-To: <D0BD8814-3F42-4F6F-B7BD-5185D8382A58@freeswitch.org>
References: <009c01ce97af$fc724020$f556c060$@experient.com>
	<D0BD8814-3F42-4F6F-B7BD-5185D8382A58@freeswitch.org>
Message-ID: <015c01ce9826$5dfe01f0$19fa05d0$@experient.com>

Thanks, I had an old modules.conf.xml and it was not in there after the upgrade. Added it and works great !

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Monday, August 12, 2013 6:02 PM
To: FreeSWITCH Users Help
Cc: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Invalid application oreka_record

 

you need to add it to your modules.conf.xml so it loads at freeswitch start time, for a 1 time load, just "load mod_oreka" from fs_cli.

Ken

Sent from my iPad


On Aug 12, 2013, at 18:02, "Bob McCarthy" <bob.mccarthy at experient.com <mailto:bob.mccarthy at experient.com> > wrote:

I am trying to test/use mod_oreka, I have configured via http://wiki.freeswitch.org/wiki/Mod_oreka but in the dialplan 

I get 2013-08-12 16:35:09.231636 [ERR] switch_core_session.c:2557 Invalid Application oreka_record

 

Is there something more I need to do to compile/inlude the oreka module ?  mod_oreka.so is present in the mod directory

 

Bob

 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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://lists.freeswitch.org/mailman/options/freeswitch-users> 
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From nick.zaitsev at mail.ru  Tue Aug 13 12:55:25 2013
From: nick.zaitsev at mail.ru (=?UTF-8?B?TmljayBaYWl0c2V2?=)
Date: Tue, 13 Aug 2013 12:55:25 +0400
Subject: [Freeswitch-users] =?utf-8?q?freeswitch_trunk_cisco_call_manager?=
Message-ID: <1376384125.739558896@f222.i.mail.ru>


Good day to you,
Could you help me to configure the trunk between freeswitch and cisco call manager at the freeswitch side,please.
I can not find any information abot that in the wiki page.
Thank you for your time,
Best regards.
Nikolay Zaitsev.

-- 
Nick Zaitsev
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From beskrovny at gmail.com  Tue Aug 13 14:44:37 2013
From: beskrovny at gmail.com (Veniamin (Benjamin) Beskrovny)
Date: Tue, 13 Aug 2013 12:44:37 +0200
Subject: [Freeswitch-users] nibblebill query error
Message-ID: <CAPhuNs8AvysHnb=Fk5x8Gr=R-uWOgqR4Ydqes5mM4_4WM9Wa4w@mail.gmail.com>

After updating nibblebill conf and freeswitch I've mentioned the
following error:

[ERR] mod_nibblebill.c:403 Error running this query: [SELECT cash AS
nibble_balance FROM clients WHERE accountcode='1'


Running isql from the command line with the same sql gives perfect answer:

[root at sip log]# isql billing
+---------------------------------------+
| Connected!                            |
|                                       |
| sql-statement                         |
| help [tablename]                      |
| quit                                  |
|                                       |
+---------------------------------------+
SQL> SELECT cash AS nibble_balance FROM clients WHERE accountcode=1
+-------------------------+
| nibble_balance          |
+-------------------------+
| -3126.29999017714       |
+-------------------------+
SQLRowCount returns 1
1 rows fetched
SQL>

What could be the problem? Any ideas?


From beskrovny at gmail.com  Tue Aug 13 17:15:43 2013
From: beskrovny at gmail.com (Veniamin (Benjamin) Beskrovny)
Date: Tue, 13 Aug 2013 15:15:43 +0200
Subject: [Freeswitch-users] mod_nibblebill.c:558 Failed to log to database
Message-ID: <CAPhuNs9BTsFX703-ydEcA7r9NSMfOuwi3xKxV8xROPgBLV2Ddg@mail.gmail.com>

Hi everyone. Can anyone tell me how can I troubleshoot that kind of
error: mod_nibblebill.c:558 Failed to log to database

PS: using MySQL via ODBC. In nibblebill conf I have: odbc-dsn="odbc://billing"

ISQL test: isql billing - works good. Show tables, queries, e.t.c.

Any ideas?


From mike at jerris.com  Tue Aug 13 18:50:11 2013
From: mike at jerris.com (Michael Jerris)
Date: Tue, 13 Aug 2013 10:50:11 -0400
Subject: [Freeswitch-users] Recommended ULIMIT settings
In-Reply-To: <CAO47HpejFCiJzA5oawdv5OmgGuAMAp0n3+-J=+Y5p8_fyZgMiA@mail.gmail.com>
References: <CAE2+g++iHtJH4q5jkVoL5y_6W6kxqDsoB848XC9Xg7_fLqyDtQ@mail.gmail.com>
	<C958FB5F-04D1-433A-815B-AA2D6D2AEA51@freeswitch.org>
	<CAO47HpffL-JkE=K8k73H3C-bRWtkope+cy53V+GoGRq5xqVbwA@mail.gmail.com>
	<EFDA627B-703B-4FD2-91AC-68ED5F343FE0@freeswitch.org>
	<CAO47HpejFCiJzA5oawdv5OmgGuAMAp0n3+-J=+Y5p8_fyZgMiA@mail.gmail.com>
Message-ID: <31BD2DE0-6D73-4FDC-81C9-D63CD63B717A@jerris.com>

240 or maybe 244 is right.. 8192 is not.. that being said.. it can cause issues on some newer linux, it will change that setting for everything, so you might need to do it in a bash script that starts freeswitch instead.

On Aug 12, 2013, at 10:53 PM, jay binks <jaybinks at gmail.com> wrote:

> can we just get a concensus from the people who know ?? :)
> 
> Anthm, MikeJ, BWK ..
> 
> it sounds like ulimit -s 240 is the way to go, right ?
> 
> 
> 
> 
> On 13 August 2013 12:28, Ken Rice <krice at freeswitch.org> wrote:
> well if you look at the wiki history there seems to be a fight over what to set it to, i always set it to 240 cause it makes that error msg anout stack > 240 at startup go away
> 
> 
> Ken
> Sent from my iPad
> 
> On Aug 12, 2013, at 21:13, jay binks <jaybinks at gmail.com> wrote:
> 
>> So Ken,
>> 
>> we should change http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
>> to be :
>> ulimit -s 240    
>> 
>> Just want to be 100% sure before changing it .
>> 
>> Jay
>> 
>> 
>> 
>> 
>> 
>> On 13 August 2013 11:18, Ken Rice <krice at freeswitch.org> wrote:
>> yes this is normal, if fs can auto adjust it to 240 it will... you dont need > that for the thread stack anyway
>> 
>> Ken
>> Sent from my iPad
>> 
>> On Aug 12, 2013, at 19:52, Nuno Reis <nreis at wavecom.pt> wrote:
>> 
>>> Hi guys.
>>> 
>>> I'm currently using the recommended ulimit settings as defined here:
>>> 
>>> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
>>> 
>>> Although there's at least one parameter that doesn't seem to be right when i do a status at fs_cli:
>>> 
>>> UP 0 years, 0 days, 10 hours, 25 minutes, 14 seconds, 617 milliseconds, 626 microseconds
>>> FreeSWITCH is ready
>>> 55390 session(s) since startup
>>> 0 session(s) 0/100
>>> 5000 session(s) max
>>> min idle cpu 0.00/98.00
>>> Current Stack Size/Max 240K/240K
>>> 
>>> The stack size should be 8192 as defined, but FS shows otherwise.
>>> 
>>> I'm using centOS x86_64 and a 64 bit version of FS.
>>> Is this normal?
>>> Thanks.
>>> 
>>> --
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From ssinyagin at yahoo.com  Tue Aug 13 19:18:35 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Tue, 13 Aug 2013 08:18:35 -0700 (PDT)
Subject: [Freeswitch-users]
 =?utf-8?q?=EF=BB=BFfreeswitch_trunk_cisco_call?=
 =?utf-8?q?_manager?=
In-Reply-To: <1376384125.739558896@f222.i.mail.ru>
References: <1376384125.739558896@f222.i.mail.ru>
Message-ID: <1376407115.82902.YahooMailNeo@web126201.mail.ne1.yahoo.com>

On CUCM side, you set "MTP required" for the SIP trunk. The trunk should terminate on port 5080 on FreeSWITCH server (5060 requires authentication).

On FreeSWITCH side, a standard <gateway> needs to be configured to be able to send calls to CUCM. The calls from CUCM will land in "public" profile.

Feel free to contact me privately, I can speak Russian :)







>________________________________
> From: Nick Zaitsev <nick.zaitsev at mail.ru>
>To: freeswitch-users at lists.freeswitch.org 
>Sent: Tuesday, August 13, 2013 10:55 AM
>Subject: [Freeswitch-users] ?freeswitch trunk cisco call manager
> 
>
>
>
>Good day to you,
>Could you help me to configure the trunk between freeswitch and cisco call manager at the freeswitch side,please.
>I can not find any information abot that in the wiki page.
>Thank you for your time,
>Best regards.
>Nikolay Zaitsev.
>
>-- 
>Nick Zaitsev
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 dgarcia at anew.com.ve  Tue Aug 13 19:24:22 2013
From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar)
Date: Tue, 13 Aug 2013 10:54:22 -0430
Subject: [Freeswitch-users] freeswitch trunk cisco call manager
In-Reply-To: <1376384125.739558896@f222.i.mail.ru>
References: <1376384125.739558896@f222.i.mail.ru>
Message-ID: <520A4FA6.5010000@anew.com.ve>

On wiki you will get a lot of info, for example:

http://wiki.freeswitch.org/wiki/Traditional_Gateway_connecting



On 8/13/2013 4:25 AM, Nick Zaitsev wrote:
>
> Good day to you,
> Could you help me to configure the trunk between freeswitch and cisco 
> call manager at the freeswitch side,please.
> I can not find any information abot that in the wiki page.
> Thank you for your time,
> Best regards.
> Nikolay Zaitsev.
>
> -- 
> Nick Zaitsev
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> No virus found in this message.
> Checked by AVG - www.avg.com <http://www.avg.com>
> Version: 2012.0.2242 / Virus Database: 3209/6042 - Release Date: 08/01/13
> Internal Virus Database is out of date.
>


-- 
Atentamente,
*Dario Garc?a*
Consultor.

CCCT, Nivel C2, Sector Yarey, Mz,
Ofc. MZ03a.
Caracas-Venezuela.
Tel?fono: +58 212 9081842
Cel: +58 412 2221515
dgarcia at anew.com.ve
http://www.anew.com.ve
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From smontour at verizon.net  Tue Aug 13 19:30:05 2013
From: smontour at verizon.net (Sami Montour)
Date: Tue, 13 Aug 2013 10:30:05 -0500
Subject: [Freeswitch-users] Real-time billing application for the
 FreeSWITCH (mod_lua, mod_perl or ESL)
Message-ID: <005701ce9839$fccc2c40$f66484c0$@verizon.net>

Thanks Mike. I appreciate it.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Monday, August 12, 2013 10:25 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Real-time billing application for the
FreeSWITCH (mod_lua, mod_perl or ESL)

 

If your happy with C, go for it.  You can take a look at mod_xml_cdr and
mod_nibblebill for some hints on hooks for billing.  I might use the state
hooks for reporting instead and the data you get from that one as you are
less time constrained to handle the data in that state.

 

Mike

 

On Aug 9, 2013, at 2:29 PM, Sami Montour <smontour at verizon.net> wrote:





Hello Everyone,

I am a long time user of OpenSIPS and have recently been tasked with
developing a call control/real-time billing application for the FreeSWITCH.
I would like to develop the application in either C/C++, Perl or lua
(totally new to lua) in this order.

 

I have used ESL to develop some basic functionality of the application using
Perl. For instance, when the call comes in, I pass over control from
FreeSWITCH to the call control application using outbound TCP socket. Then
the call control application listens for CHANNEL_ANSWER event to start
billing process and CHANNEL_HANGUP event to stop billing. The application
does some database work as well to retrieve and check user status and
balance and all that stuff. My main concern is with load testing because of
the TCP connection between the FreeSWITCH and call control application and
also TCP connection to database to retrieve subscriber info. The basic
functionality seems to work fine but haven't tested it with high call
volume, which I am about to do.

 

The last couple of days I have been experimenting with mod_lua but can't
find a method under "Session" to get information related to CHANNEL_ANSWER
and CHANNEL_HANGUP events. The "session:geVariable" does not have variables
related to session answer time or hangup time, something similar to
"$e->getHeader("Caller-Channel-Answered-Time")"  in ESL. I must be missing
something very simple here. Again, my experience with mod_lua is only a
couple of days old.

 

I am new Freeswitch and thought I would seek the advice of all the experts
and people with more experience with Freeswitch. For this type of
application, am I better off with mod_lua, or mod_perl, or shall I stick
with ESL using Perl? How about ESL with C/C++? Will I gain some performance
gain there?

 

Any feedback would be appreciated. Thanks.

 

Sam

 

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From sravi123 at yahoo.com  Tue Aug 13 22:00:45 2013
From: sravi123 at yahoo.com (Ravi)
Date: Tue, 13 Aug 2013 23:30:45 +0530
Subject: [Freeswitch-users] How does Freeswitch process incoming calls
Message-ID: <520A744D.6020301@yahoo.com>

Hello Everyone,

I am trying to configure a PRI connection. When I call from my mobile 
number to the PRI landline number, I get the following log: The call has 
never connected, I hear an engaged tone and it disconnects.

Question 1: In the fourth line that is marked in red, the message says 
Called No:[000]. The number that I actually dialed from my mobile 
(7502029000) is +914274302000. I am not able to figure out what only 
three digits are displayed.

Question 2: When freeswitch receives an inbound call - in what sequence 
is the call processed ? What dialplans should I be looking at?
Question 3: How do I or where do I find the number that is being called 
in freeswitch ?


Any pointers or advise on how to go about this, is much appreciated.

Thanks.
Ravi


2013-08-13 11:08:45.145709 [DEBUG] ftmod_wanpipe.c:1400 [s1c31][1:16] 
Link status is 1
2013-08-13 11:08:45.145709 [INFO] ftmod_sangoma_isdn_stack_rcv.c:96 
[s1c9][1:9] Received SETUP (suId:1 suInstId:0 spInstId:9)
2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:57 
[s1c9][1:9] Processing SETUP (suId:1 suInstId:0 spInstId:9)
2013-08-13 11:08:45.145709 [INFO] *ftmod_sangoma_isdn_stack_hndl.c:153 
[s1c9][1:9] Incoming call: Called No:[000] Calling No:[7502029000]*
2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:209 
[s1c9][1:9] Changed state from DOWN to RING
2013-08-13 11:08:45.145709 [DEBUG] ftdm_state.c:541 [s1c9][1:9] 
Executing state processor for RING
2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:655 [s1c9][1:9] 
processing state change to RING
2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:676 [s1c9][1:9] 
Sending incoming call from 7502029000 to 000 to FTDM core
2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:923 [s1c9][1:9] 
Completed state change from DOWN to RING in 0 ms
2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:2657 got clear channel 
sig [START]
2013-08-13 11:08:45.145709 [DEBUG] ftmod_wanpipe.c:749 [s1c9][1:9] 
Enabled DTMF events
2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:403 Set codec PCMA 20ms
2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:1890 Connect inbound 
channel FreeTDM/1:9/000
2013-08-13 11:08:45.145709 [NOTICE] switch_channel.c:1030 New Channel 
FreeTDM/1:9/000 [9eb4ce34-03da-11e3-8bc5-0b625c405e53]
2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:2093 (FreeTDM/1:9/000) 
State Change CS_NEW -> CS_INIT
2013-08-13 11:08:45.145709 [DEBUG] switch_core_session.c:1341 Send 
signal FreeTDM/1:9/000 [BREAK]
2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:416 
(FreeTDM/1:9/000) Running State Change CS_INIT
2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:455 
(FreeTDM/1:9/000) State INIT
2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:430 (FreeTDM/1:9/000) 
State Change CS_INIT -> CS_ROUTING
2013-08-13 11:08:45.145709 [DEBUG] switch_core_session.c:1341 Send 
signal FreeTDM/1:9/000 [BREAK]
2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:455 
(FreeTDM/1:9/000) State INIT going to sleep
2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:416 
(FreeTDM/1:9/000) Running State Change CS_ROUTING
2013-08-13 11:08:45.145709 [DEBUG] switch_channel.c:2116 
(FreeTDM/1:9/000) Callstate Change DOWN -> RINGING
2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:471 
(FreeTDM/1:9/000) State ROUTING
2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:453 FreeTDM/1:9/000 
CHANNEL ROUTING
2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:456 [s1c9][1:9] 
Indicating PROCEED in state RING
2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:456 [s1c9][1:9] Changed 
state from RING to PROCEED
2013-08-13 11:08:45.145709 [DEBUG] ftdm_state.c:541 [s1c9][1:9] 
Executing state processor for PROCEED
2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:655 [s1c9][1:9] 
processing state change to PROCEED
2013-08-13 11:08:45.145709 [INFO] ftmod_sangoma_isdn_stack_out.c:175 
[s1c9][1:9] Sending PROCEED (suId:1 suInstId:9 spInstId:9 dchan:1 ces:0)
2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:923 [s1c9][1:9] 
Completed state change from RING to PROCEED in 0 ms
2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:117 
FreeTDM/1:9/000 Standard ROUTING
2013-08-13 11:08:45.145709 [INFO] mod_dialplan_xml.c:558 Processing 
<7502029000>->000 in context default
Dialplan: FreeTDM/1:9/000 parsing [default->unloop] continue=false
Dialplan: FreeTDM/1:9/000 Regex (PASS) [unloop] ${unroll_loops}(true) =~ 
/^true$/ break=on-false
Dialplan: FreeTDM/1:9/000 Regex (FAIL) [unloop] ${sip_looped_call}() =~ 
/^true$/ break=on-false
Dialplan: FreeTDM/1:9/000 parsing [default->tod_example] continue=true
Dialplan: FreeTDM/1:9/000 Date/Time Match (PASS) [tod_example] 
break=on-false
Dialplan: FreeTDM/1:9/000 Action set(open=true)
Dialplan: FreeTDM/1:9/000 parsing [default->holiday_example] continue=true
Dialplan: FreeTDM/1:9/000 Date/TimeMatch (FAIL) [holiday_example] 
break=on-false
Dialplan: FreeTDM/1:9/000 parsing [default->global-intercept] continue=false
Dialplan: FreeTDM/1:9/000 Regex (FAIL) [global-intercept] 
destination_number(000) =~ /^886$/ break=on-false
Dialplan: FreeTDM/1:9/000 parsing [default->group-intercept] continue=false


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From msc at freeswitch.org  Tue Aug 13 22:29:27 2013
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Aug 2013 11:29:27 -0700
Subject: [Freeswitch-users] How does Freeswitch process incoming calls
In-Reply-To: <520A744D.6020301@yahoo.com>
References: <520A744D.6020301@yahoo.com>
Message-ID: <CAKzWOxWFeL83ymLVhWHyRjgjSY8WDOxjOGokXNQwq0xZK_uSMg@mail.gmail.com>

On Tue, Aug 13, 2013 at 11:00 AM, Ravi <sravi123 at yahoo.com> wrote:

>  Hello Everyone,
>
> I am trying to configure a PRI connection. When I call from my mobile
> number to the PRI landline number, I get the following log: The call has
> never connected, I hear an engaged tone and it disconnects.
>
> Question 1: In the fourth line that is marked in red, the message says
> Called No:[000]. The number that I actually dialed from my mobile
> (7502029000) is +914274302000. I am not able to figure out what only
> three digits are displayed.
>
It looks like the carrier is only sending you the last 3 digits of the
called DID.


>
> Question 2: When freeswitch receives an inbound call - in what sequence is
> the call processed ? What dialplans should I be looking at?
>
Once mod_freetdm figures out what the called number is it will try to route
it in whatever context is specified in your freetdm.conf.xml file. In your
case it goes to default as you can see by this log line:

2013-08-13 11:08:45.145709 [INFO] mod_dialplan_xml.c:558 Processing
<7502029000>->000 in context default


> Question 3: How do I or where do I find the number that is being called in
> freeswitch ?
>
See my answer to question #2. Basically you need an extension with a
condition that matches 000, something like:
<condition field="destination_number" expression="^000$>

-MC


>
>
> Any pointers or advise on how to go about this, is much appreciated.
>
> Thanks.
> Ravi
>
>
> 2013-08-13 11:08:45.145709 [DEBUG] ftmod_wanpipe.c:1400 [s1c31][1:16] Link
> status is 1
> 2013-08-13 11:08:45.145709 [INFO] ftmod_sangoma_isdn_stack_rcv.c:96
> [s1c9][1:9] Received SETUP (suId:1 suInstId:0 spInstId:9)
> 2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:57
> [s1c9][1:9] Processing SETUP (suId:1 suInstId:0 spInstId:9)
> 2013-08-13 11:08:45.145709 [INFO] *ftmod_sangoma_isdn_stack_hndl.c:153
> [s1c9][1:9] Incoming call: Called No:[000] Calling No:[7502029000]*
> 2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn_stack_hndl.c:209
> [s1c9][1:9] Changed state from DOWN to RING
> 2013-08-13 11:08:45.145709 [DEBUG] ftdm_state.c:541 [s1c9][1:9] Executing
> state processor for RING
> 2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:655 [s1c9][1:9]
> processing state change to RING
> 2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:676 [s1c9][1:9]
> Sending incoming call from 7502029000 to 000 to FTDM core
> 2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:923 [s1c9][1:9]
> Completed state change from DOWN to RING in 0 ms
> 2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:2657 got clear channel
> sig [START]
> 2013-08-13 11:08:45.145709 [DEBUG] ftmod_wanpipe.c:749 [s1c9][1:9] Enabled
> DTMF events
> 2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:403 Set codec PCMA 20ms
> 2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:1890 Connect inbound
> channel FreeTDM/1:9/000
> 2013-08-13 11:08:45.145709 [NOTICE] switch_channel.c:1030 New Channel
> FreeTDM/1:9/000 [9eb4ce34-03da-11e3-8bc5-0b625c405e53]
> 2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:2093 (FreeTDM/1:9/000)
> State Change CS_NEW -> CS_INIT
> 2013-08-13 11:08:45.145709 [DEBUG] switch_core_session.c:1341 Send signal
> FreeTDM/1:9/000 [BREAK]
> 2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:416
> (FreeTDM/1:9/000) Running State Change CS_INIT
> 2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:455
> (FreeTDM/1:9/000) State INIT
> 2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:430 (FreeTDM/1:9/000)
> State Change CS_INIT -> CS_ROUTING
> 2013-08-13 11:08:45.145709 [DEBUG] switch_core_session.c:1341 Send signal
> FreeTDM/1:9/000 [BREAK]
> 2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:455
> (FreeTDM/1:9/000) State INIT going to sleep
> 2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:416
> (FreeTDM/1:9/000) Running State Change CS_ROUTING
> 2013-08-13 11:08:45.145709 [DEBUG] switch_channel.c:2116 (FreeTDM/1:9/000)
> Callstate Change DOWN -> RINGING
> 2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:471
> (FreeTDM/1:9/000) State ROUTING
> 2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:453 FreeTDM/1:9/000
> CHANNEL ROUTING
> 2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:456 [s1c9][1:9]
> Indicating PROCEED in state RING
> 2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:456 [s1c9][1:9] Changed
> state from RING to PROCEED
> 2013-08-13 11:08:45.145709 [DEBUG] ftdm_state.c:541 [s1c9][1:9] Executing
> state processor for PROCEED
> 2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:655 [s1c9][1:9]
> processing state change to PROCEED
> 2013-08-13 11:08:45.145709 [INFO] ftmod_sangoma_isdn_stack_out.c:175
> [s1c9][1:9] Sending PROCEED (suId:1 suInstId:9 spInstId:9 dchan:1 ces:0)
> 2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:923 [s1c9][1:9]
> Completed state change from RING to PROCEED in 0 ms
> 2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:117
> FreeTDM/1:9/000 Standard ROUTING
> 2013-08-13 11:08:45.145709 [INFO] mod_dialplan_xml.c:558 Processing
> <7502029000>->000 in context default
> Dialplan: FreeTDM/1:9/000 parsing [default->unloop] continue=false
> Dialplan: FreeTDM/1:9/000 Regex (PASS) [unloop] ${unroll_loops}(true) =~
> /^true$/ break=on-false
> Dialplan: FreeTDM/1:9/000 Regex (FAIL) [unloop] ${sip_looped_call}() =~
> /^true$/ break=on-false
> Dialplan: FreeTDM/1:9/000 parsing [default->tod_example] continue=true
> Dialplan: FreeTDM/1:9/000 Date/Time Match (PASS) [tod_example]
> break=on-false
> Dialplan: FreeTDM/1:9/000 Action set(open=true)
> Dialplan: FreeTDM/1:9/000 parsing [default->holiday_example] continue=true
> Dialplan: FreeTDM/1:9/000 Date/TimeMatch (FAIL) [holiday_example]
> break=on-false
> Dialplan: FreeTDM/1:9/000 parsing [default->global-intercept]
> continue=false
> Dialplan: FreeTDM/1:9/000 Regex (FAIL) [global-intercept]
> destination_number(000) =~ /^886$/ break=on-false
> Dialplan: FreeTDM/1:9/000 parsing [default->group-intercept] continue=false
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From nreis at wavecom.pt  Tue Aug 13 22:31:35 2013
From: nreis at wavecom.pt (Nuno Reis)
Date: Tue, 13 Aug 2013 19:31:35 +0100
Subject: [Freeswitch-users] fsctl debug_sql (CRITICAL warnings/output)
Message-ID: <CAE2+g++Q5Jk=RwyK81SK+s5E5uxtrpjqzeYNhazKXtNoB-aY3w@mail.gmail.com>

Hi guys.

I've just installed the new FS 1.2.12 with native postgres support enabled
in core and I'm noticing that when i turn the debug_sql on with fsctl
debug_sql command i have a lot of critical messages as bellow:

2013-08-13 18:02:37.229762 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
[0|0|0|0]--[1]
2013-08-13 18:02:40.009898 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
[0|0|0|0]--[2]
2013-08-13 18:02:41.229721 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
[0|0|0|0]--[1]
2013-08-13 18:02:53.929745 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
[0|0|0|0]--[2]
2013-08-13 18:02:56.289765 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
[0|0|0|0]--[1]
2013-08-13 18:02:57.309725 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
[0|0|0|0]--[2]
2013-08-13 18:02:57.729728 [CRIT] switch_core_sqldb.c:1885 sofia:external
RUN QUEUE [0|0]--[5]
2013-08-13 18:02:57.729728 [CRIT] switch_core_sqldb.c:1885
sofia:external-local RUN QUEUE [0|0]--[5]
2013-08-13 18:02:57.729728 [CRIT] switch_core_sqldb.c:1885
sofia:internal-nmreis RUN QUEUE [0|0]--[5]
2013-08-13 18:02:57.849741 [CRIT] switch_core_sqldb.c:1885 sofia:internal
RUN QUEUE [0|0]--[5]
2013-08-13 18:02:58.849750 [CRIT] switch_core_sqldb.c:1885
sofia:nat-traversal RUN QUEUE [0|0]--[5]
2013-08-13 18:03:06.989721 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
[0|0|0|0]--[4]
2013-08-13 18:03:08.949813 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
[0|0|0|0]--[2]
2013-08-13 18:03:11.349756 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
[0|0|0|0]--[1]
2013-08-13 18:03:17.389719 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
[0|0|0|0]--[1]
2013-08-13 18:03:26.389718 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
[0|0|0|0]--[1]
2013-08-13 18:03:27.829818 [CRIT] switch_core_sqldb.c:1885 sofia:external
RUN QUEUE [0|0]--[5]
2013-08-13 18:03:27.829818 [CRIT] switch_core_sqldb.c:1885
sofia:external-local RUN QUEUE [0|0]--[5]
2013-08-13 18:03:27.829818 [CRIT] switch_core_sqldb.c:1885
sofia:internal-nmreis RUN QUEUE [0|0]--[5]
2013-08-13 18:03:27.929794 [CRIT] switch_core_sqldb.c:1885 sofia:internal
RUN QUEUE [0|0]--[5]
2013-08-13 18:03:27.989786 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
[0|0|0|0]--[2]
2013-08-13 18:03:28.969712 [CRIT] switch_core_sqldb.c:1885
sofia:nat-traversal RUN QUEUE [0|0]--[5]

IS this normal? Everything seems to be fine though.
Looking forward to hear from you.

BR,

--

*Nuno Miguel Reis* | *Unified Communication** Systems*
M. +351 913907481 | nreis at wavecom.pt
WAVECOM-Solu??es R?dio, S.A.
Cacia Park | Rua do Progresso, Lote 15
3800-639 AVEIRO | Portugal
T. +351 309 700 225 | F. +351 234 919 191
*GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
www.wavecom.pt** <http://www.wavecom.pt/>*

[image: Description: Description:
WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php>

[image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>
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From nreis at wavecom.pt  Tue Aug 13 23:36:17 2013
From: nreis at wavecom.pt (Nuno Reis)
Date: Tue, 13 Aug 2013 20:36:17 +0100
Subject: [Freeswitch-users] Recommended ULIMIT settings
In-Reply-To: <31BD2DE0-6D73-4FDC-81C9-D63CD63B717A@jerris.com>
References: <CAE2+g++iHtJH4q5jkVoL5y_6W6kxqDsoB848XC9Xg7_fLqyDtQ@mail.gmail.com>
	<C958FB5F-04D1-433A-815B-AA2D6D2AEA51@freeswitch.org>
	<CAO47HpffL-JkE=K8k73H3C-bRWtkope+cy53V+GoGRq5xqVbwA@mail.gmail.com>
	<EFDA627B-703B-4FD2-91AC-68ED5F343FE0@freeswitch.org>
	<CAO47HpejFCiJzA5oawdv5OmgGuAMAp0n3+-J=+Y5p8_fyZgMiA@mail.gmail.com>
	<31BD2DE0-6D73-4FDC-81C9-D63CD63B717A@jerris.com>
Message-ID: <CAE2+g+KY+d4+rnnEGuXXbYvdsJjqaBGPsV0MOES0GZLP1r8VbA@mail.gmail.com>

Hi guys.

I've just compiled the latest v1.2_stable branch and now i can see the
following output after a status command in fs_cli:

UP 0 years, 0 days, 0 hours, 0 minutes, 21 seconds, 678 milliseconds, 716
microseconds
FreeSWITCH (Version 1.2.12 ) is ready
0 session(s) since startup
0 session(s) - peak 0, last 5min 0
0 session(s) per Sec out of max 30, peak 0, last 5min 0
1000 session(s) max
min idle cpu 0.00/99.00
*Current Stack Size/Max 240K/8192K

*It seems FS still sticks to the 240K stack size although it can now see
the 8192K Max.
Moreover that error msg about stack > 240 at startup has disappeared.
Anymore interpretations on this?

--

*Nuno Miguel Reis* | *Unified Communication** Systems*
M. +351 913907481 | nreis at wavecom.pt
WAVECOM-Solu??es R?dio, S.A.
Cacia Park | Rua do Progresso, Lote 15
3800-639 AVEIRO | Portugal
T. +351 309 700 225 | F. +351 234 919 191
*GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
www.wavecom.pt** <http://www.wavecom.pt/>*

[image: Description: Description:
WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php>

[image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>




On Tue, Aug 13, 2013 at 3:50 PM, Michael Jerris <mike at jerris.com> wrote:

> 240 or maybe 244 is right.. 8192 is not.. that being said.. it can cause
> issues on some newer linux, it will change that setting for everything, so
> you might need to do it in a bash script that starts freeswitch instead.
>
> On Aug 12, 2013, at 10:53 PM, jay binks <jaybinks at gmail.com> wrote:
>
> can we just get a concensus from the people who know ?? :)
>
> Anthm, MikeJ, BWK ..
>
> it sounds like ulimit -s 240 is the way to go, right ?
>
>
>
>
> On 13 August 2013 12:28, Ken Rice <krice at freeswitch.org> wrote:
>
>> well if you look at the wiki history there seems to be a fight over what
>> to set it to, i always set it to 240 cause it makes that error msg anout
>> stack > 240 at startup go away
>>
>>
>> Ken
>> Sent from my iPad
>>
>> On Aug 12, 2013, at 21:13, jay binks <jaybinks at gmail.com> wrote:
>>
>> So Ken,
>>
>> we should change
>> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
>>  to be :
>>
>> ulimit -s 240
>>
>>
>> Just want to be 100% sure before changing it .
>>
>> Jay
>>
>>
>>
>>
>>
>> On 13 August 2013 11:18, Ken Rice <krice at freeswitch.org> wrote:
>>
>>> yes this is normal, if fs can auto adjust it to 240 it will... you dont
>>> need > that for the thread stack anyway
>>>
>>> Ken
>>> Sent from my iPad
>>>
>>> On Aug 12, 2013, at 19:52, Nuno Reis <nreis at wavecom.pt> wrote:
>>>
>>> Hi guys.
>>>
>>> I'm currently using the recommended ulimit settings as defined here:
>>>
>>> http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations
>>>
>>> Although there's at least one parameter that doesn't seem to be right
>>> when i do a status at fs_cli:
>>>
>>> UP 0 years, 0 days, 10 hours, 25 minutes, 14 seconds, 617 milliseconds,
>>> 626 microseconds
>>> FreeSWITCH is ready
>>> 55390 session(s) since startup
>>> 0 session(s) 0/100
>>> 5000 session(s) max
>>> min idle cpu 0.00/98.00
>>> *Current Stack Size/Max 240K/240K*
>>>
>>> The stack size should be 8192 as defined, but FS shows otherwise.
>>>
>>> I'm using centOS x86_64 and a 64 bit version of FS.
>>> Is this normal?
>>> Thanks.
>>>
>>> --
>>>
>>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Tue Aug 13 23:52:29 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Wed, 14 Aug 2013 01:22:29 +0530
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAFiqYumwEGA7zYmJPpQ51w6q_1WsQARht5Z2Gyutoqa4KuRt2w@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<CAMvhjJtBONsXnrhRoDdoMTUPhHb1NjVRSJOsdbt5u+0tjqaBug@mail.gmail.com>
	<CAFiqYuneYGGujBH-a_hanzEQ72Rdiks5kfwntwz2QGp1YC9A7A@mail.gmail.com>
	<5209575C.9030203@freeswitch.org>
	<CAMvhjJuZqjssQ896TBxAFKXUrkrNb=60EJ9kKdkOfWK4VD3Rsg@mail.gmail.com>
	<CAFiqYumwEGA7zYmJPpQ51w6q_1WsQARht5Z2Gyutoqa4KuRt2w@mail.gmail.com>
Message-ID: <CAMvhjJv3oNWNQG5soCd_TNAkaBLFxzmVzELt5JZ=Skgu2vM4mg@mail.gmail.com>

the issue was that i was only running fs_cli on my machine without running
freeswitch. fs_cli works only when freeswitch is running in background.
 thank you
Ashish Mishra


On Tue, Aug 13, 2013 at 1:47 PM, Steven Ayre <steveayre at gmail.com> wrote:

> What was the issue?
>
>
> On 13 August 2013 08:16, Ashish Mishra <itsme.kunnu at gmail.com> wrote:
>
>> Thank you for your reply...i am now able to run fs_cli on my ubuntu
>> machine on which freeswitch is installed...:-)
>> On Aug 13, 2013 3:17 AM, "I put the Who? in Mishehu" <
>> mishehu at freeswitch.org> wrote:
>>
>>>  I have the nagging feeling that Ashish isn't telling the whole story
>>> here...
>>>
>>> Ashish -  what happens if you are on the machine that is running
>>> FreeSWITCH and you execute `telnet localhost 8021` ?  Do you see something
>>> like this:
>>>
>>> # telnet localhost 8021
>>> Trying 127.0.0.1...
>>> Connected to localhost.
>>> Escape character is '^]'.
>>> Content-Type: auth/request
>>>
>>> If you are running with the standard configurations this should work in
>>> this fashion.  If you have changed the
>>> conf/autoload_configs/event_socket.conf.xml , then paste those
>>> configurations.  (If too lengthy, please post to
>>> http://pastebin.freeswitch.org )
>>>
>>> -Yossi
>>>
>>> On 08/12/2013 09:44 AM, Steven Ayre wrote:
>>>
>>> Have you checked the logfile and netstat?
>>>
>>>
>>> On 12 August 2013 09:51, Ashish Mishra <itsme.kunnu at gmail.com> wrote:
>>>
>>>> I reinstalled freeswitch but still i am getting the same error :
>>>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>>> mod_socket is also loaded...i have also flushed the firewall rules...pls
>>>> help
>>>>  On Aug 6, 2013 2:30 AM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>>>>
>>>>> When i am trying to launch fs_cli on my ubuntu 12.04 machine (on which
>>>>> i have installed freeswitch) it gives me the following error :
>>>>> fs_cli .c:1455 main() Error Connecting [Socket Connection Error ]
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>>
>>> 
>>>
>>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nreis at wavecom.pt  Tue Aug 13 23:53:44 2013
From: nreis at wavecom.pt (Nuno Reis)
Date: Tue, 13 Aug 2013 20:53:44 +0100
Subject: [Freeswitch-users] freeswitch startup with lots of errors while
	executing db queries
Message-ID: <CAE2+g++zo6AFuVbuaBNN1Er_Zw=Qzi+MVbGLEpG9gCh4625cQw@mail.gmail.com>

Hi guys.

I've just compiled the latest v1.2stable branch with native postgres
support enabled and now everytime FS startsup i see a lot of errors while
it tries i belive to create the FS database when it's already there:

see pb 21294 <http://pastebin.freeswitch.org/pastebin.php?dl=21294> for a
more complete insight on this issue.

I think this is normal right?

BR,

--

*Nuno Miguel Reis* | *Unified Communication** Systems*
M. +351 913907481 | nreis at wavecom.pt
WAVECOM-Solu??es R?dio, S.A.
Cacia Park | Rua do Progresso, Lote 15
3800-639 AVEIRO | Portugal
T. +351 309 700 225 | F. +351 234 919 191
*GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
www.wavecom.pt** <http://www.wavecom.pt/>*

[image: Description: Description:
WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php>

[image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>
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From nreis at wavecom.pt  Tue Aug 13 23:55:23 2013
From: nreis at wavecom.pt (Nuno Reis)
Date: Tue, 13 Aug 2013 20:55:23 +0100
Subject: [Freeswitch-users] freeswitch startup with lots of errors while
	executing db queries
In-Reply-To: <CAE2+g++zo6AFuVbuaBNN1Er_Zw=Qzi+MVbGLEpG9gCh4625cQw@mail.gmail.com>
References: <CAE2+g++zo6AFuVbuaBNN1Er_Zw=Qzi+MVbGLEpG9gCh4625cQw@mail.gmail.com>
Message-ID: <CAE2+g+KFfRpvTg-7MMFRYJ_1DrBb4MJ_Yy7t-Na+M+kwHFVjnw@mail.gmail.com>

Hi guys.

I've just compiled the latest v1.2stable branch with native postgres
support enabled and now everytime FS startsup i see a lot of errors while
it tries i belive to create the FS database when it's already there:

see pb 21294 <http://pastebin.freeswitch.org/21294> for a more complete
insight on this issue.

I think this is normal right?

BR,

--

*Nuno Miguel Reis* | *Unified Communication** Systems*
M. +351 913907481 | nreis at wavecom.pt
WAVECOM-Solu??es R?dio, S.A.
Cacia Park | Rua do Progresso, Lote 15
3800-639 AVEIRO | Portugal
T. +351 309 700 225 | F. +351 234 919 191
*GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
www.wavecom.pt** <http://www.wavecom.pt/>*

[image: Description: Description:
WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php>

[image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>
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From mike at jerris.com  Wed Aug 14 00:02:02 2013
From: mike at jerris.com (Michael Jerris)
Date: Tue, 13 Aug 2013 16:02:02 -0400
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CAMvhjJu8H0Y25Gd1qtJ0kEb9eiu_D56+rZVJy_zbHV7mEzMLcA@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CAMvhjJu8H0Y25Gd1qtJ0kEb9eiu_D56+rZVJy_zbHV7mEzMLcA@mail.gmail.com>
Message-ID: <72BF927D-CDAE-4682-B7A3-B8B81F969260@jerris.com>

Would you like to share the log so someone can look at it?

On Aug 13, 2013, at 3:50 AM, Ashish Mishra <itsme.kunnu at gmail.com> wrote:

> Hello Peter...i set the fs cli to debugging by typing /log 7 and then retried to connect...but still the same result...
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From mike at jerris.com  Wed Aug 14 00:03:10 2013
From: mike at jerris.com (Michael Jerris)
Date: Tue, 13 Aug 2013 16:03:10 -0400
Subject: [Freeswitch-users] freeswitch startup with lots of errors while
	executing db queries
In-Reply-To: <CAE2+g+KFfRpvTg-7MMFRYJ_1DrBb4MJ_Yy7t-Na+M+kwHFVjnw@mail.gmail.com>
References: <CAE2+g++zo6AFuVbuaBNN1Er_Zw=Qzi+MVbGLEpG9gCh4625cQw@mail.gmail.com>
	<CAE2+g+KFfRpvTg-7MMFRYJ_1DrBb4MJ_Yy7t-Na+M+kwHFVjnw@mail.gmail.com>
Message-ID: <55AEEBB4-F849-4A36-A38A-C7C75BA45C26@jerris.com>

Yeah.. There is no portable way to check if an index already exists.. so we just create them every time and its a no-op error.

On Aug 13, 2013, at 3:55 PM, Nuno Reis <nreis at wavecom.pt> wrote:

> Hi guys.
> 
> I've just compiled the latest v1.2stable branch with native postgres support enabled and now everytime FS startsup i see a lot of errors while it tries i belive to create the FS database when it's already there:
> 
> see pb 21294 for a more complete insight on this issue.
> 
> I think this is normal right?
> 
> BR,
> --
> 

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From itsme.kunnu at gmail.com  Wed Aug 14 00:10:05 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Wed, 14 Aug 2013 01:40:05 +0530
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <72BF927D-CDAE-4682-B7A3-B8B81F969260@jerris.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CAMvhjJu8H0Y25Gd1qtJ0kEb9eiu_D56+rZVJy_zbHV7mEzMLcA@mail.gmail.com>
	<72BF927D-CDAE-4682-B7A3-B8B81F969260@jerris.com>
Message-ID: <CAMvhjJszr=ULpXe=D+do19KniQN5SEW-bXfwgT=3W+=Rhxsxkg@mail.gmail.com>

ya sure. which log ?



On Wed, Aug 14, 2013 at 1:32 AM, Michael Jerris <mike at jerris.com> wrote:

> Would you like to share the log so someone can look at it?
>
> On Aug 13, 2013, at 3:50 AM, Ashish Mishra <itsme.kunnu at gmail.com> wrote:
>
> Hello Peter...i set the fs cli to debugging by typing /log 7 and then
> retried to connect...but still the same result...
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 khorsmann at gmail.com  Wed Aug 14 00:27:25 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Tue, 13 Aug 2013 22:27:25 +0200
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CAMvhjJszr=ULpXe=D+do19KniQN5SEW-bXfwgT=3W+=Rhxsxkg@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CAMvhjJu8H0Y25Gd1qtJ0kEb9eiu_D56+rZVJy_zbHV7mEzMLcA@mail.gmail.com>
	<72BF927D-CDAE-4682-B7A3-B8B81F969260@jerris.com>
	<CAMvhjJszr=ULpXe=D+do19KniQN5SEW-bXfwgT=3W+=Rhxsxkg@mail.gmail.com>
Message-ID: <CAFArqsbP6iHF=2dQHcxXv9+fe+sBof9FwzjsbZhrnC8qhbwpKA@mail.gmail.com>

Hi Ashish.

He means the /var/log/freeswitch.log that should record the fs_cli stuff.
Or copy-n-paste the stuff you see on the fs_cli. Use
http://pastebin.freeswitch.org (username/password is in the login window)
and choose Syntax "freeswitch log". Then paste and mail us the url you got.


2013/8/13 Ashish Mishra <itsme.kunnu at gmail.com>

> ya sure. which log ?
>
>
>
> --
Mit freundlichen Gr??en
*Karsten Horsmann*
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From anthony.minessale at gmail.com  Wed Aug 14 01:02:00 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 13 Aug 2013 16:02:00 -0500
Subject: [Freeswitch-users] webrtc INCOMPATIBLE_DESTINATION
In-Reply-To: <CAK59s4yH0-B8nxK30k7TBgDgivDwJzPfDTeE0kOXiuKKdrFvXQ@mail.gmail.com>
References: <CAK59s4y6YHecUr-Te-TMvJoS=ASxthL8FCuYmNYhEmkfQioFmQ@mail.gmail.com>
	<CAKbxfG96LTXzrB92TGNEkFvmTU9bZz1WX=VFK9p+a_Hra9ZDBg@mail.gmail.com>
	<CAK59s4yjYoEDVJJyEx2CE0z=tqn_aSNvg76TXBVrFUPR3as=gQ@mail.gmail.com>
	<CAK59s4yH0-B8nxK30k7TBgDgivDwJzPfDTeE0kOXiuKKdrFvXQ@mail.gmail.com>
Message-ID: <CAKbxfG-qYZcVGr80K8x3jO4V0mfQfjcDKcTFMeWDt1MOMKUDFw@mail.gmail.com>

JSSIP does not work with Firefox yet.


On Tue, Aug 6, 2013 at 12:14 PM, Javier Menendez
<menendez.garcia at gmail.com>wrote:

> After trying and trying I found out it is a compatibility version with the
> browser.
>
> Using Chrome 26 I can call from jssip client but not to jssip client
> Using Chrom 28 I can call both from and to jssip client
> Using Firefox 23 I can't do anything!
>
> is there anything I can do in freeswitch to improve compatibility? any
> experience?
>
>
> On Tue, Aug 6, 2013 at 3:41 PM, Javier Menendez <menendez.garcia at gmail.com
> > wrote:
>
>> No luck :(
>>
>> EXECUTE sofia/internal/asterisk at 192.168.90.16export(nolocal:absolute_codec_string=PCMA,PCMU)
>> 2013-08-06 15:37:47.312306 [DEBUG] switch_channel.c:1222 EXPORT
>> (export_vars) (REMOTE ONLY) [absolute_codec_string]=[PCMA,PCMU]
>> EXECUTE sofia/internal/asterisk at 192.168.90.16bridge({absolute_codec_string=PCMA}sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid
>> ;transport=ws;fs_nat=yes;fs_path=sip%3A4st0031l%40PUBLIC_IP%3A38333%3Btransport%3Dws)
>> 2013-08-06 15:37:47.312306 [DEBUG] switch_channel.c:1176 sofia/internal/
>> asterisk at 192.168.90.16 EXPORTING[export_vars]
>> [absolute_codec_string]=[PCMA,PCMU] to event
>> 2013-08-06 15:37:47.312306 [DEBUG] switch_ivr_originate.c:2050 Parsing
>> global variables
>> 2013-08-06 15:37:47.312306 [DEBUG] switch_event.c:1615 Parsing variable
>> [absolute_codec_string]=[PCMA]
>> 2013-08-06 15:37:47.312306 [NOTICE] switch_channel.c:1030 New Channel
>> sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid[617a9f0e-fe9d-11e2-b1fd-b761e46dca72]
>> 2013-08-06 15:37:47.312306 [DEBUG] mod_sofia.c:4420
>> (sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid) State Change CS_NEW ->
>> CS_INIT
>> 2013-08-06 15:37:47.312306 [DEBUG] switch_core_session.c:1341 Send signal
>> sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid [BREAK]
>> 2013-08-06 15:37:47.312306 [DEBUG] switch_core_state_machine.c:416
>> (sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid) Running State Change
>> CS_INIT
>> 2013-08-06 15:37:47.312306 [DEBUG] switch_core_state_machine.c:455
>> (sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid) State INIT
>> 2013-08-06 15:37:47.312306 [DEBUG] mod_sofia.c:87
>> sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid SOFIA INIT
>> 2013-08-06 15:37:47.312306 [DEBUG] switch_core_media.c:681 Set Local Key
>> [1 AES_CM_128_HMAC_SHA1_80 inline:SvDPyP7iDr8+xmeRvVRrrqHinvGB/Ht8+VArR6Ro]
>> 2013-08-06 15:37:47.312306 [DEBUG] switch_core_media.c:681 Set Local Key
>> [1 AES_CM_128_HMAC_SHA1_80 inline:wUFqlClOJwLZaHzE+QlEQfVQEx979Gx4e7BCbl56]
>> 2013-08-06 15:37:47.312306 [DEBUG] sofia_glue.c:1191
>> sip:4st0031l at PUBLIC_IP:38333;transport=ws Setting proxy route to
>> sofia/webrtc/sip:4st0031l at vlsoojkf29mg.invalid
>> 2013-08-06 15:37:47.312306 [DEBUG] sofia_glue.c:1220 Local SDP:
>> v=0
>> o=FreeSWITCH 1375775801 1375775802 IN IP4 PUBLIC_IP
>>
>> s=FreeSWITCH
>> c=IN IP4 PUBLIC_IP
>> t=0 0
>> a=msid-semantic: WMS B54oKgHfL2pj5wroyvgO0c1ghIMukBaL
>> m=audio 20466 RTP/SAVPF 8 101 13
>> a=rtpmap:101 telephone-event/8000
>> a=rtcp-mux
>> a=rtcp:20466 IN IP4 PUBLIC_IP
>> a=ssrc:3456367315 cname:w406axcLqrVlOdzc
>> a=ssrc:3456367315 msid:B54oKgHfL2pj5wroyvgO0c1ghIMukBaL a0
>> a=ssrc:3456367315 mslabel:B54oKgHfL2pj5wroyvgO0c1ghIMukBaL
>> a=ssrc:3456367315 label:B54oKgHfL2pj5wroyvgO0c1ghIMukBaLa0
>> a=ice-ufrag:zGQpJHvQg7bqzb2V
>> a=ice-pwd:5SS6gUIwcuIQy7TX
>> a=candidate:3772579260 1 udp 659136 PUBLIC_IP 20466 typ host generation 0
>> a=candidate:3772579260 2 udp 659136 PUBLIC_IP 20466 typ host generation 0
>> a=crypto:1 AES_CM_128_HMAC_SHA1_80
>> inline:SvDPyP7iDr8+xmeRvVRrrqHinvGB/Ht8+VArR6Ro
>> a=ptime:20
>> a=sendrecv
>>
>>
>>
>> On Tue, Aug 6, 2013 at 3:26 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> try prepending the bridge url with {absolute_codec_string=PCMA}
>>>
>>>
>>> On Tue, Aug 6, 2013 at 6:32 AM, Javier Menendez <
>>> menendez.garcia at gmail.com> wrote:
>>>
>>>> Hi,
>>>>
>>>> I am trying to make an outbound call to a webrtc softphone using jssip,
>>>> I initiate the call from an asterisk box :
>>>>
>>>> [Asterisk] -> [FS] ->[jssip]
>>>>
>>>> I always get an  INCOMPATIBLE_DESTINATION error, looking at the trace
>>>> logs I found out that the problem is the codec negotiation but I can not
>>>> make it work, AFAIK the call should use the ALAW codec as it is compatible
>>>> with all legs involved. this is the asterisk SDP
>>>>
>>>> 2013-08-06 13:16:50.892293 [DEBUG] sofia.c:5802 Remote SDP:
>>>> v=0
>>>> o=root 451068671 451068671 IN IP4 192.168.90.16
>>>> s=Asterisk PBX 1.8.16.0
>>>> c=IN IP4 192.168.90.16
>>>> t=0 0
>>>> m=audio 16126 RTP/AVP 8 101
>>>> a=rtpmap:8 PCMA/8000
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=fmtp:101 0-16
>>>> a=ptime:20
>>>>
>>>> And this is the SDP from FS wich is sent to the jssip client
>>>>
>>>> 2013-08-06 13:16:50.912323 [DEBUG] sofia_glue.c:1220 Local SDP:
>>>> v=0
>>>> o=FreeSWITCH 1375766038 1375766039 IN IP4 PUBLIC_IP
>>>> s=FreeSWITCH
>>>> c=IN IP4 PUBLIC_IP
>>>> t=0 0
>>>> a=msid-semantic: WMS 9hJmtW2hnx2uG5S2obryUBUKGF35tgps
>>>> m=audio 21772 RTP/SAVPF 8 0 101 13
>>>> a=rtpmap:101 telephone-event/8000
>>>> a=rtcp-mux
>>>> a=rtcp:21772 IN IP4 PUBLIC_IP
>>>> a=ssrc:3456358538 cname:oZFIMxPZUIkrD3jS
>>>> a=ssrc:3456358538 msid:9hJmtW2hnx2uG5S2obryUBUKGF35tgps a0
>>>> a=ssrc:3456358538 mslabel:9hJmtW2hnx2uG5S2obryUBUKGF35tgps
>>>> a=ssrc:3456358538 label:9hJmtW2hnx2uG5S2obryUBUKGF35tgpsa0
>>>> a=ice-ufrag:4BgP20qzvjuPqK7E
>>>> a=ice-pwd:ADcc0S2Lqc2T5cB0
>>>> a=candidate:3263618716 1 udp 659136 212.230.135.231 21772 typ host
>>>> generation 0
>>>> a=candidate:3263618716 2 udp 659136 212.230.135.231 21772 typ host
>>>> generation 0
>>>> a=crypto:1 AES_CM_128_HMAC_SHA1_80
>>>> inline:JbAOcaClBcoNLW6zjoucy5BU0mfS+UQqWcyYh9+7
>>>> a=ptime:20
>>>> a=sendrecv
>>>>
>>>> Shouldn't it include the PCMA codec? I hace tried to enable it in
>>>> configuration but it doesn't work, also tried to set it with
>>>>
>>>>   <action application="export"
>>>> data="nolocal:absolute_codec_string=PCMA,PCMU"/>
>>>>
>>>> before bridge, but no luck.
>>>>
>>>> Funny thing is that it is working if i innitiate the call from the
>>>> jssip client. [jssip] -> [FS] ->[Asterisk]
>>>>
>>>> Any clue?
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From mishehu at freeswitch.org  Wed Aug 14 01:54:28 2013
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Tue, 13 Aug 2013 16:54:28 -0500
Subject: [Freeswitch-users] Error in launching fs_cli
In-Reply-To: <CAMvhjJv3oNWNQG5soCd_TNAkaBLFxzmVzELt5JZ=Skgu2vM4mg@mail.gmail.com>
References: <CAMvhjJuoejKwyzsH=5JRQnQo_gZNXVZ3_60odmJw4oYpG-j=aQ@mail.gmail.com>
	<CAMvhjJuzub9PPoMx62oJoLtx1oHJ83daC1DfdCAp1dU57yBryA@mail.gmail.com>
	<CAMvhjJswbDp2iixsy8r1QsCRKiwRa4-RUmDv7j+KYM4NOucdqg@mail.gmail.com>
	<CAMvhjJtBONsXnrhRoDdoMTUPhHb1NjVRSJOsdbt5u+0tjqaBug@mail.gmail.com>
	<CAFiqYuneYGGujBH-a_hanzEQ72Rdiks5kfwntwz2QGp1YC9A7A@mail.gmail.com>
	<5209575C.9030203@freeswitch.org>
	<CAMvhjJuZqjssQ896TBxAFKXUrkrNb=60EJ9kKdkOfWK4VD3Rsg@mail.gmail.com>
	<CAFiqYumwEGA7zYmJPpQ51w6q_1WsQARht5Z2Gyutoqa4KuRt2w@mail.gmail.com>
	<CAMvhjJv3oNWNQG5soCd_TNAkaBLFxzmVzELt5JZ=Skgu2vM4mg@mail.gmail.com>
Message-ID: <520AAB14.7070803@freeswitch.org>

Actually, that's not entirely correct. I run FreeSWITCH in the 
foreground monitored under runit or djb daemontools all the time.  
FreeSWITCH needs to be running with mod_event_socket enabled and loaded 
in order for fs_cli to be able to connect to it.  Whether or not 
FreeSWITCH's pid has been forked to background not of any real importance.

-Yossi


On 08/13/2013 02:52 PM, Ashish Mishra wrote:
> the issue was that i was only running fs_cli on my machine without 
> running freeswitch. fs_cli works only when freeswitch is running in 
> background.
>  thank you
> Ashish Mishra
>
>
> On Tue, Aug 13, 2013 at 1:47 PM, Steven Ayre <steveayre at gmail.com 
> <mailto:steveayre at gmail.com>> wrote:
>
>     What was the issue?
>
>
>     On 13 August 2013 08:16, Ashish Mishra <itsme.kunnu at gmail.com
>     <mailto:itsme.kunnu at gmail.com>> wrote:
>
>         Thank you for your reply...i am now able to run fs_cli on my
>         ubuntu machine on which freeswitch is installed...:-)
>
>         On Aug 13, 2013 3:17 AM, "I put the Who? in Mishehu"
>         <mishehu at freeswitch.org <mailto:mishehu at freeswitch.org>> wrote:
>
>             I have the nagging feeling that Ashish isn't telling the
>             whole story here...
>
>             Ashish -  what happens if you are on the machine that is
>             running FreeSWITCH and you execute `telnet localhost 8021`
>             ?  Do you see something like this:
>
>             # telnet localhost 8021
>             Trying 127.0.0.1...
>             Connected to localhost.
>             Escape character is '^]'.
>             Content-Type: auth/request
>
>             If you are running with the standard configurations this
>             should work in this fashion.  If you have changed the
>             conf/autoload_configs/event_socket.conf.xml , then paste
>             those configurations. (If too lengthy, please post to
>             http://pastebin.freeswitch.org )
>
>             -Yossi
>
>             On 08/12/2013 09:44 AM, Steven Ayre wrote:
>>             Have you checked the logfile and netstat?
>>
>>
>>             On 12 August 2013 09:51, Ashish Mishra
>>             <itsme.kunnu at gmail.com <mailto:itsme.kunnu at gmail.com>> wrote:
>>
>>                 I reinstalled freeswitch but still i am getting the
>>                 same error :
>>                 fs_cli .c:1455 main() Error Connecting [Socket
>>                 Connection Error ] mod_socket is also loaded...i have
>>                 also flushed the firewall rules...pls help
>>
>>                 On Aug 6, 2013 2:30 AM, "Ashish Mishra"
>>                 <itsme.kunnu at gmail.com
>>                 <mailto:itsme.kunnu at gmail.com>> wrote:
>>
>>                     When i am trying to launch fs_cli on my ubuntu
>>                     12.04 machine (on which i have installed
>>                     freeswitch) it gives me the following error :
>>                     fs_cli .c:1455 main() Error Connecting [Socket
>>                     Connection Error ]
>>
>>
>>                 _________________________________________________________________________
>>                 Professional FreeSWITCH Consulting Services:
>>                 consulting at freeswitch.org
>>                 <mailto:consulting at freeswitch.org>
>>                 http://www.freeswitchsolutions.com
>>
>>                 FreeSWITCH-powered IP PBX: The CudaTel Communication
>>                 Server
>>                 
>>
>>                 Official FreeSWITCH Sites
>>                 http://www.freeswitch.org
>>                 http://wiki.freeswitch.org
>>                 http://www.cluecon.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
>>
>>
>>
>>
>>             _________________________________________________________________________
>>             Professional FreeSWITCH Consulting Services:
>>             consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>             http://www.freeswitchsolutions.com
>>
>>             
>>             
>>
>>             Official FreeSWITCH Sites
>>             http://www.freeswitch.org
>>             http://wiki.freeswitch.org
>>             http://www.cluecon.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
>
>
>             _________________________________________________________________________
>             Professional FreeSWITCH Consulting Services:
>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>             http://www.freeswitchsolutions.com
>
>             
>             
>
>             Official FreeSWITCH Sites
>             http://www.freeswitch.org
>             http://wiki.freeswitch.org
>             http://www.cluecon.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
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.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
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 adahary at gmail.com  Wed Aug 14 02:48:43 2013
From: adahary at gmail.com (adahary)
Date: Tue, 13 Aug 2013 15:48:43 -0700 (PDT)
Subject: [Freeswitch-users] ZRTP Secure session over local FS and remote FS
	gateway
Message-ID: <1376434123879-7593875.post@n2.nabble.com>


My CSipSimple ZRTP endpoints register on FS1 and FS2 and can connect each
other over ZRTP secured links.
I have one FS_gw that functioning as a centralized media server for
voicemail and conference server.

any endpoint from FS1&2 can interconnect to FS_gw for vm and conference
services but cannot establish ZRTP session.
I have tried to setup FS1/2/gw with variation of ZRTP setups (proxy-media
true/false) with no help.

How to establish a secured ZRTP connection between an endpoint (on local
FS1) and vm/conference service (remote FS_gw)?
endpoint -> FS1 -> Internet -> FS_gw (vm, conference).

thanks    



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ZRTP-Secure-session-over-local-FS-and-remote-FS-gateway-tp7593875.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From eidevm5 at gmail.com  Wed Aug 14 05:42:30 2013
From: eidevm5 at gmail.com (Peter)
Date: Wed, 14 Aug 2013 11:42:30 +1000
Subject: [Freeswitch-users] Establishing SRTP from SBC to endpoint
In-Reply-To: <CAAYhoXLzsB-o=5VNBaRsR9xdS8bm=j5zKxe8ideg5-jmKU=fKw@mail.gmail.com>
References: <CABkS=xqD0mukebja+w3aurrhcUJ4Gp=DcOpkh+BDnCpOKr32CA@mail.gmail.com>
	<CAAYhoXLzsB-o=5VNBaRsR9xdS8bm=j5zKxe8ideg5-jmKU=fKw@mail.gmail.com>
Message-ID: <CABkS=xrLBX2A1ZLO9gx7mW2yQ2e9LTjyDaNOqJXH2DFADS4KYg@mail.gmail.com>

Hi Carlos.

Didn't realise rtp_secure_media existed.  After searching I saw:

https://wiki.freeswitch.org/wiki/Release_Notes#rtp_secure_media_.28was_sip_secure_media.29

which says it was introduced in 1.2.9

However, it's a little ambiguous as to whether sip_secure_media was
deprecated.

Anyway, I tried using rtp_secure_media instead, but I still can't get SRTP
working.


I did some testing with some other SIP clients.   In particular,
csipsimple.  On the client, if I set SRTP to be optional, the media stream
uses RTP.   However, if I set SRTP to be mandatory, when I try to call it,
Freeswitch receives:

   SIP/2.0 488 Not Acceptable Here

Which seems to indicate that something is not is not right with the SRTP
setup.

There's a full debug from the FS1 (the freeswitch server where the
csipsimple client is registered to) at:

http://pastebin.freeswitch.org/21295

Note in the debug I have sdp_secure_savp_only set to true.   I've tried
disabling this setting, but get the same result.

Thanks

Peter





On Tue, Aug 13, 2013 at 11:06 PM, Carlos Flor <jackal at cybershroud.net>wrote:

> Try using rtp_secure_media=true instead of sip_secure_media.  If you are
> trying to set it on the b-leg, you probably want to use export instead of
> set, or use nolocal:rtp_secure_media.
>
> Hope that helps.
>
>
> On Mon, Aug 12, 2013 at 10:26 PM, Peter <eidevm5 at gmail.com> wrote:
>
>> In my environment, I have the following (simplified) setup:
>>
>> FS1  ----  FS SBC ---  FS2
>>
>> Phones registered to FS1 (100x) use TLS/SRTP and phones registered to FS2
>> (200x) use SIP/RTP
>>
>> FS1 has inbound-bypass-media set to true to allow SRTP peer to peer and
>> direct to the SBC.
>>
>> If I make an inbound call (eg: 1000 to 2000), SRTP is correctly
>> established between the phone and SBC with RTP on the other side of the SBC
>> to the internal phone.
>>
>> However, when I try it the other way, I can't get SRTP established from
>> the SBC to the external phone.
>>
>> I've been using https://wiki.freeswitch.org/wiki/Secure_RTP as a guide.
>>
>> I've even tried explicitly setting sip_secure_media to true on the SBC
>> and FS1.
>>
>> The dialplan on the SBC has:
>>
>>   <extension name="outgoing">
>>         <condition field="destination_number"
>> expression="^(10[0-9][0-9])$">
>>             <action application="set" data="sip_secure_media=true"/>
>>             <action application="bridge" data="sofia/external/${
>> destination_number}@10.1.1.204"/>
>>         </condition>
>>   </extension>
>>
>>
>> And on FS1, the dialplan has:
>>
>>    <extension name="Local-Numbers">
>>       <condition field="destination_number" expression="^(10[01][0-9])$">
>>         <action application="export" data="dialed_extension=$1"/>
>>         <action application="set" data="sip_secure_media=true"/>
>>         <action application="bridge" data="user/${dialed_extension}@
>> ${domain_name}"/>
>>       </condition>
>>     </extension>
>>
>>
>> Note that I've been testing this against two phones with SRTP enabled,
>> but only one that is using TLS.  I get the same result calling each phone.
>>
>> On a related point, what it the step required for a TLS connection from
>> the SBC to the phone?   I'm assume the phone just needs the CA cert from
>> the SBC.  Correct?
>>
>> Any information as to where I'm going wrong will be gratefully accepted.
>>
>> Thanks
>>
>> Peter
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Wed Aug 14 05:46:03 2013
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 13 Aug 2013 18:46:03 -0700
Subject: [Freeswitch-users] UniMRCP Error With Vestec ASR
Message-ID: <CAKzWOxUM2UxjRScJGSkcW=s+5eNZTyh=q_RS8dDZMFob3Em36w@mail.gmail.com>

Hey gang,

I'm having an issue trying to get Vestec ASR working with mod_unimrcp. I've
got mod_unimrcp set to use port 1554 which is the default port for Vestec's
MRCP server. I keep getting this error:


   1. 2013-08-13 18:35:58.167224 [WARNING] rtsp_client.c:386 () Failed to
   Connect to RTSP Server 192.168.1.79:1554
   2. 2013-08-13 18:35:58.167224 [DEBUG] apt_poller_task.c:227 () Wait for
   Messages [unimrcpserver-mrcp1]
   3. 2013-08-13 18:35:58.167224 [DEBUG] apt_task.c:278 () Process
Message [MRCP
   Client] [1;0]
   4. 2013-08-13 18:35:58.167224 [INFO] mrcp_client_session.c:142
(ASR-8)Receive Answer 0x7f973c1288d0 <new>
   [c:0 a:0 v:0]
   5. 2013-08-13 18:35:58.167224 [DEBUG] apt_task.c:229 () Signal Message
   to [MediaEngine] [1;0]
   6. 2013-08-13 18:35:58.167224 [DEBUG] apt_consumer_task.c:97 () Wait for
   Messages [MRCP Client]
   7. 2013-08-13 18:35:58.187262 [DEBUG] apt_task.c:278 () Process Message [
   MediaEngine] [1;0]
   8. 2013-08-13 18:35:58.187262 [DEBUG] apt_task.c:278 () Process
Message [MRCP
   Client] [3;0]
   9. 2013-08-13 18:35:58.187262 [INFO] mrcp_client_session.c:457
(ASR-8)Raise App Response 0x7f973c1288d0 <new>
   [2] FAILURE [1]
   10. 2013-08-13 18:35:58.187262 [ERR] mod_unimrcp.c:1889
(ASR-8)RECOGNIZER channel error!

Full call log is here: http://pastebin.freeswitch.org/21296

Any ideas on where to look for the culprit?

-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From eidevm5 at gmail.com  Wed Aug 14 05:46:49 2013
From: eidevm5 at gmail.com (Peter)
Date: Wed, 14 Aug 2013 11:46:49 +1000
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
Message-ID: <CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>

Connect to the FS console with fs_cli and do

sofia logging all 5

then try to register your softphone.    If you don't even see a
registration attempt on the FS console, it means you have a network problem.

If the network isn't a problem, the debug output should give you a clue as
to what the problem is.



On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Thank you Peter...you mean to say that i should first turn on the fs_cli
> and then retry to connect the softphone...??? I would also like to remind
> you that fs_cli and freeswitch are running on the same machine and i am
> trying to connect thru a network cable my windows 8 pc that has the
> softphone on it...
> Regards
> Ashish Mishra
> On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:
>
>> Have you turned on debugging from the FS cli and seen if any registration
>> requests come in?
>>
>>
>>
>> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> I installed freeswitch on my ubuntu 12.04 machine...but when i am trying
>>> to connect the softphone installed on my windows 8 pc with ubuntu machine
>>> the softphone gives me an error that account not enabled...i have used a
>>> network cable to connect the two machines...also the firewall in both the
>>> machines is disabled...
>>> Kindly help..
>>>
>>> Ashish Mishra
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 cmrienzo at gmail.com  Wed Aug 14 06:07:48 2013
From: cmrienzo at gmail.com (Christopher Rienzo)
Date: Tue, 13 Aug 2013 22:07:48 -0400
Subject: [Freeswitch-users] UniMRCP Error With Vestec ASR
In-Reply-To: <CAKzWOxUM2UxjRScJGSkcW=s+5eNZTyh=q_RS8dDZMFob3Em36w@mail.gmail.com>
References: <CAKzWOxUM2UxjRScJGSkcW=s+5eNZTyh=q_RS8dDZMFob3Em36w@mail.gmail.com>
Message-ID: <CAD-qRkgPefWJjtBAbMT5Pi4v1i4E8zFcpxqpcjdZDmKAW65hFQ@mail.gmail.com>

Make sure the vasre-mrcp server is running and is listening on
192.168.1.79:1554.  When I installed, it was listening on 127.0.0.1 and I
had to change the config in /opt/Vestec/mrcp/conf/mrcpserver.xml.

If iptables is running, make sure incoming connections to port 1554 is
allowed (or shut it off) and RTP can be sent to the port range defined in
the mrcpserver.xml file.




On Tue, Aug 13, 2013 at 9:46 PM, Michael Collins <msc at freeswitch.org> wrote:

> Hey gang,
>
> I'm having an issue trying to get Vestec ASR working with mod_unimrcp.
> I've got mod_unimrcp set to use port 1554 which is the default port for
> Vestec's MRCP server. I keep getting this error:
>
>
>    1. 2013-08-13 18:35:58.167224 [WARNING] rtsp_client.c:386 () Failed to
>    Connect to RTSP Server 192.168.1.79:1554
>    2. 2013-08-13 18:35:58.167224 [DEBUG] apt_poller_task.c:227 () Wait
>    for Messages [unimrcpserver-mrcp1]
>    3. 2013-08-13 18:35:58.167224 [DEBUG] apt_task.c:278 () Process
>    Message [MRCP Client] [1;0]
>    4. 2013-08-13 18:35:58.167224 [INFO] mrcp_client_session.c:142 (ASR-8)Receive Answer 0x7f973c1288d0 <new>
>    [c:0 a:0 v:0]
>    5. 2013-08-13 18:35:58.167224 [DEBUG] apt_task.c:229 () Signal Message
>    to [MediaEngine] [1;0]
>    6. 2013-08-13 18:35:58.167224 [DEBUG] apt_consumer_task.c:97 () Wait
>    for Messages [MRCP Client]
>    7. 2013-08-13 18:35:58.187262 [DEBUG] apt_task.c:278 () Process
>    Message [MediaEngine] [1;0]
>    8. 2013-08-13 18:35:58.187262 [DEBUG] apt_task.c:278 () Process
>    Message [MRCP Client] [3;0]
>    9. 2013-08-13 18:35:58.187262 [INFO] mrcp_client_session.c:457 (ASR-8)Raise App Response 0x7f973c1288d0 <new>
>    [2] FAILURE [1]
>    10. 2013-08-13 18:35:58.187262 [ERR] mod_unimrcp.c:1889 (ASR-8)RECOGNIZER channel error!
>
> Full call log is here: http://pastebin.freeswitch.org/21296
>
> Any ideas on where to look for the culprit?
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Wed Aug 14 09:26:26 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Wed, 14 Aug 2013 10:56:26 +0530
Subject: [Freeswitch-users] Connect fs_cli and freeswitch console
Message-ID: <CAMvhjJvn+tQqU2kB9eXhguBia6ODkQ016PF+0wYvm_wcGY8u=A@mail.gmail.com>

How should i connect fs_cli and freeswitch console if both of them are
running on the same machine...???
Thank you
Ashish Mishra
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From itsme.kunnu at gmail.com  Wed Aug 14 09:28:36 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Wed, 14 Aug 2013 10:58:36 +0530
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
Message-ID: <CAMvhjJsFd3Y7n8WPtDHMPWUOHTi6VDXPtx_WO4zUxnrvbV6oTA@mail.gmail.com>

How should i connect FS console and fs_cli...i am running both
simultaneously on the same machine...???
On Aug 14, 2013 7:24 AM, "Peter" <eidevm5 at gmail.com> wrote:

> Connect to the FS console with fs_cli and do
>
> sofia logging all 5
>
> then try to register your softphone.    If you don't even see a
> registration attempt on the FS console, it means you have a network problem.
>
> If the network isn't a problem, the debug output should give you a clue as
> to what the problem is.
>
>
>
> On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Thank you Peter...you mean to say that i should first turn on the fs_cli
>> and then retry to connect the softphone...??? I would also like to remind
>> you that fs_cli and freeswitch are running on the same machine and i am
>> trying to connect thru a network cable my windows 8 pc that has the
>> softphone on it...
>> Regards
>> Ashish Mishra
>> On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:
>>
>>> Have you turned on debugging from the FS cli and seen if any
>>> registration requests come in?
>>>
>>>
>>>
>>> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> I installed freeswitch on my ubuntu 12.04 machine...but when i am
>>>> trying to connect the softphone installed on my windows 8 pc with ubuntu
>>>> machine the softphone gives me an error that account not enabled...i have
>>>> used a network cable to connect the two machines...also the firewall in
>>>> both the machines is disabled...
>>>> Kindly help..
>>>>
>>>> Ashish Mishra
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mishehu at freeswitch.org  Wed Aug 14 09:41:41 2013
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Wed, 14 Aug 2013 00:41:41 -0500
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CAMvhjJsFd3Y7n8WPtDHMPWUOHTi6VDXPtx_WO4zUxnrvbV6oTA@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsFd3Y7n8WPtDHMPWUOHTi6VDXPtx_WO4zUxnrvbV6oTA@mail.gmail.com>
Message-ID: <520B1895.9020208@freeswitch.org>

Try to do it and see what happens. What's the worst that could happen?  :-)

-Yossi


On 08/14/2013 12:28 AM, Ashish Mishra wrote:
>
> How should i connect FS console and fs_cli...i am running both 
> simultaneously on the same machine...???
>
> On Aug 14, 2013 7:24 AM, "Peter" <eidevm5 at gmail.com 
> <mailto:eidevm5 at gmail.com>> wrote:
>
>     Connect to the FS console with fs_cli and do
>
>     sofia logging all 5
>
>     then try to register your softphone.    If you don't even see a
>     registration attempt on the FS console, it means you have a
>     network problem.
>
>     If the network isn't a problem, the debug output should give you a
>     clue as to what the problem is.
>
>
>
>     On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra
>     <itsme.kunnu at gmail.com <mailto:itsme.kunnu at gmail.com>> wrote:
>
>         Thank you Peter...you mean to say that i should first turn on
>         the fs_cli and then retry to connect the softphone...??? I
>         would also like to remind you that fs_cli and freeswitch are
>         running on the same machine and i am trying to connect thru a
>         network cable my windows 8 pc that has the softphone on it...
>         Regards
>         Ashish Mishra
>
>         On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com
>         <mailto:eidevm5 at gmail.com>> wrote:
>
>             Have you turned on debugging from the FS cli and seen if
>             any registration requests come in?
>
>
>
>             On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra
>             <itsme.kunnu at gmail.com <mailto:itsme.kunnu at gmail.com>> wrote:
>
>                 I installed freeswitch on my ubuntu 12.04
>                 machine...but when i am trying to connect the
>                 softphone installed on my windows 8 pc with ubuntu
>                 machine the softphone gives me an error that account
>                 not enabled...i have used a network cable to connect
>                 the two machines...also the firewall in both the
>                 machines is disabled...
>                 Kindly help..
>
>                 Ashish Mishra
>
>
>                 _________________________________________________________________________
>                 Professional FreeSWITCH Consulting Services:
>                 consulting at freeswitch.org
>                 <mailto:consulting at freeswitch.org>
>                 http://www.freeswitchsolutions.com
>
>                 FreeSWITCH-powered IP PBX: The CudaTel Communication
>                 Server
>                 
>
>                 Official FreeSWITCH Sites
>                 http://www.freeswitch.org
>                 http://wiki.freeswitch.org
>                 http://www.cluecon.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
>
>
>
>             _________________________________________________________________________
>             Professional FreeSWITCH Consulting Services:
>             consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>             http://www.freeswitchsolutions.com
>
>             
>             
>
>             Official FreeSWITCH Sites
>             http://www.freeswitch.org
>             http://wiki.freeswitch.org
>             http://www.cluecon.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
>
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.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
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Wed Aug 14 10:38:25 2013
From: eidevm5 at gmail.com (Peter)
Date: Wed, 14 Aug 2013 16:38:25 +1000
Subject: [Freeswitch-users] Connect fs_cli and freeswitch console
In-Reply-To: <CAMvhjJvn+tQqU2kB9eXhguBia6ODkQ016PF+0wYvm_wcGY8u=A@mail.gmail.com>
References: <CAMvhjJvn+tQqU2kB9eXhguBia6ODkQ016PF+0wYvm_wcGY8u=A@mail.gmail.com>
Message-ID: <CABkS=xq3xZO+0JajJL4CvJA9dSmdTp2o1r4A2XJ4eXz0w_HE3Q@mail.gmail.com>

On the server running freeswitch, just type in

fs_cli

from the command line.


On Wed, Aug 14, 2013 at 3:26 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> How should i connect fs_cli and freeswitch console if both of them are
> running on the same machine...???
> Thank you
> Ashish Mishra
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Wed Aug 14 12:34:56 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Wed, 14 Aug 2013 14:04:56 +0530
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
Message-ID: <CAMvhjJsqYqqnCNU_Z_Ea6MDwnUYW-rSi5xpbk9kY--G-HJpMNA@mail.gmail.com>

Hi peter can you tell me how to debug the output...???
On Aug 14, 2013 7:24 AM, "Peter" <eidevm5 at gmail.com> wrote:

> Connect to the FS console with fs_cli and do
>
> sofia logging all 5
>
> then try to register your softphone.    If you don't even see a
> registration attempt on the FS console, it means you have a network problem.
>
> If the network isn't a problem, the debug output should give you a clue as
> to what the problem is.
>
>
>
> On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Thank you Peter...you mean to say that i should first turn on the fs_cli
>> and then retry to connect the softphone...??? I would also like to remind
>> you that fs_cli and freeswitch are running on the same machine and i am
>> trying to connect thru a network cable my windows 8 pc that has the
>> softphone on it...
>> Regards
>> Ashish Mishra
>> On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:
>>
>>> Have you turned on debugging from the FS cli and seen if any
>>> registration requests come in?
>>>
>>>
>>>
>>> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> I installed freeswitch on my ubuntu 12.04 machine...but when i am
>>>> trying to connect the softphone installed on my windows 8 pc with ubuntu
>>>> machine the softphone gives me an error that account not enabled...i have
>>>> used a network cable to connect the two machines...also the firewall in
>>>> both the machines is disabled...
>>>> Kindly help..
>>>>
>>>> Ashish Mishra
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Wed Aug 14 12:44:27 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Wed, 14 Aug 2013 14:14:27 +0530
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
Message-ID: <CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>

Also the command that you had mentioned "sofia logging all 5" gives me an
error message :
Unknown command [logging]
On Aug 14, 2013 7:24 AM, "Peter" <eidevm5 at gmail.com> wrote:

> Connect to the FS console with fs_cli and do
>
> sofia logging all 5
>
> then try to register your softphone.    If you don't even see a
> registration attempt on the FS console, it means you have a network problem.
>
> If the network isn't a problem, the debug output should give you a clue as
> to what the problem is.
>
>
>
> On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Thank you Peter...you mean to say that i should first turn on the fs_cli
>> and then retry to connect the softphone...??? I would also like to remind
>> you that fs_cli and freeswitch are running on the same machine and i am
>> trying to connect thru a network cable my windows 8 pc that has the
>> softphone on it...
>> Regards
>> Ashish Mishra
>> On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:
>>
>>> Have you turned on debugging from the FS cli and seen if any
>>> registration requests come in?
>>>
>>>
>>>
>>> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> I installed freeswitch on my ubuntu 12.04 machine...but when i am
>>>> trying to connect the softphone installed on my windows 8 pc with ubuntu
>>>> machine the softphone gives me an error that account not enabled...i have
>>>> used a network cable to connect the two machines...also the firewall in
>>>> both the machines is disabled...
>>>> Kindly help..
>>>>
>>>> Ashish Mishra
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sravi123 at yahoo.com  Wed Aug 14 12:45:50 2013
From: sravi123 at yahoo.com (Ravi)
Date: Wed, 14 Aug 2013 14:15:50 +0530
Subject: [Freeswitch-users] How does Freeswitch process incoming calls
In-Reply-To: <CAKzWOxWFeL83ymLVhWHyRjgjSY8WDOxjOGokXNQwq0xZK_uSMg@mail.gmail.com>
References: <520A744D.6020301@yahoo.com>
	<CAKzWOxWFeL83ymLVhWHyRjgjSY8WDOxjOGokXNQwq0xZK_uSMg@mail.gmail.com>
Message-ID: <520B43BE.2050405@yahoo.com>

Thanks a ton, Michael. Your clarification was key in resolving this 
issue. The carrier is sending only the last three digits. I configured 
the dialplan as you had mentioned in the mail mapping for the last three 
digits, and I am able to successfully receive calls. I am having some 
issues with the outbound calls, but I will post that in a separate thread.

Thanks again !!
Ravi


On 13/08/13 11:59 PM, Michael Collins wrote:
>
>
>
> On Tue, Aug 13, 2013 at 11:00 AM, Ravi <sravi123 at yahoo.com 
> <mailto:sravi123 at yahoo.com>> wrote:
>
>     Hello Everyone,
>
>     I am trying to configure a PRI connection. When I call from my
>     mobile number to the PRI landline number, I get the following log:
>     The call has never connected, I hear an engaged tone and it
>     disconnects.
>
>     Question 1: In the fourth line that is marked in red, the message
>     says Called No:[000]. The number that I actually dialed from my
>     mobile (7502029000) is +914274302000 <tel:%2B914274302000>. I am
>     not able to figure out what only three digits are displayed.
>
> It looks like the carrier is only sending you the last 3 digits of the 
> called DID.
>
>
>     Question 2: When freeswitch receives an inbound call - in what
>     sequence is the call processed ? What dialplans should I be
>     looking at?
>
> Once mod_freetdm figures out what the called number is it will try to 
> route it in whatever context is specified in your freetdm.conf.xml 
> file. In your case it goes to default as you can see by this log line:
>
> 2013-08-13 11:08:45.145709 [INFO] mod_dialplan_xml.c:558 Processing  
> <7502029000>->000 in context default
>
>     Question 3: How do I or where do I find the number that is being
>     called in freeswitch ?
>
> See my answer to question #2. Basically you need an extension with a 
> condition that matches 000, something like:
> <condition field="destination_number" expression="^000$>
>
> -MC
>
>
>
>     Any pointers or advise on how to go about this, is much appreciated.
>
>     Thanks.
>     Ravi
>
>
>     2013-08-13 11:08:45.145709 [DEBUG] ftmod_wanpipe.c:1400
>     [s1c31][1:16] Link status is 1
>     2013-08-13 11:08:45.145709 [INFO]
>     ftmod_sangoma_isdn_stack_rcv.c:96 [s1c9][1:9] Received SETUP
>     (suId:1 suInstId:0 spInstId:9)
>     2013-08-13 11:08:45.145709 [DEBUG]
>     ftmod_sangoma_isdn_stack_hndl.c:57 [s1c9][1:9] Processing SETUP
>     (suId:1 suInstId:0 spInstId:9)
>     2013-08-13 11:08:45.145709 [INFO]
>     *ftmod_sangoma_isdn_stack_hndl.c:153 [s1c9][1:9] Incoming call:
>     Called No:[000] Calling No:[7502029000]*
>     2013-08-13 11:08:45.145709 [DEBUG]
>     ftmod_sangoma_isdn_stack_hndl.c:209 [s1c9][1:9] Changed state from
>     DOWN to RING
>     2013-08-13 11:08:45.145709 [DEBUG] ftdm_state.c:541 [s1c9][1:9]
>     Executing state processor for RING
>     2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:655
>     [s1c9][1:9] processing state change to RING
>     2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:676
>     [s1c9][1:9] Sending incoming call from 7502029000 to 000 to FTDM core
>     2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:923
>     [s1c9][1:9] Completed state change from DOWN to RING in 0 ms
>     2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:2657 got clear
>     channel sig [START]
>     2013-08-13 11:08:45.145709 [DEBUG] ftmod_wanpipe.c:749 [s1c9][1:9]
>     Enabled DTMF events
>     2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:403 Set codec
>     PCMA 20ms
>     2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:1890 Connect
>     inbound channel FreeTDM/1:9/000
>     2013-08-13 11:08:45.145709 [NOTICE] switch_channel.c:1030 New
>     Channel FreeTDM/1:9/000 [9eb4ce34-03da-11e3-8bc5-0b625c405e53]
>     2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:2093
>     (FreeTDM/1:9/000) State Change CS_NEW -> CS_INIT
>     2013-08-13 11:08:45.145709 [DEBUG] switch_core_session.c:1341 Send
>     signal FreeTDM/1:9/000 [BREAK]
>     2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:416
>     (FreeTDM/1:9/000) Running State Change CS_INIT
>     2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:455
>     (FreeTDM/1:9/000) State INIT
>     2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:430
>     (FreeTDM/1:9/000) State Change CS_INIT -> CS_ROUTING
>     2013-08-13 11:08:45.145709 [DEBUG] switch_core_session.c:1341 Send
>     signal FreeTDM/1:9/000 [BREAK]
>     2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:455
>     (FreeTDM/1:9/000) State INIT going to sleep
>     2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:416
>     (FreeTDM/1:9/000) Running State Change CS_ROUTING
>     2013-08-13 11:08:45.145709 [DEBUG] switch_channel.c:2116
>     (FreeTDM/1:9/000) Callstate Change DOWN -> RINGING
>     2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:471
>     (FreeTDM/1:9/000) State ROUTING
>     2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:453
>     FreeTDM/1:9/000 CHANNEL ROUTING
>     2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:456 [s1c9][1:9]
>     Indicating PROCEED in state RING
>     2013-08-13 11:08:45.145709 [DEBUG] mod_freetdm.c:456 [s1c9][1:9]
>     Changed state from RING to PROCEED
>     2013-08-13 11:08:45.145709 [DEBUG] ftdm_state.c:541 [s1c9][1:9]
>     Executing state processor for PROCEED
>     2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:655
>     [s1c9][1:9] processing state change to PROCEED
>     2013-08-13 11:08:45.145709 [INFO]
>     ftmod_sangoma_isdn_stack_out.c:175 [s1c9][1:9] Sending PROCEED
>     (suId:1 suInstId:9 spInstId:9 dchan:1 ces:0)
>     2013-08-13 11:08:45.145709 [DEBUG] ftmod_sangoma_isdn.c:923
>     [s1c9][1:9] Completed state change from RING to PROCEED in 0 ms
>     2013-08-13 11:08:45.145709 [DEBUG] switch_core_state_machine.c:117
>     FreeTDM/1:9/000 Standard ROUTING
>     2013-08-13 11:08:45.145709 [INFO] mod_dialplan_xml.c:558
>     Processing  <7502029000>->000 in context default
>     Dialplan: FreeTDM/1:9/000 parsing [default->unloop] continue=false
>     Dialplan: FreeTDM/1:9/000 Regex (PASS) [unloop]
>     ${unroll_loops}(true) =~ /^true$/ break=on-false
>     Dialplan: FreeTDM/1:9/000 Regex (FAIL) [unloop]
>     ${sip_looped_call}() =~ /^true$/ break=on-false
>     Dialplan: FreeTDM/1:9/000 parsing [default->tod_example] continue=true
>     Dialplan: FreeTDM/1:9/000 Date/Time Match (PASS) [tod_example]
>     break=on-false
>     Dialplan: FreeTDM/1:9/000 Action set(open=true)
>     Dialplan: FreeTDM/1:9/000 parsing [default->holiday_example]
>     continue=true
>     Dialplan: FreeTDM/1:9/000 Date/TimeMatch (FAIL) [holiday_example]
>     break=on-false
>     Dialplan: FreeTDM/1:9/000 parsing [default->global-intercept]
>     continue=false
>     Dialplan: FreeTDM/1:9/000 Regex (FAIL) [global-intercept]
>     destination_number(000) =~ /^886$/ break=on-false
>     Dialplan: FreeTDM/1:9/000 parsing [default->group-intercept]
>     continue=false
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
>
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Wed Aug 14 13:04:27 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Wed, 14 Aug 2013 14:34:27 +0530
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
Message-ID: <CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>

I went for "sofia loglevel 5" instead of "sofia logging 5" but still no
progress in connection of softphone...:-(
Thanks
Ashish Mishra
On Aug 14, 2013 2:14 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:

> Also the command that you had mentioned "sofia logging all 5" gives me an
> error message :
> Unknown command [logging]
> On Aug 14, 2013 7:24 AM, "Peter" <eidevm5 at gmail.com> wrote:
>
>> Connect to the FS console with fs_cli and do
>>
>> sofia logging all 5
>>
>> then try to register your softphone.    If you don't even see a
>> registration attempt on the FS console, it means you have a network problem.
>>
>> If the network isn't a problem, the debug output should give you a clue
>> as to what the problem is.
>>
>>
>>
>> On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> Thank you Peter...you mean to say that i should first turn on the fs_cli
>>> and then retry to connect the softphone...??? I would also like to remind
>>> you that fs_cli and freeswitch are running on the same machine and i am
>>> trying to connect thru a network cable my windows 8 pc that has the
>>> softphone on it...
>>> Regards
>>> Ashish Mishra
>>> On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:
>>>
>>>> Have you turned on debugging from the FS cli and seen if any
>>>> registration requests come in?
>>>>
>>>>
>>>>
>>>> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>
>>>>> I installed freeswitch on my ubuntu 12.04 machine...but when i am
>>>>> trying to connect the softphone installed on my windows 8 pc with ubuntu
>>>>> machine the softphone gives me an error that account not enabled...i have
>>>>> used a network cable to connect the two machines...also the firewall in
>>>>> both the machines is disabled...
>>>>> Kindly help..
>>>>>
>>>>> Ashish Mishra
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 steveayre at gmail.com  Wed Aug 14 13:16:51 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 14 Aug 2013 10:16:51 +0100
Subject: [Freeswitch-users] Connect fs_cli and freeswitch console
In-Reply-To: <CAMvhjJvn+tQqU2kB9eXhguBia6ODkQ016PF+0wYvm_wcGY8u=A@mail.gmail.com>
References: <CAMvhjJvn+tQqU2kB9eXhguBia6ODkQ016PF+0wYvm_wcGY8u=A@mail.gmail.com>
Message-ID: <CAFiqYukxx8rR6x5o2m4c+5TGfUHi_6PeTS_MbCt0s+COpNfiZQ@mail.gmail.com>

fs_cli is just a client for the mod_event_socket server.

freeswitch foreground console (-c) is a separate console.

You can use both at the same time. Just run fs_cli from another terminal.


On 14 August 2013 06:26, Ashish Mishra <itsme.kunnu at gmail.com> wrote:

> How should i connect fs_cli and freeswitch console if both of them are
> running on the same machine...???
> Thank you
> Ashish Mishra
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 Aug 14 17:20:00 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Aug 2013 08:20:00 -0500
Subject: [Freeswitch-users] fsctl debug_sql (CRITICAL warnings/output)
In-Reply-To: <CAE2+g++Q5Jk=RwyK81SK+s5E5uxtrpjqzeYNhazKXtNoB-aY3w@mail.gmail.com>
References: <CAE2+g++Q5Jk=RwyK81SK+s5E5uxtrpjqzeYNhazKXtNoB-aY3w@mail.gmail.com>
Message-ID: <CAKbxfG_jHx+sySLRJ0V9JmX2GgqNEnLxB30xra00AnWm7PFtYw@mail.gmail.com>

Yes, the debug messages are crit level so they stand out.
On Aug 13, 2013 2:38 PM, "Nuno Reis" <nreis at wavecom.pt> wrote:

> Hi guys.
>
> I've just installed the new FS 1.2.12 with native postgres support enabled
> in core and I'm noticing that when i turn the debug_sql on with fsctl
> debug_sql command i have a lot of critical messages as bellow:
>
> 2013-08-13 18:02:37.229762 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
> [0|0|0|0]--[1]
> 2013-08-13 18:02:40.009898 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
> [0|0|0|0]--[2]
> 2013-08-13 18:02:41.229721 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
> [0|0|0|0]--[1]
> 2013-08-13 18:02:53.929745 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
> [0|0|0|0]--[2]
> 2013-08-13 18:02:56.289765 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
> [0|0|0|0]--[1]
> 2013-08-13 18:02:57.309725 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
> [0|0|0|0]--[2]
> 2013-08-13 18:02:57.729728 [CRIT] switch_core_sqldb.c:1885 sofia:external
> RUN QUEUE [0|0]--[5]
> 2013-08-13 18:02:57.729728 [CRIT] switch_core_sqldb.c:1885
> sofia:external-local RUN QUEUE [0|0]--[5]
> 2013-08-13 18:02:57.729728 [CRIT] switch_core_sqldb.c:1885
> sofia:internal-nmreis RUN QUEUE [0|0]--[5]
> 2013-08-13 18:02:57.849741 [CRIT] switch_core_sqldb.c:1885 sofia:internal
> RUN QUEUE [0|0]--[5]
> 2013-08-13 18:02:58.849750 [CRIT] switch_core_sqldb.c:1885
> sofia:nat-traversal RUN QUEUE [0|0]--[5]
> 2013-08-13 18:03:06.989721 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
> [0|0|0|0]--[4]
> 2013-08-13 18:03:08.949813 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
> [0|0|0|0]--[2]
> 2013-08-13 18:03:11.349756 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
> [0|0|0|0]--[1]
> 2013-08-13 18:03:17.389719 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
> [0|0|0|0]--[1]
> 2013-08-13 18:03:26.389718 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
> [0|0|0|0]--[1]
> 2013-08-13 18:03:27.829818 [CRIT] switch_core_sqldb.c:1885 sofia:external
> RUN QUEUE [0|0]--[5]
> 2013-08-13 18:03:27.829818 [CRIT] switch_core_sqldb.c:1885
> sofia:external-local RUN QUEUE [0|0]--[5]
> 2013-08-13 18:03:27.829818 [CRIT] switch_core_sqldb.c:1885
> sofia:internal-nmreis RUN QUEUE [0|0]--[5]
> 2013-08-13 18:03:27.929794 [CRIT] switch_core_sqldb.c:1885 sofia:internal
> RUN QUEUE [0|0]--[5]
> 2013-08-13 18:03:27.989786 [CRIT] switch_core_sqldb.c:1885 CORE RUN QUEUE
> [0|0|0|0]--[2]
> 2013-08-13 18:03:28.969712 [CRIT] switch_core_sqldb.c:1885
> sofia:nat-traversal RUN QUEUE [0|0]--[5]
>
> IS this normal? Everything seems to be fine though.
> Looking forward to hear from you.
>
> BR,
>
> --
>
> *Nuno Miguel Reis* | *Unified Communication** Systems*
> M. +351 913907481 | nreis at wavecom.pt
> WAVECOM-Solu??es R?dio, S.A.
> Cacia Park | Rua do Progresso, Lote 15
> 3800-639 AVEIRO | Portugal
> T. +351 309 700 225 | F. +351 234 919 191
> *GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>|
> www.wavecom.pt** <http://www.wavecom.pt/>*
>
> [image: Description: Description: WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php>
>
> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sravi123 at yahoo.com  Wed Aug 14 17:34:37 2013
From: sravi123 at yahoo.com (Ravi)
Date: Wed, 14 Aug 2013 06:34:37 -0700 (PDT)
Subject: [Freeswitch-users] Recording calls with Freeswitch - log says
	recorded, but no file found
Message-ID: <1376487277.65226.YahooMailNeo@web160503.mail.bf1.yahoo.com>

Hello Everyone !

I am trying to record calls received in an extension (1003). The dialplan is given below 


extension name="003">
??? <condition field="destination_number" expression="^003$">
??????? <action application="set" data="record_file_name={recordings_ dir}/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.wav" inli$
??????? <action application="record_session" data="{record_file_name}"/>
??????? <action application="bridge" data="user/1003"/>
??? </condition>
</extension>


When I call this extension from an extension 1006, this is what I get in the log. But when I go to the folder /usr/local/freeswitch/recordings/ there are no files at all. I am just wondering if this has got to do with any file permissions, or if I am missing something. Please help. 


Thanks.
Ravi


freeswitch at bfree-server> 2013-08-14 06:53:51.185709 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/1006 at 10.0.0.16 [2d3314f4-0480-11e3-9a2f-4d1cb7d8b5b4]
2013-08-14 06:53:51.245710 [INFO] mod_dialplan_xml.c:558 Processing Ravi <1006>->1003 in context default
2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *1 execute_extension::dx XML features
2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1006.2013-08-14-06-53-51.wav
2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *3 execute_extension::cf XML features
2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *4 execute_extension::att_xfer XML features
2013-08-14 06:53:51.245710 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/sip:1003 at 10.0.0.10:5060 [2d3d1a26-0480-11e3-9a4d-4d1cb7d8b5b4]
2013-08-14 06:53:51.445707 [NOTICE] sofia.c:5884 Ring-Ready sofia/internal/sip:1003 at 10.0.0.10:5060!
2013-08-14 06:53:51.445707 [INFO] switch_ivr_originate.c:1190 Sending early media
2013-08-14 06:53:51.445707 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1006 at 10.0.0.16!
2013-08-14 06:53:52.705700 [NOTICE] sofia.c:6547 Channel [sofia/internal/sip:1003 at 10.0.0.10:5060] has been answered
2013-08-14 06:53:52.725708 [NOTICE] switch_ivr_originate.c:3437 Channel [sofia/internal/1006 at 10.0.0.16] has been answered
2013-08-14 06:54:05.005696 [NOTICE] sofia.c:715 Hangup sofia/internal/sip:1003 at 10.0.0.10:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2013-08-14 06:54:05.025711 [NOTICE] switch_ivr_bridge.c:1575 Hangup sofia/internal/1006 at 10.0.0.16 [CS_EXECUTE] [NORMAL_CLEARING]
2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1567 Session 139 (sofia/internal/sip:1003 at 10.0.0.10:5060) Ended
2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1571 Close Channel sofia/internal/sip:1003 at 10.0.0.10:5060 [CS_DESTROY]
2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1567 Session 138 (sofia/internal/1006 at 10.0.0.16) Ended
2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1571 Close Channel sofia/internal/1006 at 10.0.0.16 [CS_DESTROY]
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From Adam.Lappe at qsc.de  Wed Aug 14 18:06:39 2013
From: Adam.Lappe at qsc.de (Lappe, Adam)
Date: Wed, 14 Aug 2013 16:06:39 +0200
Subject: [Freeswitch-users] SIP TLS Issues
Message-ID: <F1EBD7E7528EA446BB9898D22914AA3B3D947FEAD9@CGNMSG01.oc.qsc.de>

Hi all,

i am trying to configure FreeSWITCH to speak TLS with all Clients.
I followed the tutorial on http://wiki.freeswitch.com/wiki/SIP_TLS but I am still not sure what key / cert belongs in which file.

I have a SSL123 Thawte Wildcard Certificate.
Am I supposed to cat this cert + priv. key into agent.pem and the primary and secondary intermediate into the cafile.pem?

I did this and set the right permissions. The internal sofia profile on port 5061 (TLS) is RUNNING.

But no client (for example Polycom VVX1500) can register now.
If I set it TCP and Port 5060 (which is RUNNING as well) everything works fine.

Wireshark shows me the following

Client    ->         FS                    Client Hello
FS        ->         Client                Alert (Level Fatal, Description: Handshake Failure)

I also tested openssl s_client -connect (IP):5061 -showcerts but it only says:
CONNECTED(00000003)
139847050823328:error:14077410:SSL routines:SSL23_GET_SERVER_HELLO:sslv3 alert handshake failure:s23_clnt.c:724:
---
no peer certificate available
---
No client certificate CA names sent
---
SSL handshake has read 7 bytes and written 225 bytes
---
New, (NONE), Cipher is (NONE)
Secure Renegotiation IS NOT supported
Compression: NONE
Expansion: NONE
---

I guess the problem is the agent.pem and/or cafile.pem

agent.pem looks like this
-----BEGIN CERTIFICATE-----
(Thawte SSL123 Wildcard Web Certificate)
-----END CERTIFICATE-----
-----BEGIN RSA PRIVATE KEY-----
(Unencrypted Private Key)
-----END RSA PRIVATE KEY-----

cafile.pem like that:
-----BEGIN CERTIFICATE-----
(Thawte Primary Intermediate)
-----END CERTIFICATE-----
-----BEGIN CERTIFICATE-----
(Thawte Secondary Intermediate
-----END CERTIFICATE-----

Any suggestions?

Thanks in advance,
Adam
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From marketing at cluecon.com  Wed Aug 14 19:58:06 2013
From: marketing at cluecon.com (Michael Collins)
Date: Wed, 14 Aug 2013 08:58:06 -0700
Subject: [Freeswitch-users] ClueCon Weekly Conference Call Today
Message-ID: <CAKzWOxWNU=8k+zvntfb=912VWaT+ZPDVL2Amzq88jw1pHP6K6A@mail.gmail.com>

Hello all!

Today's conference call agenda is here:

https://wiki.freeswitch.org/wiki/FS_weekly_2013_08_14

We are going to recap ClueCon 2013 and talk about some of the interesting
discussions we had. If you attended ClueCon 2013 and have something you'd
like to share with the group then please call in. We would especially be
interested in having people give a brief recap of their favorite
presentations from this year. We'll also have an update on the uploading of
videos and presentation slides.

Talk to you soon!

-- 
Michael S Collins
ClueCon Team
http://www.cluecon.com
877-7-4ACLUE
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From jbarbosa at wavecom.pt  Wed Aug 14 20:20:53 2013
From: jbarbosa at wavecom.pt (Joao Barbosa)
Date: Wed, 14 Aug 2013 09:20:53 -0700 (PDT)
Subject: [Freeswitch-users] Offer SAVPF
Message-ID: <1376497253665-7593894.post@n2.nabble.com>

Hi,

how can i set Freeswitch 1.5 to offer SAVPF always?

Thank you



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Offer-SAVPF-tp7593894.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From jorgefren12 at gmail.com  Wed Aug 14 20:30:18 2013
From: jorgefren12 at gmail.com (=?ISO-8859-1?Q?Jorge_N=FA=F1ez?=)
Date: Wed, 14 Aug 2013 11:30:18 -0500
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
Message-ID: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>

Hi I am using mod_shout to send a conference call to icecast and hear it
from a tag audio in html5 in realtime, but it has a big delay like 12
seconds. How can I reduce the latency of the audio sent from freeswitch or
what can I do to improve this.

Regards

Jorge
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From krice at freeswitch.org  Wed Aug 14 20:35:24 2013
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 14 Aug 2013 11:35:24 -0500
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>
Message-ID: <CE311BFC.78EA0%krice@freeswitch.org>

You have to keep in mind that icecast itself has a fair bit of buffering
involved with it and theres not a lot you can do about that


On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com> wrote:

> Hi I am using mod_shout to send a conference call to icecast and hear it from
> a tag audio in html5 in realtime, but it has a big delay like 12 seconds. How
> can I reduce the latency of the audio sent from freeswitch or what can I do to
> improve this.
> 
> Regards
> 
> Jorge
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From sdame at 207me.com  Wed Aug 14 20:48:38 2013
From: sdame at 207me.com (Stephen Dame)
Date: Wed, 14 Aug 2013 12:48:38 -0400
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <CE311BFC.78EA0%krice@freeswitch.org>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>
	<CE311BFC.78EA0%krice@freeswitch.org>
Message-ID: <045801ce990e$205b4f90$6111eeb0$@207me.com>

Jorge, 

 

Play around with the burst size and queue size
  here is my xml config
around 3-4 second delay from 16k freeswitch conference. 

To small a buffer and the players disconnect
 Im also running icecast on
same server.

 

<icecast>

    <limits>

        <clients>100</clients>

        <sources>10</sources>

        <threadpool>5</threadpool>

        <queue-size>524288</queue-size>

        <client-timeout>30</client-timeout>

        <header-timeout>15</header-timeout>

        <source-timeout>10</source-timeout>

        <burst-on-connect>1</burst-on-connect>

        <!-- same as burst-on-connect, but this allows for being more

             specific on how much to burst. Most people won't need to

             change from the default 64k. Applies to all mountpoints  -->

        <burst-size>65535</burst-size>

    </limits>

 

Regards,

Stephen

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Wednesday, August 14, 2013 12:35 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Mod_shout using with icecast Big delay

 

You have to keep in mind that icecast itself has a fair bit of buffering
involved with it and theres not a lot you can do about that


On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com> wrote:

Hi I am using mod_shout to send a conference call to icecast and hear it
from a tag audio in html5 in realtime, but it has a big delay like 12
seconds. How can I reduce the latency of the audio sent from freeswitch or
what can I do to improve this.

Regards

Jorge


  _____  


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From denis.gasparin at edistar.com  Wed Aug 14 20:49:03 2013
From: denis.gasparin at edistar.com (Denis Gasparin)
Date: Wed, 14 Aug 2013 18:49:03 +0200 (CEST)
Subject: [Freeswitch-users] Httapi: getting digits in playback
In-Reply-To: <921523836.3920.1376497330997.JavaMail.root@mailserver.edistar.com>
Message-ID: <1224793464.3953.1376498943199.JavaMail.root@mailserver.edistar.com>

Hi. 


After playback I ask to the user to digit a number between 1 and 9999 with or without terminating the input with '#'. 


The first httapi xml I tried was the following: 



<document type="text/freeswitch-httapi"> 
<work> 
<playback file="ivr/please_enter_number_between_1_9999.wav" digit-timeout="5000" name="user_input" terminators="#"> 
<bind>~\d{1,4}#{0,1}</bind> 
</playback> 
</work> 
</document> 


Freeswitch calls my action as soon as the user input the first digit. 
So I modified the xml in order to use two bindings: 



<document type="text/freeswitch-httapi"> 
<work> 
<playback file="ivr/please_enter_number_between_1_9999.wav" digit-timeout="5000" name="user_input" terminators="#"> 
<bind>~\d{1,3}#{0,1}</bind> 
<bind>~\d{4}#{0,1}</bind> 
</playback> 
</work> 
</document> 


Using this xml I get the user input only if the user press "#" after inserting the number. 
If the user presses 123 and then waits for timeout, Freeswitch doesn't send any digits to my action (but sometimes it does). Why? 


>From the logs I see that the digits are always received from Freeswitch. 


Thank you in advance for your help. 
Denis Gasparin 
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From brian at freeswitch.org  Wed Aug 14 20:49:33 2013
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Aug 2013 11:49:33 -0500
Subject: [Freeswitch-users] FreeSWITCH Security Topics...
Message-ID: <9CE26E3B-8A38-488A-B7B3-DE0F24E3631F@freeswitch.org>

We need to start the security discussion in terms of FreeSWITCH and security related topics.  It was a very popular topic at ClueCon 2013 and I would like to continue the discussion.

freeswitch-sec at lists.freeswitch.org for those wanting to join in.

Thanks,
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

T: +1.918.420.9001  |  F: +1.918.420.9002  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9001
ISN: 410*543
Skype:briankwest
PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)









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From brian at freeswitch.org  Wed Aug 14 21:12:49 2013
From: brian at freeswitch.org (Brian West)
Date: Wed, 14 Aug 2013 12:12:49 -0500
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Security
	Topics...
In-Reply-To: <9CE26E3B-8A38-488A-B7B3-DE0F24E3631F@freeswitch.org>
References: <9CE26E3B-8A38-488A-B7B3-DE0F24E3631F@freeswitch.org>
Message-ID: <55664C25-FDF0-401D-B3D9-F236E07A14BD@freeswitch.org>

Please use http://lists.freeswitch.org/mailman/listinfo/freeswitch-sec 

To subscribe.

I've manually added the few so far that have sent in.  ;)

/b

On Aug 14, 2013, at 11:49 AM, Brian West <brian at freeswitch.org> wrote:

> We need to start the security discussion in terms of FreeSWITCH and security related topics.  It was a very popular topic at ClueCon 2013 and I would like to continue the discussion.
> 
> freeswitch-sec at lists.freeswitch.org for those wanting to join in.
> 
> Thanks,
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
> 
> T: +1.918.420.9001  |  F: +1.918.420.9002  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9001
> ISN: 410*543
> Skype:briankwest
> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
> 
> 
> 
> 
> 
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-dev mailing list
> FreeSWITCH-dev at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
> http://www.freeswitch.org

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From ryangard at gmail.com  Wed Aug 14 21:19:09 2013
From: ryangard at gmail.com (Ryan Gard)
Date: Wed, 14 Aug 2013 13:19:09 -0400
Subject: [Freeswitch-users] deny_refer_requests failure
Message-ID: <CAAQxoQH9R_nb5hv=nMqAknwL+ZsSaEiL3EcUF5jKnb=-YMva4w@mail.gmail.com>

Hey,

Running into issues with deny_refer_requests. I have verified that it is
setting the variable appropriately on both the a-leg and b-leg of the call,
but it still processes the refer request without a second thought upon
receiving it.

Is there anything specific on top of setting the variable to 'true' I
should be keeping tabs on? It doesn't seem to have too much in the way of
documentation, but the references I have seen say that it should work
without issue as long as one of the legs (both legs have it set) has it set
to true.

Thanks :)

-- 
Ryan Gard
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From jorgefren12 at gmail.com  Wed Aug 14 21:20:20 2013
From: jorgefren12 at gmail.com (=?ISO-8859-1?Q?Jorge_N=FA=F1ez?=)
Date: Wed, 14 Aug 2013 12:20:20 -0500
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <045801ce990e$205b4f90$6111eeb0$@207me.com>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>
	<CE311BFC.78EA0%krice@freeswitch.org>
	<045801ce990e$205b4f90$6111eeb0$@207me.com>
Message-ID: <CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>

Sorry but I am very new with freeswitch!!! what do you mean with 16K or how
can I know with how can I configure that?

Regards

Jorge


2013/8/14 Stephen Dame <sdame at 207me.com>

> Jorge, ****
>
> ** **
>
> Play around with the burst size and queue size?  here is my xml config
> around 3-4 second delay from 16k freeswitch conference. ****
>
> To small a buffer and the players disconnect? Im also running icecast on
> same server.****
>
> ** **
>
> <icecast>****
>
>     <limits>****
>
>         <clients>100</clients>****
>
>         <sources>10</sources>****
>
>         <threadpool>5</threadpool>****
>
>         <queue-size>524288</queue-size>****
>
>         <client-timeout>30</client-timeout>****
>
>         <header-timeout>15</header-timeout>****
>
>         <source-timeout>10</source-timeout>****
>
>         <burst-on-connect>1</burst-on-connect>****
>
>         <!-- same as burst-on-connect, but this allows for being more****
>
>              specific on how much to burst. Most people won't need to****
>
>              change from the default 64k. Applies to all mountpoints  -->*
> ***
>
>         <burst-size>65535</burst-size>****
>
>     </limits>****
>
> ** **
>
> Regards,****
>
> Stephen****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice
> *Sent:* Wednesday, August 14, 2013 12:35 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big delay**
> **
>
> ** **
>
> You have to keep in mind that icecast itself has a fair bit of buffering
> involved with it and theres not a lot you can do about that
>
>
> On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com> wrote:****
>
> Hi I am using mod_shout to send a conference call to icecast and hear it
> from a tag audio in html5 in realtime, but it has a big delay like 12
> seconds. How can I reduce the latency of the audio sent from freeswitch or
> what can I do to improve this.
>
> Regards
>
> Jorge****
> ------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sdame at 207me.com  Wed Aug 14 21:44:09 2013
From: sdame at 207me.com (Stephen Dame)
Date: Wed, 14 Aug 2013 13:44:09 -0400
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>	<CE311BFC.78EA0%krice@freeswitch.org>	<045801ce990e$205b4f90$6111eeb0$@207me.com>
	<CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>
Message-ID: <049101ce9915$e1cbb5f0$a56321d0$@207me.com>

Jorge, In freeswitch you can set the resolution of a conference with
different profiles in conference ,   8k,16k,32k etc..  I have noticed the
delay increases as you increase rate which makes sense as more data is
sampled.

 

<param name="rate" value="8000"/>

 

In /autoload_configs/conference.conf.xml

 

Regards,

Stephen

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jorge
N??ez
Sent: Wednesday, August 14, 2013 1:20 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Mod_shout using with icecast Big delay

 

Sorry but I am very new with freeswitch!!! what do you mean with 16K or how
can I know with how can I configure that?

 

Regards

 

Jorge

 

2013/8/14 Stephen Dame <sdame at 207me.com <mailto:sdame at 207me.com> >

Jorge, 

 

Play around with the burst size and queue size
  here is my xml config
around 3-4 second delay from 16k freeswitch conference. 

To small a buffer and the players disconnect
 Im also running icecast on
same server.

 

<icecast>

    <limits>

        <clients>100</clients>

        <sources>10</sources>

        <threadpool>5</threadpool>

        <queue-size>524288</queue-size>

        <client-timeout>30</client-timeout>

        <header-timeout>15</header-timeout>

        <source-timeout>10</source-timeout>

        <burst-on-connect>1</burst-on-connect>

        <!-- same as burst-on-connect, but this allows for being more

             specific on how much to burst. Most people won't need to

             change from the default 64k. Applies to all mountpoints  -->

        <burst-size>65535</burst-size>

    </limits>

 

Regards,

Stephen

 

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 Ken
Rice
Sent: Wednesday, August 14, 2013 12:35 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Mod_shout using with icecast Big delay

 

You have to keep in mind that icecast itself has a fair bit of buffering
involved with it and theres not a lot you can do about that


On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com
<http://jorgefren12 at gmail.com> > wrote:

Hi I am using mod_shout to send a conference call to icecast and hear it
from a tag audio in html5 in realtime, but it has a big delay like 12
seconds. How can I reduce the latency of the audio sent from freeswitch or
what can I do to improve this.

Regards

Jorge


  _____  


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <http://consulting at freeswitch.org> 
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net <http://irc.freenode.net>  #freeswitch


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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

 

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From msc at freeswitch.org  Wed Aug 14 22:02:05 2013
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 14 Aug 2013 11:02:05 -0700
Subject: [Freeswitch-users] Recording calls with Freeswitch - log says
 recorded, but no file found
In-Reply-To: <1376487277.65226.YahooMailNeo@web160503.mail.bf1.yahoo.com>
References: <1376487277.65226.YahooMailNeo@web160503.mail.bf1.yahoo.com>
Message-ID: <CAKzWOxWk9bUD4skaAO1cpYJC2kAiY6jK7+77DCQpBardNPsKcg@mail.gmail.com>

You'll need debug level output. That looks like it's info level output. The
freeswitch.log file should have debug level output. Also, if you're on the
freeswitch console (not the fs_cli) then info is the default level. Try
"console loglevel debug" and then retest and re-capture. Also, best place
to put fs log data is: pastebin.freeswitch.org. Use "FreeSWITCH Log" as the
syntax highlighting.

-MC


On Wed, Aug 14, 2013 at 6:34 AM, Ravi <sravi123 at yahoo.com> wrote:

> Hello Everyone !
>
> I am trying to record calls received in an extension (1003). The dialplan
> is given below
>
> extension name="003">
>     <condition field="destination_number" expression="^003$">
>         <action application="set" data="record_file_name={recordings_
> dir}/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.wav" inli$
>         <action application="record_session" data="{record_file_name}"/>
>         <action application="bridge" data="user/1003"/>
>     </condition>
> </extension>
>
> When I call this extension from an extension 1006, this is what I get in
> the log. But when I go to the folder /usr/local/freeswitch/recordings/
> there are no files at all. I am just wondering if this has got to do with
> any file permissions, or if I am missing something. Please help.
>
> Thanks.
> Ravi
>
>
> freeswitch at bfree-server> 2013-08-14 06:53:51.185709 [NOTICE]
> switch_channel.c:1030 New Channel sofia/internal/1006 at 10.0.0.16[2d3314f4-0480-11e3-9a2f-4d1cb7d8b5b4]
> 2013-08-14 06:53:51.245710 [INFO] mod_dialplan_xml.c:558 Processing Ravi
> <1006>->1003 in context default
> 2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *1
> execute_extension::dx XML features
> 2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *2
> record_session::/usr/local/freeswitch/recordings/1006.2013-08-14-06-53-51.wav
> 2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *3
> execute_extension::cf XML features
> 2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *4
> execute_extension::att_xfer XML features
> 2013-08-14 06:53:51.245710 [NOTICE] switch_channel.c:1030 New Channel
> sofia/internal/sip:1003 at 10.0.0.10:5060[2d3d1a26-0480-11e3-9a4d-4d1cb7d8b5b4]
> 2013-08-14 06:53:51.445707 [NOTICE] sofia.c:5884 Ring-Ready sofia/internal/
> sip:1003 at 10.0.0.10:5060!
> 2013-08-14 06:53:51.445707 [INFO] switch_ivr_originate.c:1190 Sending
> early media
> 2013-08-14 06:53:51.445707 [NOTICE] sofia_media.c:92 Pre-Answer
> sofia/internal/1006 at 10.0.0.16!
> 2013-08-14 06:53:52.705700 [NOTICE] sofia.c:6547 Channel [sofia/internal/
> sip:1003 at 10.0.0.10:5060] has been answered
> 2013-08-14 06:53:52.725708 [NOTICE] switch_ivr_originate.c:3437 Channel
> [sofia/internal/1006 at 10.0.0.16] has been answered
> 2013-08-14 06:54:05.005696 [NOTICE] sofia.c:715 Hangup sofia/internal/
> sip:1003 at 10.0.0.10:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2013-08-14 06:54:05.025711 [NOTICE] switch_ivr_bridge.c:1575 Hangup
> sofia/internal/1006 at 10.0.0.16 [CS_EXECUTE] [NORMAL_CLEARING]
> 2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1567 Session 139
> (sofia/internal/sip:1003 at 10.0.0.10:5060) Ended
> 2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1571 Close
> Channel sofia/internal/sip:1003 at 10.0.0.10:5060 [CS_DESTROY]
> 2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1567 Session 138
> (sofia/internal/1006 at 10.0.0.16) Ended
> 2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1571 Close
> Channel sofia/internal/1006 at 10.0.0.16 [CS_DESTROY]
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jorgefren12 at gmail.com  Thu Aug 15 00:18:18 2013
From: jorgefren12 at gmail.com (=?ISO-8859-1?Q?Jorge_N=FA=F1ez?=)
Date: Wed, 14 Aug 2013 15:18:18 -0500
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <049101ce9915$e1cbb5f0$a56321d0$@207me.com>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>
	<CE311BFC.78EA0%krice@freeswitch.org>
	<045801ce990e$205b4f90$6111eeb0$@207me.com>
	<CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>
	<049101ce9915$e1cbb5f0$a56321d0$@207me.com>
Message-ID: <CAOP4xcy6NZG0JKPMYhJp_wxaLqnAjsEkGMgTD5itvBPNdSCPzQ@mail.gmail.com>

Hi thanks, I set your configuration but nothing changed, I reduced the
burst size and it takes me just 11s and also I tried with 8k, 16k and 32k
and nothing change

    <limits>
        <clients>100</clients>
        <sources>10</sources>
        <threadpool>5</threadpool>
        <queue-size>524288</queue-size>
        <!-- <queue-size>524288</queue-size> -->
        <client-timeout>30</client-timeout>
        <header-timeout>15</header-timeout>
        <source-timeout>10</source-timeout>
        <!-- If enabled, this will provide a burst of data when a client
             first connects, thereby significantly reducing the startup
             time for listeners that do substantial buffering. However,
             it also significantly increases latency between the source
             client and listening client.  For low-latency setups, you
             might want to disable this. -->
        <burst-on-connect>0</burst-on-connect>
        <!-- same as burst-on-connect, but this allows for being more
             specific on how much to burst. Most people won't need to
             change from the default 64k. Applies to all mountpoints

 -->
        <burst-size>4096</burst-size>
    </limits>

Regards

Jorge



2013/8/14 Stephen Dame <sdame at 207me.com>

> Jorge, ****
>
> ** **
>
> Play around with the burst size and queue size?  here is my xml config
> around 3-4 second delay from 16k freeswitch conference. ****
>
> To small a buffer and the players disconnect? Im also running icecast on
> same server.****
>
> ** **
>
> <icecast>****
>
>     <limits>****
>
>         <clients>100</clients>****
>
>         <sources>10</sources>****
>
>         <threadpool>5</threadpool>****
>
>         <queue-size>524288</queue-size>****
>
>         <client-timeout>30</client-timeout>****
>
>         <header-timeout>15</header-timeout>****
>
>         <source-timeout>10</source-timeout>****
>
>         <burst-on-connect>1</burst-on-connect>****
>
>         <!-- same as burst-on-connect, but this allows for being more****
>
>              specific on how much to burst. Most people won't need to****
>
>              change from the default 64k. Applies to all mountpoints  -->*
> ***
>
>         <burst-size>65535</burst-size>****
>
>     </limits>****
>
> ** **
>
> Regards,****
>
> Stephen****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice
> *Sent:* Wednesday, August 14, 2013 12:35 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big delay**
> **
>
> ** **
>
> You have to keep in mind that icecast itself has a fair bit of buffering
> involved with it and theres not a lot you can do about that
>
>
> On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com> wrote:****
>
> Hi I am using mod_shout to send a conference call to icecast and hear it
> from a tag audio in html5 in realtime, but it has a big delay like 12
> seconds. How can I reduce the latency of the audio sent from freeswitch or
> what can I do to improve this.
>
> Regards
>
> Jorge****
> ------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 babak.freeswitch at gmail.com  Thu Aug 15 00:28:38 2013
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Thu, 15 Aug 2013 00:58:38 +0430
Subject: [Freeswitch-users] loopback problem with lua and rxfax
Message-ID: <CAEVfcaa8VOw7Ajt9nBymzSLNF9-np0TaUH-j1uhH7sdfS6tAoA@mail.gmail.com>

Hi
I'm using these 2 extensions for receiving faxes. to receive a fax first
call hits "faxin" extension and then is bridged to "dofaxin". I'm using
loopback to be able to handle attended transfers from ivrs that include
fax. the problem is if I send a call directly to 704 everything works fine.
but if I send it to 705 rxfax (which is the main case)it is not working as
expected and after a long waiting rxfax fails with " The call dropped
prematurely".

<extension name="dofaxin">
     <condition field="destination_number" expression="^704$">
       <action application="answer"/>
       <action application="playback" data="silence_stream://2000"/>
       <action application="set" data="fax_enable_t38_request=true"/>
       <action application="set" data="fax_enable_t38=true"/>
       <action application="lua" data="lua/applications/fax.lua"/>
      <!--action application="rxfax" data="/tmp/shit.tiff"/-->
     </condition>
   </extension>
   <extension name="faxin">
    <condition field="destination_number" expression="^705$">
       <action application="bridge" data="loopback/704/public/XML"/>
    </condition>
  </extension>

and the simplified version of file "lua/applications/fax.lua" is:
...
session:execute('rxfax','/tmp/test.tiff')
....
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From cordes.matthew at gmail.com  Thu Aug 15 00:34:32 2013
From: cordes.matthew at gmail.com (Matthew Cordes)
Date: Wed, 14 Aug 2013 16:34:32 -0400
Subject: [Freeswitch-users] Recording conference only when a particular user
	is connected
Message-ID: <CADcWJKGb2q+z1wQNTdjdY5YoYZuR7JoFimTggWrK3rsDqyq_Pg@mail.gmail.com>

Hello,

I'd like to record a conference call, but only when a particular user is
part of the call and even when the conference just contains that user. When
that user leaves the conference, even if there are other users in the
conference I want to stop recording. Any recommendations on the best
approach?

-Matt
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From pasha at prosperity4ever.com  Thu Aug 15 00:47:49 2013
From: pasha at prosperity4ever.com (Paul)
Date: Wed, 14 Aug 2013 20:40:49 -0007
Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH
	Security	Topics...
In-Reply-To: <55664C25-FDF0-401D-B3D9-F236E07A14BD@freeswitch.org>
References: <9CE26E3B-8A38-488A-B7B3-DE0F24E3631F@freeswitch.org>
	<55664C25-FDF0-401D-B3D9-F236E07A14BD@freeswitch.org>
Message-ID: <20130814204757.EBCA957E001@mail.mydcs.ca>

I will second that! :)

On Wed, 14 Aug, 2013 at 10:12 AM, Brian West <brian at freeswitch.org> 
wrote:
> Please use 
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-sec 
> 
> To subscribe.
> 
> I've manually added the few so far that have sent in.  ;)
> 
> /b
> 
> On Aug 14, 2013, at 11:49 AM, Brian West <brian at freeswitch.org> wrote:
> 
>>  We need to start the security discussion in terms of FreeSWITCH and 
>> security related topics.  It was a very popular topic at ClueCon 
>> 2013 and I would like to continue the discussion.
>>  
>>  freeswitch-sec at lists.freeswitch.org for those wanting to join in.
>>  
>>  Thanks,
>>  --
>>  Brian West
>>  brian at freeswitch.org
>>  FreeSWITCH Solutions, LLC
>>  PO BOX PO BOX 2531
>>  Brookfield, WI 53008-2531
>>  Twitter: @FreeSWITCH_Wire
>>  http://www.freeswitchbook.com
>>  http://www.freeswitchcookbook.com
>>  
>>  T: +1.918.420.9001  |  F: +1.918.420.9002  |  M: +1.918.424.WEST
>>  iNUM: +883 5100 1420 9001
>>  ISN: 410*543
>>  Skype:briankwest
>>  PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
>>  
>>  
>>  
>>  
>>  
>>  
>>  
>>  
>>  
>>  
>> _________________________________________________________________________
>>  Professional FreeSWITCH Consulting Services:
>>  consulting at freeswitch.org
>>  http://www.freeswitchsolutions.com
>>  
>>  
>>  
>>  
>>  Official FreeSWITCH Sites
>>  http://www.freeswitch.org
>>  http://wiki.freeswitch.org
>>  http://www.cluecon.com
>>  
>>  FreeSWITCH-dev mailing list
>>  FreeSWITCH-dev at lists.freeswitch.org
>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev
>>  
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev
>>  http://www.freeswitch.org
>> 
> 
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From khorsmann at gmail.com  Thu Aug 15 01:11:17 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Wed, 14 Aug 2013 23:11:17 +0200
Subject: [Freeswitch-users] Recording conference only when a particular
 user is connected
In-Reply-To: <CADcWJKGb2q+z1wQNTdjdY5YoYZuR7JoFimTggWrK3rsDqyq_Pg@mail.gmail.com>
References: <CADcWJKGb2q+z1wQNTdjdY5YoYZuR7JoFimTggWrK3rsDqyq_Pg@mail.gmail.com>
Message-ID: <CAFArqsaSqsHxda8XF2x_eBM5kZm7rJsd9XYUC2TDk=M=4rjddg@mail.gmail.com>

Hello Matt,

a shot in the dark may an lua script that start/stops recording if your
user join/parts the conference.

There are some lua conference scripts that could be a good starting point:

https://code.google.com/p/fusionpbx/source/browse/trunk/fusionpbx/includes/install/scripts/recordings.lua
https://github.com/amooma/GS5/blob/master/misc/freeswitch/scripts/common/conference.lua

(the last one is not opensource IMHO)

And then you add an function that checks your wanted phone-number as
trigger.


2013/8/14 Matthew Cordes <cordes.matthew at gmail.com>

> Hello,
>
> I'd like to record a conference call, but only when a particular user is
> part of the call and even when the conference just contains that user. When
> that user leaves the conference, even if there are other users in the
> conference I want to stop recording. Any recommendations on the best
> approach?
>
> -Matt
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From sdame at 207me.com  Thu Aug 15 02:12:01 2013
From: sdame at 207me.com (Stephen Dame)
Date: Wed, 14 Aug 2013 18:12:01 -0400
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <CAOP4xcy6NZG0JKPMYhJp_wxaLqnAjsEkGMgTD5itvBPNdSCPzQ@mail.gmail.com>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>	<CE311BFC.78EA0%krice@freeswitch.org>	<045801ce990e$205b4f90$6111eeb0$@207me.com>	<CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>	<049101ce9915$e1cbb5f0$a56321d0$@207me.com>
	<CAOP4xcy6NZG0JKPMYhJp_wxaLqnAjsEkGMgTD5itvBPNdSCPzQ@mail.gmail.com>
Message-ID: <050301ce993b$4dc37b60$e94a7220$@207me.com>

Did you restart the freeswitch server after changes to the config files,
I?m not sure 100% if reloadxml resets the conference settings. 

 

Also I use some flash based players, I think there where client settiings
for buffering.   Can you set buffer in HTML tag?

 

Regards,

Stephen

 

207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jorge
N??ez
Sent: Wednesday, August 14, 2013 4:18 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Mod_shout using with icecast Big delay

 

Hi thanks, I set your configuration but nothing changed, I reduced the burst
size and it takes me just 11s and also I tried with 8k, 16k and 32k and
nothing change

 

    <limits>

        <clients>100</clients>

        <sources>10</sources>

        <threadpool>5</threadpool>

        <queue-size>524288</queue-size>

        <!-- <queue-size>524288</queue-size> -->

        <client-timeout>30</client-timeout>

        <header-timeout>15</header-timeout>

        <source-timeout>10</source-timeout>

        <!-- If enabled, this will provide a burst of data when a client 

             first connects, thereby significantly reducing the startup 

             time for listeners that do substantial buffering. However,

             it also significantly increases latency between the source

             client and listening client.  For low-latency setups, you

             might want to disable this. -->

        <burst-on-connect>0</burst-on-connect>

        <!-- same as burst-on-connect, but this allows for being more

             specific on how much to burst. Most people won't need to

             change from the default 64k. Applies to all mountpoints 

 

 -->

        <burst-size>4096</burst-size>

    </limits>

 

Regards

 

Jorge

 

 

2013/8/14 Stephen Dame <sdame at 207me.com <mailto:sdame at 207me.com> >

Jorge, 

 

Play around with the burst size and queue size
  here is my xml config
around 3-4 second delay from 16k freeswitch conference. 

To small a buffer and the players disconnect
 Im also running icecast on
same server.

 

<icecast>

    <limits>

        <clients>100</clients>

        <sources>10</sources>

        <threadpool>5</threadpool>

        <queue-size>524288</queue-size>

        <client-timeout>30</client-timeout>

        <header-timeout>15</header-timeout>

        <source-timeout>10</source-timeout>

        <burst-on-connect>1</burst-on-connect>

        <!-- same as burst-on-connect, but this allows for being more

             specific on how much to burst. Most people won't need to

             change from the default 64k. Applies to all mountpoints  -->

        <burst-size>65535</burst-size>

    </limits>

 

Regards,

Stephen

 

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 Ken
Rice
Sent: Wednesday, August 14, 2013 12:35 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Mod_shout using with icecast Big delay

 

You have to keep in mind that icecast itself has a fair bit of buffering
involved with it and theres not a lot you can do about that


On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com
<http://jorgefren12 at gmail.com> > wrote:

Hi I am using mod_shout to send a conference call to icecast and hear it
from a tag audio in html5 in realtime, but it has a big delay like 12
seconds. How can I reduce the latency of the audio sent from freeswitch or
what can I do to improve this.

Regards

Jorge


  _____  


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <http://consulting at freeswitch.org> 
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net <http://irc.freenode.net>  #freeswitch


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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

 

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From ssinyagin at yahoo.com  Thu Aug 15 02:26:12 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Wed, 14 Aug 2013 15:26:12 -0700 (PDT)
Subject: [Freeswitch-users] Recording conference only when a particular
	user	is connected
In-Reply-To: <CADcWJKGb2q+z1wQNTdjdY5YoYZuR7JoFimTggWrK3rsDqyq_Pg@mail.gmail.com>
References: <CADcWJKGb2q+z1wQNTdjdY5YoYZuR7JoFimTggWrK3rsDqyq_Pg@mail.gmail.com>
Message-ID: <1376519172.51469.YahooMailNeo@web126202.mail.ne1.yahoo.com>

some possible scenarios, need further digging:

a) for that user, execute the "record" dialplan application, then transfer to the conference extension. It can be achieved by a1) setting a variable in the user directory entry, and then matching the variable in the dialplan; a2) defining a separate context for the user.

b) an external script would monitor the ESL events and execute uuid_record when needed







>________________________________
> From: Matthew Cordes <cordes.matthew at gmail.com>
>To: freeswitch-users at lists.freeswitch.org 
>Sent: Wednesday, August 14, 2013 10:34 PM
>Subject: [Freeswitch-users] Recording conference only when a particular user	is connected
> 
>
>
>Hello, 
>
>I'd like to record a conference call, but only when a particular user is part of the call and even when the conference just contains that user. When that user leaves the conference, even if there are other users in the conference I want to stop recording. Any recommendations on the best approach? 
>
>-Matt
>
>
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 eidevm5 at gmail.com  Thu Aug 15 03:06:01 2013
From: eidevm5 at gmail.com (Peter)
Date: Thu, 15 Aug 2013 09:06:01 +1000
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
	<CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
Message-ID: <CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>

My mistake, I meant loglevel rather than logging.

So when you say "no progress" are you seeing nothing at all on the FS
console?

If so, then that indicates a network problem as FS isn't even seeing the
registration requests.

Can you ping the windows box from the FS server?


On Wed, Aug 14, 2013 at 7:04 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> I went for "sofia loglevel 5" instead of "sofia logging 5" but still no
> progress in connection of softphone...:-(
> Thanks
> Ashish Mishra
> On Aug 14, 2013 2:14 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>
>> Also the command that you had mentioned "sofia logging all 5" gives me an
>> error message :
>> Unknown command [logging]
>> On Aug 14, 2013 7:24 AM, "Peter" <eidevm5 at gmail.com> wrote:
>>
>>> Connect to the FS console with fs_cli and do
>>>
>>> sofia logging all 5
>>>
>>> then try to register your softphone.    If you don't even see a
>>> registration attempt on the FS console, it means you have a network problem.
>>>
>>> If the network isn't a problem, the debug output should give you a clue
>>> as to what the problem is.
>>>
>>>
>>>
>>> On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> Thank you Peter...you mean to say that i should first turn on the
>>>> fs_cli and then retry to connect the softphone...??? I would also like to
>>>> remind you that fs_cli and freeswitch are running on the same machine and i
>>>> am trying to connect thru a network cable my windows 8 pc that has the
>>>> softphone on it...
>>>> Regards
>>>> Ashish Mishra
>>>> On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>
>>>>> Have you turned on debugging from the FS cli and seen if any
>>>>> registration requests come in?
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>>
>>>>>> I installed freeswitch on my ubuntu 12.04 machine...but when i am
>>>>>> trying to connect the softphone installed on my windows 8 pc with ubuntu
>>>>>> machine the softphone gives me an error that account not enabled...i have
>>>>>> used a network cable to connect the two machines...also the firewall in
>>>>>> both the machines is disabled...
>>>>>> Kindly help..
>>>>>>
>>>>>> Ashish Mishra
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Thu Aug 15 03:19:06 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 14 Aug 2013 18:19:06 -0500
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <050301ce993b$4dc37b60$e94a7220$@207me.com>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>
	<CE311BFC.78EA0%krice@freeswitch.org>
	<045801ce990e$205b4f90$6111eeb0$@207me.com>
	<CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>
	<049101ce9915$e1cbb5f0$a56321d0$@207me.com>
	<CAOP4xcy6NZG0JKPMYhJp_wxaLqnAjsEkGMgTD5itvBPNdSCPzQ@mail.gmail.com>
	<050301ce993b$4dc37b60$e94a7220$@207me.com>
Message-ID: <CAKbxfG80vr3b+BULjn5ZhKW6ZjxY6Khy4Ymhk8RWGVFLpuSKoA@mail.gmail.com>

Icecast is not designed for low-latency and realtime audio.  Its designed
for higher quality reliable audio.  The servers and the client libs both
use latency and buffering to guarantee this.  There are some techniques to
reduce the buffering on at least the FS side but then you will start
getting skips and resets if you miss any packets.

This comes up all the time when people first try doing this.  Either you
need to just accept the delay since most people will not even know its
there or use some other method.




On Wed, Aug 14, 2013 at 5:12 PM, Stephen Dame <sdame at 207me.com> wrote:

> Did you restart the freeswitch server after changes to the config files,
> I?m not sure 100% if reloadxml resets the conference settings. ****
>
> ** **
>
> Also I use some flash based players, I think there where client settiings
> for buffering.   Can you set buffer in HTML tag?****
>
> ** **
>
> Regards,****
>
> Stephen****
>
> ** **
>
> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jorge N??ez
> *Sent:* Wednesday, August 14, 2013 4:18 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big delay**
> **
>
> ** **
>
> Hi thanks, I set your configuration but nothing changed, I reduced the
> burst size and it takes me just 11s and also I tried with 8k, 16k and 32k
> and nothing change****
>
> ** **
>
>     <limits>****
>
>         <clients>100</clients>****
>
>         <sources>10</sources>****
>
>         <threadpool>5</threadpool>****
>
>         <queue-size>524288</queue-size>****
>
>         <!-- <queue-size>524288</queue-size> -->****
>
>         <client-timeout>30</client-timeout>****
>
>         <header-timeout>15</header-timeout>****
>
>         <source-timeout>10</source-timeout>****
>
>         <!-- If enabled, this will provide a burst of data when a client *
> ***
>
>              first connects, thereby significantly reducing the startup **
> **
>
>              time for listeners that do substantial buffering. However,***
> *
>
>              it also significantly increases latency between the source***
> *
>
>              client and listening client.  For low-latency setups, you****
>
>              might want to disable this. -->****
>
>         <burst-on-connect>0</burst-on-connect>****
>
>         <!-- same as burst-on-connect, but this allows for being more****
>
>              specific on how much to burst. Most people won't need to****
>
>              change from the default 64k. Applies to all mountpoints ****
>
> ** **
>
>  -->****
>
>         <burst-size>4096</burst-size>****
>
>     </limits>****
>
> ** **
>
> Regards****
>
> ** **
>
> Jorge****
>
> ** **
>
> ** **
>
> 2013/8/14 Stephen Dame <sdame at 207me.com>****
>
> Jorge, ****
>
>  ****
>
> Play around with the burst size and queue size?  here is my xml config
> around 3-4 second delay from 16k freeswitch conference. ****
>
> To small a buffer and the players disconnect? Im also running icecast on
> same server.****
>
>  ****
>
> <icecast>****
>
>     <limits>****
>
>         <clients>100</clients>****
>
>         <sources>10</sources>****
>
>         <threadpool>5</threadpool>****
>
>         <queue-size>524288</queue-size>****
>
>         <client-timeout>30</client-timeout>****
>
>         <header-timeout>15</header-timeout>****
>
>         <source-timeout>10</source-timeout>****
>
>         <burst-on-connect>1</burst-on-connect>****
>
>         <!-- same as burst-on-connect, but this allows for being more****
>
>              specific on how much to burst. Most people won't need to****
>
>              change from the default 64k. Applies to all mountpoints  -->*
> ***
>
>         <burst-size>65535</burst-size>****
>
>     </limits>****
>
>  ****
>
> Regards,****
>
> Stephen****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice
> *Sent:* Wednesday, August 14, 2013 12:35 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big delay**
> **
>
>  ****
>
> You have to keep in mind that icecast itself has a fair bit of buffering
> involved with it and theres not a lot you can do about that
>
>
> On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com> wrote:****
>
> Hi I am using mod_shout to send a conference call to icecast and hear it
> from a tag audio in html5 in realtime, but it has a big delay like 12
> seconds. How can I reduce the latency of the audio sent from freeswitch or
> what can I do to improve this.
>
> Regards
>
> Jorge****
> ------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From mitch.capper at gmail.com  Thu Aug 15 03:36:50 2013
From: mitch.capper at gmail.com (Mitch Capper)
Date: Wed, 14 Aug 2013 16:36:50 -0700
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
	<CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
	<CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
Message-ID: <CALn233ovuLR8rgD==bqiOi+Fa=yDwS8azsrD3cWkBz9sz7Mw7w@mail.gmail.com>

You may want to try FSClient its a freeswitch based windows soft phone and
you can connect fs_cli (or fs_logger.pl) to it and see what happens when it
tries to connect to your server.

~mitch


On Wed, Aug 14, 2013 at 4:06 PM, Peter <eidevm5 at gmail.com> wrote:

> My mistake, I meant loglevel rather than logging.
>
> So when you say "no progress" are you seeing nothing at all on the FS
> console?
>
> If so, then that indicates a network problem as FS isn't even seeing the
> registration requests.
>
> Can you ping the windows box from the FS server?
>
>
> On Wed, Aug 14, 2013 at 7:04 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> I went for "sofia loglevel 5" instead of "sofia logging 5" but still no
>> progress in connection of softphone...:-(
>> Thanks
>> Ashish Mishra
>> On Aug 14, 2013 2:14 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>>
>>> Also the command that you had mentioned "sofia logging all 5" gives me
>>> an error message :
>>> Unknown command [logging]
>>> On Aug 14, 2013 7:24 AM, "Peter" <eidevm5 at gmail.com> wrote:
>>>
>>>> Connect to the FS console with fs_cli and do
>>>>
>>>> sofia logging all 5
>>>>
>>>> then try to register your softphone.    If you don't even see a
>>>> registration attempt on the FS console, it means you have a network problem.
>>>>
>>>> If the network isn't a problem, the debug output should give you a clue
>>>> as to what the problem is.
>>>>
>>>>
>>>>
>>>> On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>
>>>>> Thank you Peter...you mean to say that i should first turn on the
>>>>> fs_cli and then retry to connect the softphone...??? I would also like to
>>>>> remind you that fs_cli and freeswitch are running on the same machine and i
>>>>> am trying to connect thru a network cable my windows 8 pc that has the
>>>>> softphone on it...
>>>>> Regards
>>>>> Ashish Mishra
>>>>> On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>>
>>>>>> Have you turned on debugging from the FS cli and seen if any
>>>>>> registration requests come in?
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <itsme.kunnu at gmail.com
>>>>>> > wrote:
>>>>>>
>>>>>>> I installed freeswitch on my ubuntu 12.04 machine...but when i am
>>>>>>> trying to connect the softphone installed on my windows 8 pc with ubuntu
>>>>>>> machine the softphone gives me an error that account not enabled...i have
>>>>>>> used a network cable to connect the two machines...also the firewall in
>>>>>>> both the machines is disabled...
>>>>>>> Kindly help..
>>>>>>>
>>>>>>> Ashish Mishra
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sirimmfs at gmail.com  Thu Aug 15 04:26:36 2013
From: sirimmfs at gmail.com (Siri MM)
Date: Thu, 15 Aug 2013 10:26:36 +1000
Subject: [Freeswitch-users] Auto-detecting Local IP when no gateway set
Message-ID: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>

Hi All,

I am using freeswitch without local_ip_v4 explicitly set, i.e, I am
dependant on freeswitch auto-detecting the local IP address.

>From the code, what I understand is, in order to determine the local IP
address,  freeswitch tries to ?connect()? using a DGRAM socket with
BROADCAST option set.

The problem that I am having is, my customer is working in a Local LAN, and
is not very keen on setting the interface gateway. Without gateway set,
connect() returns 'Network unreachable' error.

Is there any way I could workaround this problem? (apart from setting
local_ip_v4) Also, any other issues forseen if gateway isn't set ont he
interface?
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From jleung at v10networks.ca  Thu Aug 15 06:00:06 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Wed, 14 Aug 2013 19:00:06 -0700
Subject: [Freeswitch-users] Auto-detecting Local IP when no gateway set
In-Reply-To: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
Message-ID: <CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>

Why can't you just specify a static IP address instead? Specifying a static
IP address sure does make life a heck a lot easier that way.


On Wed, Aug 14, 2013 at 5:26 PM, Siri MM <sirimmfs at gmail.com> wrote:

> Hi All,
>
> I am using freeswitch without local_ip_v4 explicitly set, i.e, I am
> dependant on freeswitch auto-detecting the local IP address.
>
> From the code, what I understand is, in order to determine the local IP
> address,  freeswitch tries to ?connect()? using a DGRAM socket with
> BROADCAST option set.
>
> The problem that I am having is, my customer is working in a Local LAN,
> and is not very keen on setting the interface gateway. Without gateway set,
> connect() returns 'Network unreachable' error.
>
> Is there any way I could workaround this problem? (apart from setting
> local_ip_v4) Also, any other issues forseen if gateway isn't set ont he
> interface?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sirimmfs at gmail.com  Thu Aug 15 06:12:00 2013
From: sirimmfs at gmail.com (Siri MM)
Date: Thu, 15 Aug 2013 12:12:00 +1000
Subject: [Freeswitch-users] Auto-detecting Local IP when no gateway set
In-Reply-To: <CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
Message-ID: <CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>

Hi Jeff,

Couple of reasons why I dont want to specify IP address explicitly:
1. Customer is using multiple devices with freeswitch on each, and my
strategy is to provide a generically configured freeswitch, which would
work on all
2. Customer might at any point in time change the static IP address of
devices, or decide to use DHCP!


On Thu, Aug 15, 2013 at 12:00 PM, Jeff Leung <jleung at v10networks.ca> wrote:

> Why can't you just specify a static IP address instead? Specifying a
> static IP address sure does make life a heck a lot easier that way.
>
>
> On Wed, Aug 14, 2013 at 5:26 PM, Siri MM <sirimmfs at gmail.com> wrote:
>
>> Hi All,
>>
>> I am using freeswitch without local_ip_v4 explicitly set, i.e, I am
>> dependant on freeswitch auto-detecting the local IP address.
>>
>> From the code, what I understand is, in order to determine the local IP
>> address,  freeswitch tries to ?connect()? using a DGRAM socket with
>> BROADCAST option set.
>>
>> The problem that I am having is, my customer is working in a Local LAN,
>> and is not very keen on setting the interface gateway. Without gateway set,
>> connect() returns 'Network unreachable' error.
>>
>> Is there any way I could workaround this problem? (apart from setting
>> local_ip_v4) Also, any other issues forseen if gateway isn't set ont he
>> interface?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 cordes.matthew at gmail.com  Thu Aug 15 07:06:25 2013
From: cordes.matthew at gmail.com (Matthew Cordes)
Date: Wed, 14 Aug 2013 23:06:25 -0400
Subject: [Freeswitch-users] Recording conference only when a particular
	user is connected
Message-ID: <CADcWJKFsMnW47mOd_2F=s-nnhOoMkH2wnH07mBSfeHK+aK9nyA@mail.gmail.com>

Hi Stanislav,

For a) (below) how do I stop the recording? Will that happen automatically
when the user hangs up even though there are others still in the
conference? If so, this sounds like it could work.

Thanks,
-Matt


On Wed, Aug 14, 2013 at 7:19 PM, <
freeswitch-users-request at lists.freeswitch.org> wrote:

> ---------- Forwarded message ----------
> From: Stanislav Sinyagin <ssinyagin at yahoo.com>
> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> Cc:
> Date: Wed, 14 Aug 2013 15:26:12 -0700 (PDT)
> Subject: Re: [Freeswitch-users] Recording conference only when a
> particular user is connected
> some possible scenarios, need further digging:
>
> a) for that user, execute the "record" dialplan application, then transfer
> to the conference extension. It can be achieved by a1) setting a variable
> in the user directory entry, and then matching the variable in the
> dialplan; a2) defining a separate context for the user.
>
> b) an external script would monitor the ESL events and execute uuid_record
> when needed
>
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From karl at xtronics.com  Thu Aug 15 07:21:28 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Wed, 14 Aug 2013 22:21:28 -0500
Subject: [Freeswitch-users] freetdm rxgain txgain
In-Reply-To: <CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
	<CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
	<CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
Message-ID: <520C4938.4010202@xtronics.com>

Looks like rxgain and txgain are not working - I was testing using the cool mW extension (thanks - 
whoever put that in)  and could not change the level - did some digging and dahdi_diag shows that 
the gains don't change.

; Aug 14 18:16:51 malaysia kernel: [350647.964612] dahdi: rxgain: ffffffffa02e7c70, txgain: 
ffffffffa02e7c70, gainalloc: 0

; Aug 14 18:20:33 malaysia kernel: [350869.980742] dahdi: rxgain: ffffffffa02e7c70, txgain: 
ffffffffa02e7c70, gainalloc: 0

Even tried changes over a reboot.


This card has absorbed a lot of time:
(Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)) [OT]:(and now I find that it fails 
to put out a real ring signal.  Apparently, whoever designed it did not know the difference between 
RMS and peak voltage... fails to ring 3 out of 4 modems I tested. Gump, hiss,groan).

I can go in and modify the modem so it can detect the weak ring signal, but out of the box the audio 
levels on the card are way off ( they were also off in Asterisk ) and if I want to have this working 
well , I need to tweak those levels.


BTW - I can use the 1 dBm test tone stream to monitor the transmitted level - is there a way to 
monitor a level coming back the other way?





--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

The lawyer was about two bubbles off plumb.
--------------------------------------------------------------------------------


From eidevm5 at gmail.com  Thu Aug 15 09:52:59 2013
From: eidevm5 at gmail.com (Peter)
Date: Thu, 15 Aug 2013 15:52:59 +1000
Subject: [Freeswitch-users] Establishing SRTP from SBC to endpoint
In-Reply-To: <CABkS=xrLBX2A1ZLO9gx7mW2yQ2e9LTjyDaNOqJXH2DFADS4KYg@mail.gmail.com>
References: <CABkS=xqD0mukebja+w3aurrhcUJ4Gp=DcOpkh+BDnCpOKr32CA@mail.gmail.com>
	<CAAYhoXLzsB-o=5VNBaRsR9xdS8bm=j5zKxe8ideg5-jmKU=fKw@mail.gmail.com>
	<CABkS=xrLBX2A1ZLO9gx7mW2yQ2e9LTjyDaNOqJXH2DFADS4KYg@mail.gmail.com>
Message-ID: <CABkS=xp5XcskU6uJoM-7iHy3taPm3M=QiD29UO9EfToDTR-Fug@mail.gmail.com>

Finally got it going.  I don't know how many combinations I tried.

All I needed was the sip_secure_media (or rtp_secure_media, which is the
new name) set to true in the dialplan on the SBC.


On Wed, Aug 14, 2013 at 11:42 AM, Peter <eidevm5 at gmail.com> wrote:

> Hi Carlos.
>
> Didn't realise rtp_secure_media existed.  After searching I saw:
>
>
> https://wiki.freeswitch.org/wiki/Release_Notes#rtp_secure_media_.28was_sip_secure_media.29
>
> which says it was introduced in 1.2.9
>
> However, it's a little ambiguous as to whether sip_secure_media was
> deprecated.
>
> Anyway, I tried using rtp_secure_media instead, but I still can't get SRTP
> working.
>
>
> I did some testing with some other SIP clients.   In particular,
> csipsimple.  On the client, if I set SRTP to be optional, the media stream
> uses RTP.   However, if I set SRTP to be mandatory, when I try to call it,
> Freeswitch receives:
>
>    SIP/2.0 488 Not Acceptable Here
>
> Which seems to indicate that something is not is not right with the SRTP
> setup.
>
> There's a full debug from the FS1 (the freeswitch server where the
> csipsimple client is registered to) at:
>
> http://pastebin.freeswitch.org/21295
>
> Note in the debug I have sdp_secure_savp_only set to true.   I've tried
> disabling this setting, but get the same result.
>
> Thanks
>
> Peter
>
>
>
>
>
> On Tue, Aug 13, 2013 at 11:06 PM, Carlos Flor <jackal at cybershroud.net>wrote:
>
>> Try using rtp_secure_media=true instead of sip_secure_media.  If you are
>> trying to set it on the b-leg, you probably want to use export instead of
>> set, or use nolocal:rtp_secure_media.
>>
>> Hope that helps.
>>
>>
>> On Mon, Aug 12, 2013 at 10:26 PM, Peter <eidevm5 at gmail.com> wrote:
>>
>>> In my environment, I have the following (simplified) setup:
>>>
>>> FS1  ----  FS SBC ---  FS2
>>>
>>> Phones registered to FS1 (100x) use TLS/SRTP and phones registered to
>>> FS2 (200x) use SIP/RTP
>>>
>>> FS1 has inbound-bypass-media set to true to allow SRTP peer to peer and
>>> direct to the SBC.
>>>
>>> If I make an inbound call (eg: 1000 to 2000), SRTP is correctly
>>> established between the phone and SBC with RTP on the other side of the SBC
>>> to the internal phone.
>>>
>>> However, when I try it the other way, I can't get SRTP established from
>>> the SBC to the external phone.
>>>
>>> I've been using https://wiki.freeswitch.org/wiki/Secure_RTP as a guide.
>>>
>>> I've even tried explicitly setting sip_secure_media to true on the SBC
>>> and FS1.
>>>
>>> The dialplan on the SBC has:
>>>
>>>   <extension name="outgoing">
>>>         <condition field="destination_number"
>>> expression="^(10[0-9][0-9])$">
>>>             <action application="set" data="sip_secure_media=true"/>
>>>             <action application="bridge" data="sofia/external/${
>>> destination_number}@10.1.1.204"/>
>>>         </condition>
>>>   </extension>
>>>
>>>
>>> And on FS1, the dialplan has:
>>>
>>>    <extension name="Local-Numbers">
>>>       <condition field="destination_number" expression="^(10[01][0-9])$">
>>>         <action application="export" data="dialed_extension=$1"/>
>>>         <action application="set" data="sip_secure_media=true"/>
>>>         <action application="bridge" data="user/${dialed_extension}@
>>> ${domain_name}"/>
>>>       </condition>
>>>     </extension>
>>>
>>>
>>> Note that I've been testing this against two phones with SRTP enabled,
>>> but only one that is using TLS.  I get the same result calling each phone.
>>>
>>> On a related point, what it the step required for a TLS connection from
>>> the SBC to the phone?   I'm assume the phone just needs the CA cert from
>>> the SBC.  Correct?
>>>
>>> Any information as to where I'm going wrong will be gratefully accepted.
>>>
>>> Thanks
>>>
>>> Peter
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 eidevm5 at gmail.com  Thu Aug 15 09:57:25 2013
From: eidevm5 at gmail.com (Peter)
Date: Thu, 15 Aug 2013 15:57:25 +1000
Subject: [Freeswitch-users] Establishing SRTP from SBC to endpoint
In-Reply-To: <CABkS=xp5XcskU6uJoM-7iHy3taPm3M=QiD29UO9EfToDTR-Fug@mail.gmail.com>
References: <CABkS=xqD0mukebja+w3aurrhcUJ4Gp=DcOpkh+BDnCpOKr32CA@mail.gmail.com>
	<CAAYhoXLzsB-o=5VNBaRsR9xdS8bm=j5zKxe8ideg5-jmKU=fKw@mail.gmail.com>
	<CABkS=xrLBX2A1ZLO9gx7mW2yQ2e9LTjyDaNOqJXH2DFADS4KYg@mail.gmail.com>
	<CABkS=xp5XcskU6uJoM-7iHy3taPm3M=QiD29UO9EfToDTR-Fug@mail.gmail.com>
Message-ID: <CABkS=xqFUnp0D1SFNZh+n8XNUbxTKYACYQ__+tTW49ffL_uPbQ@mail.gmail.com>

Let me correct my last email.

If I use rtp_secure_media instead of sip_secure_media, the outgoing call
uses RTP and not SRTP.

rtp_secure_media was supposed to have been introduced in 1.2.9, so I wonder
what the difference is?


On Thu, Aug 15, 2013 at 3:52 PM, Peter <eidevm5 at gmail.com> wrote:

> Finally got it going.  I don't know how many combinations I tried.
>
> All I needed was the sip_secure_media (or rtp_secure_media, which is the
> new name) set to true in the dialplan on the SBC.
>
>
> On Wed, Aug 14, 2013 at 11:42 AM, Peter <eidevm5 at gmail.com> wrote:
>
>> Hi Carlos.
>>
>> Didn't realise rtp_secure_media existed.  After searching I saw:
>>
>>
>> https://wiki.freeswitch.org/wiki/Release_Notes#rtp_secure_media_.28was_sip_secure_media.29
>>
>> which says it was introduced in 1.2.9
>>
>> However, it's a little ambiguous as to whether sip_secure_media was
>> deprecated.
>>
>> Anyway, I tried using rtp_secure_media instead, but I still can't get
>> SRTP working.
>>
>>
>> I did some testing with some other SIP clients.   In particular,
>> csipsimple.  On the client, if I set SRTP to be optional, the media stream
>> uses RTP.   However, if I set SRTP to be mandatory, when I try to call it,
>> Freeswitch receives:
>>
>>    SIP/2.0 488 Not Acceptable Here
>>
>> Which seems to indicate that something is not is not right with the SRTP
>> setup.
>>
>> There's a full debug from the FS1 (the freeswitch server where the
>> csipsimple client is registered to) at:
>>
>> http://pastebin.freeswitch.org/21295
>>
>> Note in the debug I have sdp_secure_savp_only set to true.   I've tried
>> disabling this setting, but get the same result.
>>
>> Thanks
>>
>> Peter
>>
>>
>>
>>
>>
>> On Tue, Aug 13, 2013 at 11:06 PM, Carlos Flor <jackal at cybershroud.net>wrote:
>>
>>> Try using rtp_secure_media=true instead of sip_secure_media.  If you are
>>> trying to set it on the b-leg, you probably want to use export instead of
>>> set, or use nolocal:rtp_secure_media.
>>>
>>> Hope that helps.
>>>
>>>
>>> On Mon, Aug 12, 2013 at 10:26 PM, Peter <eidevm5 at gmail.com> wrote:
>>>
>>>> In my environment, I have the following (simplified) setup:
>>>>
>>>> FS1  ----  FS SBC ---  FS2
>>>>
>>>> Phones registered to FS1 (100x) use TLS/SRTP and phones registered to
>>>> FS2 (200x) use SIP/RTP
>>>>
>>>> FS1 has inbound-bypass-media set to true to allow SRTP peer to peer and
>>>> direct to the SBC.
>>>>
>>>> If I make an inbound call (eg: 1000 to 2000), SRTP is correctly
>>>> established between the phone and SBC with RTP on the other side of the SBC
>>>> to the internal phone.
>>>>
>>>> However, when I try it the other way, I can't get SRTP established from
>>>> the SBC to the external phone.
>>>>
>>>> I've been using https://wiki.freeswitch.org/wiki/Secure_RTP as a guide.
>>>>
>>>> I've even tried explicitly setting sip_secure_media to true on the SBC
>>>> and FS1.
>>>>
>>>> The dialplan on the SBC has:
>>>>
>>>>   <extension name="outgoing">
>>>>         <condition field="destination_number"
>>>> expression="^(10[0-9][0-9])$">
>>>>             <action application="set" data="sip_secure_media=true"/>
>>>>             <action application="bridge" data="sofia/external/${
>>>> destination_number}@10.1.1.204"/>
>>>>         </condition>
>>>>   </extension>
>>>>
>>>>
>>>> And on FS1, the dialplan has:
>>>>
>>>>    <extension name="Local-Numbers">
>>>>       <condition field="destination_number"
>>>> expression="^(10[01][0-9])$">
>>>>         <action application="export" data="dialed_extension=$1"/>
>>>>         <action application="set" data="sip_secure_media=true"/>
>>>>         <action application="bridge" data="user/${dialed_extension}@
>>>> ${domain_name}"/>
>>>>       </condition>
>>>>     </extension>
>>>>
>>>>
>>>> Note that I've been testing this against two phones with SRTP enabled,
>>>> but only one that is using TLS.  I get the same result calling each phone.
>>>>
>>>> On a related point, what it the step required for a TLS connection from
>>>> the SBC to the phone?   I'm assume the phone just needs the CA cert from
>>>> the SBC.  Correct?
>>>>
>>>> Any information as to where I'm going wrong will be gratefully accepted.
>>>>
>>>> Thanks
>>>>
>>>> Peter
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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 ssinyagin at yahoo.com  Thu Aug 15 10:44:02 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Wed, 14 Aug 2013 23:44:02 -0700 (PDT)
Subject: [Freeswitch-users] Recording conference only when a
	particular	user is connected
In-Reply-To: <CADcWJKFsMnW47mOd_2F=s-nnhOoMkH2wnH07mBSfeHK+aK9nyA@mail.gmail.com>
References: <CADcWJKFsMnW47mOd_2F=s-nnhOoMkH2wnH07mBSfeHK+aK9nyA@mail.gmail.com>
Message-ID: <1376549042.31131.YahooMailNeo@web126201.mail.ne1.yahoo.com>

hi Matt, 

if you launch the "record" application before the user is transferred to the conference,it will exist only in his channel, and should stop when the channel is ended.

but this needs to be tested :)





>________________________________
> From: Matthew Cordes <cordes.matthew at gmail.com>
>To: freeswitch-users at lists.freeswitch.org 
>Sent: Thursday, August 15, 2013 5:06 AM
>Subject: Re: [Freeswitch-users] Recording conference only when a particular	user is connected
> 
>
>
>Hi Stanislav, 
>
>
>
>For a) (below) how do I stop the recording? Will that happen automatically when the user hangs up even though there are others still in the conference? If so, this sounds like it could work. 
>
>
>Thanks, 
>
>-Matt
>
>
>
>
>On Wed, Aug 14, 2013 at 7:19 PM, <freeswitch-users-request at lists.freeswitch.org> wrote:
>
>---------- Forwarded message ----------
>>From:?Stanislav Sinyagin <ssinyagin at yahoo.com>
>>To:?FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>Cc:?
>>Date:?Wed, 14 Aug 2013 15:26:12 -0700 (PDT)
>>Subject:?Re: [Freeswitch-users] Recording conference only when a particular user	is connected
>>
>>some possible scenarios, need further digging:
>>
>>a) for that user, execute the "record" dialplan application, then transfer to the conference extension. It can be achieved by a1) setting a variable in the user directory entry, and then matching the variable in the dialplan; a2) defining a separate context for the user.
>>
>>b) an external script would monitor the ESL events and execute uuid_record when needed
>>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 sravi123 at yahoo.com  Thu Aug 15 11:44:55 2013
From: sravi123 at yahoo.com (Ravi)
Date: Thu, 15 Aug 2013 00:44:55 -0700 (PDT)
Subject: [Freeswitch-users] Recording calls with Freeswitch - log says
	recorded, but no file found
In-Reply-To: <CAKzWOxWk9bUD4skaAO1cpYJC2kAiY6jK7+77DCQpBardNPsKcg@mail.gmail.com>
References: <1376487277.65226.YahooMailNeo@web160503.mail.bf1.yahoo.com>
	<CAKzWOxWk9bUD4skaAO1cpYJC2kAiY6jK7+77DCQpBardNPsKcg@mail.gmail.com>
Message-ID: <1376552695.54693.YahooMailNeo@web160505.mail.bf1.yahoo.com>

Thanks Michael. 

I got the log with the "debug" level output. The pastebin link for the log is :
http://pastebin.freeswitch.org/21301

The call is made from extension 1020 to 1030. The dialplan code snippet is given below as well. If any other information is required, please let me know. 


<extension name="003">
??? <condition field="destination_number" expression="^003$">
??????? <action application="set" data="record_file_name={recordings_
 dir}/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.wav" inli$
??????? <action application="record_session" data="{record_file_name}"/>
??????? <action application="bridge" data="user/1003"/>
??? </condition>
</extension>


Please review and let me know where I am stuck, and what can be done. Freeswitch crashed a couple of times, while I was testing. I will mail that in a separate post. 

Thanks again !
Ravi




________________________________
 From: Michael Collins <msc at freeswitch.org>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Wednesday, August 14, 2013 11:32 PM
Subject: Re: [Freeswitch-users] Recording calls with Freeswitch - log says recorded, but no file found
 


You'll need debug level output. That looks like it's info level output. The freeswitch.log file should have debug level output. Also, if you're on the freeswitch console (not the fs_cli) then info is the default level. Try "console loglevel debug" and then retest and re-capture. Also, best place to put fs log data is: pastebin.freeswitch.org. Use "FreeSWITCH Log" as the syntax highlighting.

-MC




On Wed, Aug 14, 2013 at 6:34 AM, Ravi <sravi123 at yahoo.com> wrote:

Hello Everyone !
>
>
>I am trying to record calls received in an extension (1003). The dialplan is given below 
>
>
>
><extension name="003">
>??? <condition field="destination_number" expression="^003$">
>??????? <action application="set" data="record_file_name={recordings_
 dir}/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.wav" inli$
>??????? <action application="record_session" data="{record_file_name}"/>
>??????? <action application="bridge" data="user/1003"/>
>??? </condition>
></extension>
>
>
>
>When I call this extension from an extension 1006, this is what I get in the log. But when I go to the folder /usr/local/freeswitch/recordings/ there are no files at all. I am just wondering if this has got to do with any file permissions, or if I am missing something. Please help. 
>
>
>
>Thanks.
>Ravi
>
>
>
>
>freeswitch at bfree-server> 2013-08-14 06:53:51.185709 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/1006 at 10.0.0.16 [2d3314f4-0480-11e3-9a2f-4d1cb7d8b5b4]
>2013-08-14 06:53:51.245710 [INFO] mod_dialplan_xml.c:558 Processing Ravi <1006>->1003 in context default
>2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *1 execute_extension::dx XML features
>2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1006.2013-08-14-06-53-51.wav
>2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *3 execute_extension::cf XML features
>2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *4 execute_extension::att_xfer XML features
>2013-08-14 06:53:51.245710 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/sip:1003 at 10.0.0.10:5060 [2d3d1a26-0480-11e3-9a4d-4d1cb7d8b5b4]
>2013-08-14
 06:53:51.445707 [NOTICE] sofia.c:5884 Ring-Ready sofia/internal/sip:1003 at 10.0.0.10:5060!
>2013-08-14 06:53:51.445707 [INFO] switch_ivr_originate.c:1190 Sending early media
>2013-08-14 06:53:51.445707 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1006 at 10.0.0.16!
>2013-08-14 06:53:52.705700 [NOTICE] sofia.c:6547 Channel [sofia/internal/sip:1003 at 10.0.0.10:5060] has been answered
>2013-08-14 06:53:52.725708 [NOTICE] switch_ivr_originate.c:3437 Channel [sofia/internal/1006 at 10.0.0.16] has been answered
>2013-08-14 06:54:05.005696 [NOTICE] sofia.c:715 Hangup sofia/internal/sip:1003 at 10.0.0.10:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>2013-08-14 06:54:05.025711 [NOTICE] switch_ivr_bridge.c:1575 Hangup sofia/internal/1006 at 10.0.0.16 [CS_EXECUTE] [NORMAL_CLEARING]
>2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1567 Session 139 (sofia/internal/sip:1003 at 10.0.0.10:5060) Ended
>2013-08-14 06:54:05.025711 [NOTICE]
 switch_core_session.c:1571 Close Channel sofia/internal/sip:1003 at 10.0.0.10:5060 [CS_DESTROY]
>2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1567 Session 138 (sofia/internal/1006 at 10.0.0.16) Ended
>2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1571 Close Channel sofia/internal/1006 at 10.0.0.16 [CS_DESTROY]
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 william.king at quentustech.com  Thu Aug 15 12:11:53 2013
From: william.king at quentustech.com (William King)
Date: Thu, 15 Aug 2013 01:11:53 -0700
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <20130812053755.96624167@mail.tritonwest.net>
References: <20130812053755.96624167@mail.tritonwest.net>
Message-ID: <520C8D49.3000402@quentustech.com>

Dave,

Know of any current implementations? or would this be entirely new
functionality?

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 08/11/2013 10:37 PM, Dave R. Kompel wrote:
> William,
>  
> The business case is for those of us that use FreeSWITCH as a main
> switch at a CLEC, where we have our own prefix assignments. In order to
> do SMS on those numbers we need to interconnect with a SMS backbone
> provider. SMPP is you're only option for that...
>  
> --Dave
> 
>     ------------------------------------------------------------------------
>     *From:* William King [mailto:william.king at quentustech.com]
>     *To:* freeswitch-users at lists.freeswitch.org
>     *Sent:* Fri, 09 Aug 2013 00:00:33 -0700
>     *Subject:* Re: [Freeswitch-users] Potential FreeSWITCH support for SMPP
> 
>     >From what I've seen, retail level SMS carriers(not sms aggregators)
>     usually offer a mix of HTTP, XMPP, SIP SIMPLE, or SMPP interfaces. I've
>     seen the higher volume SMS connections prefer SMPP or HTTP.
> 
>     I'm exploring to see if there is a business case for FreeSWITCH support
>     for SMPP(either direct support, or interface support through an
>     independent application) that is not already covered by the current FS
>     feature set.
> 
>     William King
>     Senior Engineer
>     Quentus Technologies, INC
>     1037 NE 65th St Suite 273
>     Seattle, WA 98115
>     Main: (877) 211-9337
>     Office: (206) 388-4772
>     Cell: (253) 686-5518
>     william.king at quentustech.com <mailto:william.king at quentustech.com>
> 
>     On 08/08/2013 04:23 AM, Raymond Chandler wrote:
>     > So, to the best of my knowledge, SMPP is strictly for SMS so you can
>     > route SMS to your clients via FS to SIMPLE / dingaling / etc clients
>     >
>     > -Ray
>     >
>     > On Aug 7, 2013 11:05 PM, "Victor Chukalovskiy"
>     > <victor.chukalovskiy at gmail.com
>     <mailto:victor.chukalovskiy at gmail.com>
>     <mailto:victor.chukalovskiy at gmail.com
>     <mailto:victor.chukalovskiy at gmail.com>>>
>     > wrote:
>     >
>     > So am I as well as another ClueCon attendee I spoke to.
>     >
>     > What is the usage scenario you are looking at?
>     >
>     > On 13-08-07 06:46 PM, Dmitry Lysenko wrote:
>     >> Yes. I am interested.
>     >>
>     >> Dmitry.
>     >>
>     >>
>     >> 2013/8/8 William King <william.king at quentustech.com
>     <mailto:william.king at quentustech.com>
>     >> <mailto:william.king at quentustech.com
>     <mailto:william.king at quentustech.com>>>
>     >>
>     >> Is anyone on this list interested in FreeSWITCH support for
>     >> SMPP for SMS
>     >> messages?
>     >>
>     >> For more information about the SMPP protocol checkout:
>     >> http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
>     >>
>     >> If so feel free to contact me on or off this list.
>     >> --
>     >> William King
>     >> Senior Engineer
>     >> Quentus Technologies, INC
>     >> 1037 NE 65th St Suite 273
>     >> Seattle, WA 98115
>     >> Main: (877) 211-9337 <tel:%28877%29%20211-9337>
>     >> Office: (206) 388-4772 <tel:%28206%29%20388-4772>
>     >> Cell: (253) 686-5518 <tel:%28253%29%20686-5518>
>     >> william.king at quentustech.com
>     <mailto:william.king at quentustech.com>
>     <mailto:william.king at quentustech.com
>     <mailto:william.king at quentustech.com>>
>     >>
>     >>
>     _________________________________________________________________________
>     >> Professional FreeSWITCH Consulting Services:
>     >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     >> http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com/>
>     >>
>     >> 
>     >>  </>
>     >>
>     >> Official FreeSWITCH Sites
>     >> http://www.freeswitch.org <http://www.freeswitch.org/>
>     >> http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>     >> http://www.cluecon.com <http://www.cluecon.com/>
>     >>
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >> <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org <http://www.freeswitch.org/>
>     >>
>     >>
>     >>
>     >>
>     >>
>     _________________________________________________________________________
>     >> Professional FreeSWITCH Consulting Services:
>     >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     >> http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com/>
>     >>
>     >> 
>     >>  </>
>     >>
>     >> Official FreeSWITCH Sites
>     >> http://www.freeswitch.org <http://www.freeswitch.org/>
>     >> http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>     >> http://www.cluecon.com <http://www.cluecon.com/>
>     >>
>     >> FreeSWITCH-users mailing list
>     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >>
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     >> http://www.freeswitch.org <http://www.freeswitch.org/>
>     >
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     > http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com/>
>     >
>     > 
>     >  </>
>     >
>     > Official FreeSWITCH Sites
>     > http://www.freeswitch.org <http://www.freeswitch.org/>
>     > http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>     > http://www.cluecon.com <http://www.cluecon.com/>
>     >
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org <http://www.freeswitch.org/>
>     >
>     >
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     > http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com/>
>     >
>     > 
>     >  </>
>     >
>     > Official FreeSWITCH Sites
>     > http://www.freeswitch.org <http://www.freeswitch.org/>
>     > http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>     > http://www.cluecon.com <http://www.cluecon.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 <http://www.freeswitch.org/>
>     >
> 
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com <http://www.freeswitchsolutions.com/>
> 
>     
>      </>
> 
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org <http://www.freeswitch.org/>
>     http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>     http://www.cluecon.com <http://www.cluecon.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 <http://www.freeswitch.org/>
> 
>  
>  
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ssinyagin at yahoo.com  Thu Aug 15 12:12:27 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Thu, 15 Aug 2013 01:12:27 -0700 (PDT)
Subject: [Freeswitch-users] Recording calls with Freeswitch - log
	says	recorded, but no file found
In-Reply-To: <1376552695.54693.YahooMailNeo@web160505.mail.bf1.yahoo.com>
References: <1376487277.65226.YahooMailNeo@web160503.mail.bf1.yahoo.com>	<CAKzWOxWk9bUD4skaAO1cpYJC2kAiY6jK7+77DCQpBardNPsKcg@mail.gmail.com>
	<1376552695.54693.YahooMailNeo@web160505.mail.bf1.yahoo.com>
Message-ID: <1376554347.89086.YahooMailNeo@web126204.mail.ne1.yahoo.com>

is the dollar sign before {recordings_ dir} and {record_file_name} eaten by your mail client, or it's really so in your XML?







>________________________________
> From: Ravi <sravi123 at yahoo.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Thursday, August 15, 2013 9:44 AM
>Subject: Re: [Freeswitch-users] Recording calls with Freeswitch - log says	recorded, but no file found
> 
>
>
>Thanks Michael. 
>
>I got the log with the "debug" level output. The pastebin link for the log is :
>http://pastebin.freeswitch.org/21301
>
>The call is made from extension 1020 to 1030. The dialplan code snippet is given below as well. If any other information is required, please let me know. 
>
>
><extension name="003">
>??? <condition field="destination_number" expression="^003$">
>??????? <action application="set" data="record_file_name={recordings_
 dir}/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.wav" inli$
>??????? <action application="record_session" data="{record_file_name}"/>
>??????? <action application="bridge" data="user/1003"/>
>??? </condition>
></extension>
>
>
>
>Please review and let me know where I am stuck, and what can be done. Freeswitch crashed a couple of times, while I was testing. I will mail that in a separate post. 
>
>Thanks again !
>Ravi
>
>
>
>
>
>________________________________
> From: Michael Collins <msc at freeswitch.org>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Wednesday, August 14, 2013 11:32 PM
>Subject: Re: [Freeswitch-users] Recording calls with Freeswitch - log says recorded, but no file found
> 
>
>
>You'll need debug level output. That looks like it's info level output. The freeswitch.log file should have debug level output. Also, if you're on the freeswitch console (not the fs_cli) then info is the default level. Try "console loglevel debug" and then retest and re-capture. Also, best place to put fs log data is: pastebin.freeswitch.org. Use "FreeSWITCH Log" as the syntax highlighting.
>
>-MC
>
>
>
>
>On Wed, Aug 14, 2013 at 6:34 AM, Ravi <sravi123 at yahoo.com> wrote:
>
>Hello Everyone !
>>
>>
>>I am trying to record calls received in an extension (1003). The dialplan is given below 
>>
>>
>>
>><extension name="003">
>>??? <condition field="destination_number" expression="^003$">
>>??????? <action application="set" data="record_file_name={recordings_
 dir}/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.wav" inli$
>>??????? <action application="record_session" data="{record_file_name}"/>
>>??????? <action application="bridge" data="user/1003"/>
>>??? </condition>
>></extension>
>>
>>
>>
>>When I call this extension from an extension 1006, this is what I get in the log. But when I go to the folder /usr/local/freeswitch/recordings/ there are no files at all. I am just wondering if this has got to do with any file permissions, or if I am missing something. Please help. 
>>
>>
>>
>>Thanks.
>>Ravi
>>
>>
>>
>>
>>freeswitch at bfree-server> 2013-08-14 06:53:51.185709 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/1006 at 10.0.0.16 [2d3314f4-0480-11e3-9a2f-4d1cb7d8b5b4]
>>2013-08-14 06:53:51.245710 [INFO] mod_dialplan_xml.c:558 Processing Ravi <1006>->1003 in context default
>>2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *1 execute_extension::dx XML features
>>2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1006.2013-08-14-06-53-51.wav
>>2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *3 execute_extension::cf XML features
>>2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *4 execute_extension::att_xfer XML features
>>2013-08-14 06:53:51.245710 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/sip:1003 at 10.0.0.10:5060 [2d3d1a26-0480-11e3-9a4d-4d1cb7d8b5b4]
>>2013-08-14
 06:53:51.445707 [NOTICE] sofia.c:5884 Ring-Ready sofia/internal/sip:1003 at 10.0.0.10:5060!
>>2013-08-14 06:53:51.445707 [INFO] switch_ivr_originate.c:1190 Sending early media
>>2013-08-14 06:53:51.445707 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1006 at 10.0.0.16!
>>2013-08-14 06:53:52.705700 [NOTICE] sofia.c:6547 Channel [sofia/internal/sip:1003 at 10.0.0.10:5060] has been answered
>>2013-08-14 06:53:52.725708 [NOTICE] switch_ivr_originate.c:3437 Channel [sofia/internal/1006 at 10.0.0.16] has been answered
>>2013-08-14 06:54:05.005696 [NOTICE] sofia.c:715 Hangup sofia/internal/sip:1003 at 10.0.0.10:5060 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
>>2013-08-14 06:54:05.025711 [NOTICE] switch_ivr_bridge.c:1575 Hangup sofia/internal/1006 at 10.0.0.16 [CS_EXECUTE] [NORMAL_CLEARING]
>>2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1567 Session 139 (sofia/internal/sip:1003 at 10.0.0.10:5060) Ended
>>2013-08-14 06:54:05.025711 [NOTICE]
 switch_core_session.c:1571 Close Channel sofia/internal/sip:1003 at 10.0.0.10:5060 [CS_DESTROY]
>>2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1567 Session 138 (sofia/internal/1006 at 10.0.0.16) Ended
>>2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1571 Close Channel sofia/internal/1006 at 10.0.0.16 [CS_DESTROY]
>>
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
>>consulting at freeswitch.org
>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
>>http://www.freeswitch.org
>>http://wiki.freeswitch.org
>>http://www.cluecon.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
>>
>>
>
>
>-- 
>Michael S Collins
>Twitter: @mercutioviz
>http://www.FreeSWITCH.org
>http://www.ClueCon.com
>http://www.OSTAG.org
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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
>
>
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 itsme.kunnu at gmail.com  Thu Aug 15 13:12:43 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Thu, 15 Aug 2013 14:42:43 +0530
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CALn233ovuLR8rgD==bqiOi+Fa=yDwS8azsrD3cWkBz9sz7Mw7w@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
	<CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
	<CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
	<CALn233ovuLR8rgD==bqiOi+Fa=yDwS8azsrD3cWkBz9sz7Mw7w@mail.gmail.com>
Message-ID: <CAMvhjJut4W_hwnAG45o2wFWCOKknwPzE8vFGrkNv6JoZZA8o1Q@mail.gmail.com>

I also think that there may be some network problem as my windows pc is not
showing that some linux machine is connected but the linux machine shows
the windows system connected to it. Also can i install the softphone on the
same linux machine on which my freeswitch runs and can check whether it
gets connected or not...???
 On Aug 15, 2013 5:10 AM, "Mitch Capper" <mitch.capper at gmail.com> wrote:

> You may want to try FSClient its a freeswitch based windows soft phone and
> you can connect fs_cli (or fs_logger.pl) to it and see what happens when
> it tries to connect to your server.
>
> ~mitch
>
>
> On Wed, Aug 14, 2013 at 4:06 PM, Peter <eidevm5 at gmail.com> wrote:
>
>> My mistake, I meant loglevel rather than logging.
>>
>> So when you say "no progress" are you seeing nothing at all on the FS
>> console?
>>
>> If so, then that indicates a network problem as FS isn't even seeing the
>> registration requests.
>>
>> Can you ping the windows box from the FS server?
>>
>>
>> On Wed, Aug 14, 2013 at 7:04 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> I went for "sofia loglevel 5" instead of "sofia logging 5" but still no
>>> progress in connection of softphone...:-(
>>> Thanks
>>> Ashish Mishra
>>> On Aug 14, 2013 2:14 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>>>
>>>> Also the command that you had mentioned "sofia logging all 5" gives me
>>>> an error message :
>>>> Unknown command [logging]
>>>> On Aug 14, 2013 7:24 AM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>
>>>>> Connect to the FS console with fs_cli and do
>>>>>
>>>>> sofia logging all 5
>>>>>
>>>>> then try to register your softphone.    If you don't even see a
>>>>> registration attempt on the FS console, it means you have a network problem.
>>>>>
>>>>> If the network isn't a problem, the debug output should give you a
>>>>> clue as to what the problem is.
>>>>>
>>>>>
>>>>>
>>>>> On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>>
>>>>>> Thank you Peter...you mean to say that i should first turn on the
>>>>>> fs_cli and then retry to connect the softphone...??? I would also like to
>>>>>> remind you that fs_cli and freeswitch are running on the same machine and i
>>>>>> am trying to connect thru a network cable my windows 8 pc that has the
>>>>>> softphone on it...
>>>>>> Regards
>>>>>> Ashish Mishra
>>>>>> On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>>>
>>>>>>> Have you turned on debugging from the FS cli and seen if any
>>>>>>> registration requests come in?
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <
>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>
>>>>>>>> I installed freeswitch on my ubuntu 12.04 machine...but when i am
>>>>>>>> trying to connect the softphone installed on my windows 8 pc with ubuntu
>>>>>>>> machine the softphone gives me an error that account not enabled...i have
>>>>>>>> used a network cable to connect the two machines...also the firewall in
>>>>>>>> both the machines is disabled...
>>>>>>>> Kindly help..
>>>>>>>>
>>>>>>>> Ashish Mishra
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 denis.gasparin at edistar.com  Thu Aug 15 13:25:08 2013
From: denis.gasparin at edistar.com (Denis Gasparin)
Date: Thu, 15 Aug 2013 11:25:08 +0200 (CEST)
Subject: [Freeswitch-users] Httapi: getting digits in playback
In-Reply-To: <1224793464.3953.1376498943199.JavaMail.root@mailserver.edistar.com>
Message-ID: <907894905.4003.1376558708222.JavaMail.root@mailserver.edistar.com>

Hi. 


The problem seems definetely due to bind_digit_action (used by <bind> tag of httapi). 
I tried to execute the "play_and_get_digits" dialplan application in xml returned to freeswitch and reading a channel var with stored digits and all is working fine. 


This is the working xml: 



<document type="text/freeswitch-httapi"> 
<work> 
<execute application="play_and_get_digits" data="1 4 1 5000 # ivr/ivr-please_enter_the_number_where_we_can_reach_you.wav '' myChannelVar \d+" /> 
<getVariable name=" myChannelVar "/> 
</work> 
</document> 


This is not the first time I have problem with bind_digit_action. I tried the same configuration on different linux distributions (Sles 11.1 and Centos 5.9) and mac os x with the same results: bind_digit_action sometimes works sometimes not. 


A question for Freeswitch developers: why has been bind_digit_action used in place of switch_play_and_get_digits for developing httapi? 
Lua mod uses switch_play_and_get_digits and has a more predictable and stable behaviour than httapi. 
I tried successfully to make a patch for httapi in order to use play_and_get_digits. If useful I can post it to jira. 


Thank you in advance 
Denis Gasparin 



----- Messaggio originale -----


Da: "Denis Gasparin" <denis.gasparin at edistar.com> 
A: "FreeSWITCH Users Help" <freeswitch-users at lists.freeswitch.org> 
Inviato: Mercoled?, 14 agosto 2013 18:49:03 
Oggetto: Httapi: getting digits in playback 


Hi. 


After playback I ask to the user to digit a number between 1 and 9999 with or without terminating the input with '#'. 


The first httapi xml I tried was the following: 



<document type="text/freeswitch-httapi"> 
<work> 
<playback file="ivr/please_enter_number_between_1_9999.wav" digit-timeout="5000" name="user_input" terminators="#"> 
<bind>~\d{1,4}#{0,1}</bind> 
</playback> 
</work> 
</document> 


Freeswitch calls my action as soon as the user input the first digit. 
So I modified the xml in order to use two bindings: 



<document type="text/freeswitch-httapi"> 
<work> 
<playback file="ivr/please_enter_number_between_1_9999.wav" digit-timeout="5000" name="user_input" terminators="#"> 
<bind>~\d{1,3}#{0,1}</bind> 
<bind>~\d{4}#{0,1}</bind> 
</playback> 
</work> 
</document> 


Using this xml I get the user input only if the user press "#" after inserting the number. 
If the user presses 123 and then waits for timeout, Freeswitch doesn't send any digits to my action (but sometimes it does). Why? 


>From the logs I see that the digits are always received from Freeswitch. 


Thank you in advance for your help. 
Denis Gasparin 
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From Adam.Lappe at qsc.de  Thu Aug 15 13:56:55 2013
From: Adam.Lappe at qsc.de (Lappe, Adam)
Date: Thu, 15 Aug 2013 11:56:55 +0200
Subject: [Freeswitch-users] SIP TLS Issues
Message-ID: <F1EBD7E7528EA446BB9898D22914AA3B3D947FEAE2@CGNMSG01.oc.qsc.de>

Hi all,

Some more things I tried so far:

openssl x509 -noout -modulus -in agent.pem | openssl md5
(stdin)= ebdfb317206ba89d07217c06e1f0d6eb
openssl rsa -noout -modulus -in agent.pem | openssl md5
(stdin)= ebdfb317206ba89d07217c06e1f0d6eb

At least the certificate and private key in the agent.pem are correct.

There is no output on the cli when I try to register a phone.

My guess is that the content of agent.pem and/or cafile.pem is wrong.

Can someone please confirm this?

Best regards,
Adam


Wed, 14 Aug, 2013 at 16:07 PM, Adam <ala at qsc.de>:

Hi all,

i am trying to configure FreeSWITCH to speak TLS with all Clients.
I followed the tutorial on http://wiki.freeswitch.com/wiki/SIP_TLS but I am still not sure what key / cert belongs in which file.

I have a SSL123 Thawte Wildcard Certificate.
Am I supposed to cat this cert + priv. key into agent.pem and the primary and secondary intermediate into the cafile.pem?

I did this and set the right permissions. The internal sofia profile on port 5061 (TLS) is RUNNING.

But no client (for example Polycom VVX1500) can register now.
If I set it TCP and Port 5060 (which is RUNNING as well) everything works fine.

Wireshark shows me the following

Client    ->         FS                    Client Hello
FS        ->         Client                Alert (Level Fatal, Description: Handshake Failure)

I also tested openssl s_client -connect (IP):5061 -showcerts but it only says:
CONNECTED(00000003)
139847050823328:error:14077410:SSL routines:SSL23_GET_SERVER_HELLO:sslv3 alert handshake failure:s23_clnt.c:724:
---
no peer certificate available
---
No client certificate CA names sent
---
SSL handshake has read 7 bytes and written 225 bytes
---
New, (NONE), Cipher is (NONE)
Secure Renegotiation IS NOT supported
Compression: NONE
Expansion: NONE
---

I guess the problem is the agent.pem and/or cafile.pem

agent.pem looks like this
-----BEGIN CERTIFICATE-----
(Thawte SSL123 Wildcard Web Certificate)
-----END CERTIFICATE-----
-----BEGIN RSA PRIVATE KEY-----
(Unencrypted Private Key)
-----END RSA PRIVATE KEY-----

cafile.pem like that:
-----BEGIN CERTIFICATE-----
(Thawte Primary Intermediate)
-----END CERTIFICATE-----
-----BEGIN CERTIFICATE-----
(Thawte Secondary Intermediate
-----END CERTIFICATE-----

Any suggestions?

Thanks in advance,
Adam
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From nandy1925 at gmail.com  Thu Aug 15 13:59:28 2013
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 15 Aug 2013 17:59:28 +0800
Subject: [Freeswitch-users] freetdm rxgain txgain
In-Reply-To: <520C4938.4010202@xtronics.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
	<CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
	<CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
	<520C4938.4010202@xtronics.com>
Message-ID: <CAHKqMb7uoZKy0O4NAdSnFxdCzxv5sTJ5S4rJYDncv6dgyjPNEA@mail.gmail.com>

Hi Karl,

Did you test the milliwatt tone using the FXS port? I asked because you
said it's not sending the ringing signal. Do you mean the phones did not
ring. The Digium cards require a separate power supply feed. Without it -
it can't generate the 100V ringing voltage. This is very obvious but let's
not discounting this possibility. :-)

/Nandy


On Thu, Aug 15, 2013 at 11:21 AM, Karl Schmidt <karl at xtronics.com> wrote:

> Looks like rxgain and txgain are not working - I was testing using the
> cool mW extension (thanks -
> whoever put that in)  and could not change the level - did some digging
> and dahdi_diag shows that
> the gains don't change.
>
> ; Aug 14 18:16:51 malaysia kernel: [350647.964612] dahdi: rxgain:
> ffffffffa02e7c70, txgain:
> ffffffffa02e7c70, gainalloc: 0
>
> ; Aug 14 18:20:33 malaysia kernel: [350869.980742] dahdi: rxgain:
> ffffffffa02e7c70, txgain:
> ffffffffa02e7c70, gainalloc: 0
>
> Even tried changes over a reboot.
>
>
> This card has absorbed a lot of time:
> (Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)) [OT]:(and
> now I find that it fails
> to put out a real ring signal.  Apparently, whoever designed it did not
> know the difference between
> RMS and peak voltage... fails to ring 3 out of 4 modems I tested. Gump,
> hiss,groan).
>
> I can go in and modify the modem so it can detect the weak ring signal,
> but out of the box the audio
> levels on the card are way off ( they were also off in Asterisk ) and if I
> want to have this working
> well , I need to tweak those levels.
>
>
> BTW - I can use the 1 dBm test tone stream to monitor the transmitted
> level - is there a way to
> monitor a level coming back the other way?
>
>
>
>
>
>
> --------------------------------------------------------------------------------
> Karl Schmidt                                  EMail Karl at xtronics.com
> Transtronics, Inc.                              WEB
> http://secure.transtronics.com
> 3209 West 9th Street                             Ph (785) 841-3089
> Lawrence, KS 66049                              FAX (785) 841-0434
>
> The lawyer was about two bubbles off plumb.
>
> --------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 peter at olssononline.se  Thu Aug 15 15:01:37 2013
From: peter at olssononline.se (Peter Olsson)
Date: Thu, 15 Aug 2013 13:01:37 +0200
Subject: [Freeswitch-users] deny_refer_requests failure
In-Reply-To: <CAAQxoQH9R_nb5hv=nMqAknwL+ZsSaEiL3EcUF5jKnb=-YMva4w@mail.gmail.com>
References: <CAAQxoQH9R_nb5hv=nMqAknwL+ZsSaEiL3EcUF5jKnb=-YMva4w@mail.gmail.com>
Message-ID: <CAPEdYTh5DUjxfGtyFk+QfT2HaaFaZZnpyyJfX5sgoesp+dxbyw@mail.gmail.com>

I believe it only works when using REFER with replaces headers set. Are you
trying REFER as a blind transfer maybe? In that case it won't work.
However, it probably should be implemented to work in that case as well, so
please file a Jira for it.

/Peter


2013/8/14 Ryan Gard <ryangard at gmail.com>

> Hey,
>
> Running into issues with deny_refer_requests. I have verified that it is
> setting the variable appropriately on both the a-leg and b-leg of the call,
> but it still processes the refer request without a second thought upon
> receiving it.
>
> Is there anything specific on top of setting the variable to 'true' I
> should be keeping tabs on? It doesn't seem to have too much in the way of
> documentation, but the references I have seen say that it should work
> without issue as long as one of the legs (both legs have it set) has it set
> to true.
>
> Thanks :)
>
> --
> Ryan Gard
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveayre at gmail.com  Thu Aug 15 15:12:01 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 15 Aug 2013 12:12:01 +0100
Subject: [Freeswitch-users] Auto-detecting Local IP when no gateway set
In-Reply-To: <CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
Message-ID: <CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>

The entire point is to auto-detect which IP is used for Internet access,
which is obviously impossible without a default gateway. The reason for
that is in the frequent case where a machine may have multiple IP
addresses. So you are going to have to set local_ip_v4 explicitly.

As far as a generic config goes - why not have common configuration files
and handle all the machine-specific stuff in vars.xml, then any
machine-specific stuff is in a single file which'll make it easier to
manage.

If you want to auto-detect the static IP, then modify the init.d script to
first run a script that picks up the current IP and updates vars.xml. That
would need to be done before freeswitch is started.

Thinking about the best way to handle this long term though...
Perhaps you could file a wishlist Jira that allows you to set local_ip_v4
by an interface name. FreeSWITCH should be able to lookup the primary IP of
an interface pretty easily.



On 15 August 2013 03:12, Siri MM <sirimmfs at gmail.com> wrote:

> Hi Jeff,
>
> Couple of reasons why I dont want to specify IP address explicitly:
> 1. Customer is using multiple devices with freeswitch on each, and my
> strategy is to provide a generically configured freeswitch, which would
> work on all
> 2. Customer might at any point in time change the static IP address of
> devices, or decide to use DHCP!
>
>
> On Thu, Aug 15, 2013 at 12:00 PM, Jeff Leung <jleung at v10networks.ca>wrote:
>
>> Why can't you just specify a static IP address instead? Specifying a
>> static IP address sure does make life a heck a lot easier that way.
>>
>>
>> On Wed, Aug 14, 2013 at 5:26 PM, Siri MM <sirimmfs at gmail.com> wrote:
>>
>>>  Hi All,
>>>
>>> I am using freeswitch without local_ip_v4 explicitly set, i.e, I am
>>> dependant on freeswitch auto-detecting the local IP address.
>>>
>>> From the code, what I understand is, in order to determine the local IP
>>> address,  freeswitch tries to ?connect()? using a DGRAM socket with
>>> BROADCAST option set.
>>>
>>> The problem that I am having is, my customer is working in a Local LAN,
>>> and is not very keen on setting the interface gateway. Without gateway set,
>>> connect() returns 'Network unreachable' error.
>>>
>>> Is there any way I could workaround this problem? (apart from setting
>>> local_ip_v4) Also, any other issues forseen if gateway isn't set ont he
>>> interface?
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ivan at c3i.bg  Thu Aug 15 15:37:40 2013
From: ivan at c3i.bg (Ivan)
Date: Thu, 15 Aug 2013 14:37:40 +0300
Subject: [Freeswitch-users] garbled audio with G726-32,
 other codecs are fine
In-Reply-To: <5209D8EE.4090903@c3i.bg>
References: <51FD0E8B.3060702@c3i.bg>
	<CAHgk4BVNA+NOg5ChyYxSddDxhMxgSmWVe1tQ7NiCPRKo6M5BCw@mail.gmail.com>
	<003901ce90b9$749f0090$5ddd01b0$@v10networks.ca>
	<51FDEC5B.2010307@c3i.bg>
	<065260E6-8DAB-4273-8C5A-0D0A48BD175B@jerris.com>
	<5209D8EE.4090903@c3i.bg>
Message-ID: <520CBD84.40806@c3i.bg>

Just found some time to do more tests: in the end it was simply a matter 
of enabling aal2 bitpacking in sofia. I really didn't think that would 
solve the problem since both endpoints had the same problem. (BTW, 
looking back at the interop page with the right set of keywords, I now 
see that someone had the same problem with a linksys SPA2000).

Anyway, thanks to everybody who replied with pointers to bitpacking - I 
assumed wrongly there was no bitpacking at all on G726.

Note: as I'm not knowledgeable enough on G726/bitpacking to update the 
wiki, I've set up a bounty - see the wiki page.



On 08/13/2013 09:57 AM, Ivan wrote:
> Ah, I didn't know there was always bit packing - I guess I should read
> how G726 works, thanks for pointing that.
>
> The thing is, both linksys PAPs and linphone have the same problem, and
> they are not "exotic" endpoints so that makes me think the cause is my
> freeswitch setup. I'll try to test with Xlite when I have a chance, and
> also see if wireshark shows enough codec detail to find out if the
> endpoints get it wrong. I'm of course interested if you have other ideas
> on how to debug that.
>
> ivan
>
>
> On 08/12/2013 05:14 PM, Michael Jerris wrote:
>> There is always bit packing, but there are 2 different ways to do the bit packing.  A lot of devices get it wrong so its worth looking at that.
>>
>> Mike
>>
>> On Aug 4, 2013, at 1:53 AM, Ivan Mitev <imitev at c3i.bg> wrote:
>>
>>> Thanks for the suggestion but I'm testing with G726-32, not AAL2-G726-32
>>> ; so bitpacking shouldn't be used. By the way when I tested with AAL2 to
>>> the linphone client I only got cracks and whitenoise, I've forgot to
>>> mention that in my post.
>>>
>>> That said I've tried to uncomment and set <param name="bitpacking"
>>> value="none"/> in internal.xml ("none" is a wild guess - I couldn't find
>>> any doc on values accepted by this parameter), but that doesn't help.
>>>
>>> Speex: the ATAs don't support it. And being stubborn I'd like to
>>> understand what's wrong with G726 :)
>>>
>>>
>>> On 08/04/2013 05:22 AM, Jeff Leung wrote:
>>>>
>>>> You can turn off G726 AAC bit-packing in spandsp.conf.xml.
>>>>
>>>> By the way, there are other codecs out there you can try. SPEEX comes
>>>> to mind if all your endpoints don?t deal with the PSTN.
>>>>
>>>> *From:*freeswitch-users-bounces at lists.freeswitch.org
>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of
>>>> *Brian Foster
>>>> *Sent:* Saturday, August 3, 2013 2:37 PM
>>>> *To:* FreeSWITCH Users Help
>>>> *Subject:* Re: [Freeswitch-users] garbled audio with G726-32, other
>>>> codecs are fine
>>>>
>>>> AAC bitpacking by any chance? I thought I had a similar issue,
>>>> happened so long ago I cant remember what I did.
>>>>
>>>> Thank you,
>>>>
>>>> Brian Foster
>>>> Project Manager/Owner's Rep.
>>>> Davri Investments, Inc.
>>>> O: 317-787-2686 x2102
>>>> M: 317-600-9753
>>>> E: bdfoster at davri.com <mailto:bdfoster at davri.com>
>>>> Indianapolis, Indiana
>>>>
>>>> Sent from a mobile device.
>>>>
>>>> On Aug 3, 2013 5:20 PM, "Ivan Mitev" <imitev at c3i.bg
>>>> <mailto:imitev at c3i.bg>> wrote:
>>>>
>>>> Hello
>>>>
>>>> I'm migrating an office setup from asterisk to FS and in the process I
>>>> was considering using G726-32 for some bandwidth starved remote
>>>> endpoints. However I only get metallic/garbled audio with that codec
>>>> even when simply playing moh to the endpoint, while other codecs work
>>>> fine (G711U/A, G722, GSM). G732-16 is inaudible, G732-40 sounds
>>>> marginally better but still garbled and really worse than G711.
>>>>
>>>> The setup is FS 1.2.12 from FS' yum repo on a centos6 64bit KVM guest
>>>> (centos6 64bit host). But please don't shoot ! :) - I know about virtual
>>>> environment limitations but for these tests the host is only lightly
>>>> loaded, there aren't any calls to the FS instance except my tests, and
>>>> the fact that it works with other codecs makes me think that
>>>> virtualization is not the issue here. I may be wrong though.
>>>>
>>>> Is there any guide for debugging that kind of problem before reverting
>>>> to a fresh install on bare-metal with the latest HEAD ? Until now I've
>>>> tried:
>>>>
>>>> - improving timers ; but the default soft timer (which I guess uses
>>>> timerd) works best. The time interval between sent packets on a tcpdump
>>>> trace looks identical to the output of "timer_test", so that doesn't
>>>> seem to be a network/jitter problem. And there's no problem with other
>>>> codecs, but maybe G726-XX is specific. For info the guest's clocksource
>>>> is kvm_clock, while the host uses tsc.
>>>>
>>>> - using different endpoints: the production ones are Linksys PAP2
>>>> ("fixed" for 20ms psize, and G726-32 SDP type indentification), but the
>>>> same thing happens with linphone on a fedora 19 laptop.
>>>>
>>>> A call with rtp media going through FS without transcoding - G726-32 to
>>>> G726-32 - works perfectly (I can't hear the difference with G711). The
>>>> problem is only when there's transcoding to G726 (from wav for moh, or
>>>> from any other codec when bridging). I've looked at the wiki, posts,
>>>> changelogs, jira, ..., but am a bit at a loss now.
>>>>
>>>> Any pointers ?
>>>>
>>>> Except that little problem, FS rocks, and I'm happy I can finally ditch
>>>> asterisk. Kudos to the core devs and contributors.
>>>>
>>>> Ivan
>>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 intralanman at freeswitch.org  Thu Aug 15 16:14:53 2013
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 15 Aug 2013 08:14:53 -0400
Subject: [Freeswitch-users] Auto-detecting Local IP when no gateway set
In-Reply-To: <CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
Message-ID: <CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>

Alternatively, you could use xml_curl to serve your configs so that you
don't actually need local_ip_v4 as you can figure it out based on the IP
hitting the web server. Doing that will allow you to set the IP in all of
the configs that need it using functionality that already exists in
FreeSWITCH today.

-Ray
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From GB at cm.nl  Thu Aug 15 16:40:32 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Thu, 15 Aug 2013 14:40:32 +0200
Subject: [Freeswitch-users] Handle individual SIP messages
Message-ID: <FB7D97A214987F458242ACBDF876140738C39E7642@clubvirtual40.ClubMessage.local>

Hello,

Is there a way to process individual SIP messages in the dialplan or somewhere else?
When receiving a 200 OK from our Carrier, I'd like to process (add some custom headers) the 200 OK manually before I send it off to the next hop (SIP Proxy server).

Regards,

Grant
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From mike at jerris.com  Thu Aug 15 17:27:38 2013
From: mike at jerris.com (Michael Jerris)
Date: Thu, 15 Aug 2013 09:27:38 -0400
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <CAKbxfG80vr3b+BULjn5ZhKW6ZjxY6Khy4Ymhk8RWGVFLpuSKoA@mail.gmail.com>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>
	<CE311BFC.78EA0%krice@freeswitch.org>
	<045801ce990e$205b4f90$6111eeb0$@207me.com>
	<CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>
	<049101ce9915$e1cbb5f0$a56321d0$@207me.com>
	<CAOP4xcy6NZG0JKPMYhJp_wxaLqnAjsEkGMgTD5itvBPNdSCPzQ@mail.gmail.com>
	<050301ce993b$4dc37b60$e94a7220$@207me.com>
	<CAKbxfG80vr3b+BULjn5ZhKW6ZjxY6Khy4Ymhk8RWGVFLpuSKoA@mail.gmail.com>
Message-ID: <0E0D5ED7-E97F-4F17-8290-2D893649A8F8@jerris.com>

The alternative if you want in sync audio would be to use webrtc endpoints to listen.

On Aug 14, 2013, at 7:19 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> Icecast is not designed for low-latency and realtime audio.  Its designed for higher quality reliable audio.  The servers and the client libs both use latency and buffering to guarantee this.  There are some techniques to reduce the buffering on at least the FS side but then you will start getting skips and resets if you miss any packets.
> 
> This comes up all the time when people first try doing this.  Either you need to just accept the delay since most people will not even know its there or use some other method.
> 
> 
> 
> 
> On Wed, Aug 14, 2013 at 5:12 PM, Stephen Dame <sdame at 207me.com> wrote:
> Did you restart the freeswitch server after changes to the config files,   I?m not sure 100% if reloadxml resets the conference settings.
> 
>  
> 
> Also I use some flash based players, I think there where client settiings for buffering.   Can you set buffer in HTML tag?
> 
>  
> 
> Regards,
> 
> Stephen
> 
>  
> 
> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame
> 
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jorge N??ez
> Sent: Wednesday, August 14, 2013 4:18 PM
> 
> 
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Mod_shout using with icecast Big delay
> 
>  
> 
> Hi thanks, I set your configuration but nothing changed, I reduced the burst size and it takes me just 11s and also I tried with 8k, 16k and 32k and nothing change
> 
>  
> 
>     <limits>
> 
>         <clients>100</clients>
> 
>         <sources>10</sources>
> 
>         <threadpool>5</threadpool>
> 
>         <queue-size>524288</queue-size>
> 
>         <!-- <queue-size>524288</queue-size> -->
> 
>         <client-timeout>30</client-timeout>
> 
>         <header-timeout>15</header-timeout>
> 
>         <source-timeout>10</source-timeout>
> 
>         <!-- If enabled, this will provide a burst of data when a client 
> 
>              first connects, thereby significantly reducing the startup 
> 
>              time for listeners that do substantial buffering. However,
> 
>              it also significantly increases latency between the source
> 
>              client and listening client.  For low-latency setups, you
> 
>              might want to disable this. -->
> 
>         <burst-on-connect>0</burst-on-connect>
> 
>         <!-- same as burst-on-connect, but this allows for being more
> 
>              specific on how much to burst. Most people won't need to
> 
>              change from the default 64k. Applies to all mountpoints 
> 
>  
> 
>  -->
> 
>         <burst-size>4096</burst-size>
> 
>     </limits>
> 
>  
> 
> Regards
> 
>  
> 
> Jorge
> 
>  
> 
>  
> 
> 2013/8/14 Stephen Dame <sdame at 207me.com>
> 
> Jorge,
> 
>  
> 
> Play around with the burst size and queue size?  here is my xml config around 3-4 second delay from 16k freeswitch conference.
> 
> To small a buffer and the players disconnect? Im also running icecast on same server.
> 
>  
> 
> <icecast>
> 
>     <limits>
> 
>         <clients>100</clients>
> 
>         <sources>10</sources>
> 
>         <threadpool>5</threadpool>
> 
>         <queue-size>524288</queue-size>
> 
>         <client-timeout>30</client-timeout>
> 
>         <header-timeout>15</header-timeout>
> 
>         <source-timeout>10</source-timeout>
> 
>         <burst-on-connect>1</burst-on-connect>
> 
>         <!-- same as burst-on-connect, but this allows for being more
> 
>              specific on how much to burst. Most people won't need to
> 
>              change from the default 64k. Applies to all mountpoints  -->
> 
>         <burst-size>65535</burst-size>
> 
>     </limits>
> 
>  
> 
> Regards,
> 
> Stephen
> 
>  
> 
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
> Sent: Wednesday, August 14, 2013 12:35 PM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Mod_shout using with icecast Big delay
> 
>  
> 
> You have to keep in mind that icecast itself has a fair bit of buffering involved with it and theres not a lot you can do about that
> 
> 
> On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com> wrote:
> 
> Hi I am using mod_shout to send a conference call to icecast and hear it from a tag audio in html5 in realtime, but it has a big delay like 12 seconds. How can I reduce the latency of the audio sent from freeswitch or what can I do to improve this.
> 
> Regards
> 
> Jorge
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
>  
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mike at jerris.com  Thu Aug 15 17:29:24 2013
From: mike at jerris.com (Michael Jerris)
Date: Thu, 15 Aug 2013 09:29:24 -0400
Subject: [Freeswitch-users] Establishing SRTP from SBC to endpoint
In-Reply-To: <CABkS=xqFUnp0D1SFNZh+n8XNUbxTKYACYQ__+tTW49ffL_uPbQ@mail.gmail.com>
References: <CABkS=xqD0mukebja+w3aurrhcUJ4Gp=DcOpkh+BDnCpOKr32CA@mail.gmail.com>
	<CAAYhoXLzsB-o=5VNBaRsR9xdS8bm=j5zKxe8ideg5-jmKU=fKw@mail.gmail.com>
	<CABkS=xrLBX2A1ZLO9gx7mW2yQ2e9LTjyDaNOqJXH2DFADS4KYg@mail.gmail.com>
	<CABkS=xp5XcskU6uJoM-7iHy3taPm3M=QiD29UO9EfToDTR-Fug@mail.gmail.com>
	<CABkS=xqFUnp0D1SFNZh+n8XNUbxTKYACYQ__+tTW49ffL_uPbQ@mail.gmail.com>
Message-ID: <3BF36541-3D31-4460-AF01-5564BB0FD4CC@jerris.com>

I thought the change from sip_ to rtp_ for some variables was only in the 1.4 branch, not in 1.2.9.

On Aug 15, 2013, at 1:57 AM, Peter <eidevm5 at gmail.com> wrote:

> Let me correct my last email.
> 
> If I use rtp_secure_media instead of sip_secure_media, the outgoing call uses RTP and not SRTP.
> 
> rtp_secure_media was supposed to have been introduced in 1.2.9, so I wonder what the difference is?
> 
> 
> On Thu, Aug 15, 2013 at 3:52 PM, Peter <eidevm5 at gmail.com> wrote:
> Finally got it going.  I don't know how many combinations I tried.
> 
> All I needed was the sip_secure_media (or rtp_secure_media, which is the new name) set to true in the dialplan on the SBC.
> 
> 
> On Wed, Aug 14, 2013 at 11:42 AM, Peter <eidevm5 at gmail.com> wrote:
> Hi Carlos.
> 
> Didn't realise rtp_secure_media existed.  After searching I saw:
> 
> https://wiki.freeswitch.org/wiki/Release_Notes#rtp_secure_media_.28was_sip_secure_media.29
> 
> which says it was introduced in 1.2.9
> 
> However, it's a little ambiguous as to whether sip_secure_media was deprecated.
> 
> Anyway, I tried using rtp_secure_media instead, but I still can't get SRTP working.
> 
> 
> I did some testing with some other SIP clients.   In particular, csipsimple.  On the client, if I set SRTP to be optional, the media stream uses RTP.   However, if I set SRTP to be mandatory, when I try to call it, Freeswitch receives:
> 
>    SIP/2.0 488 Not Acceptable Here
> 
> Which seems to indicate that something is not is not right with the SRTP setup.
> 
> There's a full debug from the FS1 (the freeswitch server where the csipsimple client is registered to) at:
> 
> http://pastebin.freeswitch.org/21295
> 
> Note in the debug I have sdp_secure_savp_only set to true.   I've tried disabling this setting, but get the same result.
> 
> Thanks
> 
> Peter
> 
> 
>  
> 
> 
> On Tue, Aug 13, 2013 at 11:06 PM, Carlos Flor <jackal at cybershroud.net> wrote:
> Try using rtp_secure_media=true instead of sip_secure_media.  If you are trying to set it on the b-leg, you probably want to use export instead of set, or use nolocal:rtp_secure_media.
> 
> Hope that helps.
> 
> 
> On Mon, Aug 12, 2013 at 10:26 PM, Peter <eidevm5 at gmail.com> wrote:
> In my environment, I have the following (simplified) setup:
> 
> FS1  ----  FS SBC ---  FS2
> 
> Phones registered to FS1 (100x) use TLS/SRTP and phones registered to FS2 (200x) use SIP/RTP 
> 
> FS1 has inbound-bypass-media set to true to allow SRTP peer to peer and direct to the SBC.
> 
> If I make an inbound call (eg: 1000 to 2000), SRTP is correctly established between the phone and SBC with RTP on the other side of the SBC to the internal phone.
> 
> However, when I try it the other way, I can't get SRTP established from the SBC to the external phone.
> 
> I've been using https://wiki.freeswitch.org/wiki/Secure_RTP as a guide.
> 
> I've even tried explicitly setting sip_secure_media to true on the SBC and FS1.
> 
> The dialplan on the SBC has:
> 
>   <extension name="outgoing">
>         <condition field="destination_number" expression="^(10[0-9][0-9])$">
>             <action application="set" data="sip_secure_media=true"/>
>             <action application="bridge" data="sofia/external/${destination_number}@10.1.1.204"/>
>         </condition>
>   </extension>
> 
> 
> And on FS1, the dialplan has:
> 
>    <extension name="Local-Numbers">
>       <condition field="destination_number" expression="^(10[01][0-9])$">
>         <action application="export" data="dialed_extension=$1"/>
>         <action application="set" data="sip_secure_media=true"/>
>         <action application="bridge" data="user/${dialed_extension}@${domain_name}"/>
>       </condition>
>     </extension>
> 
> 
> Note that I've been testing this against two phones with SRTP enabled, but only one that is using TLS.  I get the same result calling each phone.
> 
> On a related point, what it the step required for a TLS connection from the SBC to the phone?   I'm assume the phone just needs the CA cert from the SBC.  Correct?
> 
> Any information as to where I'm going wrong will be gratefully accepted.
> 
> Thanks
> 
> Peter
>  
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Thu Aug 15 18:40:09 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Aug 2013 09:40:09 -0500
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <520C8D49.3000402@quentustech.com>
References: <20130812053755.96624167@mail.tritonwest.net>
	<520C8D49.3000402@quentustech.com>
Message-ID: <CAKbxfG8X3G4t+swgC2e8k-y+kaYCx_ukBUi4mKK+oDh=snB9LQ@mail.gmail.com>

Didn't Steve say he has an old implementation to dust off earlier in this
thread?
On Aug 15, 2013 3:17 AM, "William King" <william.king at quentustech.com>
wrote:

> Dave,
>
> Know of any current implementations? or would this be entirely new
> functionality?
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
> On 08/11/2013 10:37 PM, Dave R. Kompel wrote:
> > William,
> >
> > The business case is for those of us that use FreeSWITCH as a main
> > switch at a CLEC, where we have our own prefix assignments. In order to
> > do SMS on those numbers we need to interconnect with a SMS backbone
> > provider. SMPP is you're only option for that...
> >
> > --Dave
> >
> >
> ------------------------------------------------------------------------
> >     *From:* William King [mailto:william.king at quentustech.com]
> >     *To:* freeswitch-users at lists.freeswitch.org
> >     *Sent:* Fri, 09 Aug 2013 00:00:33 -0700
> >     *Subject:* Re: [Freeswitch-users] Potential FreeSWITCH support for
> SMPP
> >
> >     >From what I've seen, retail level SMS carriers(not sms aggregators)
> >     usually offer a mix of HTTP, XMPP, SIP SIMPLE, or SMPP interfaces.
> I've
> >     seen the higher volume SMS connections prefer SMPP or HTTP.
> >
> >     I'm exploring to see if there is a business case for FreeSWITCH
> support
> >     for SMPP(either direct support, or interface support through an
> >     independent application) that is not already covered by the current
> FS
> >     feature set.
> >
> >     William King
> >     Senior Engineer
> >     Quentus Technologies, INC
> >     1037 NE 65th St Suite 273
> >     Seattle, WA 98115
> >     Main: (877) 211-9337
> >     Office: (206) 388-4772
> >     Cell: (253) 686-5518
> >     william.king at quentustech.com <mailto:william.king at quentustech.com>
> >
> >     On 08/08/2013 04:23 AM, Raymond Chandler wrote:
> >     > So, to the best of my knowledge, SMPP is strictly for SMS so you
> can
> >     > route SMS to your clients via FS to SIMPLE / dingaling / etc
> clients
> >     >
> >     > -Ray
> >     >
> >     > On Aug 7, 2013 11:05 PM, "Victor Chukalovskiy"
> >     > <victor.chukalovskiy at gmail.com
> >     <mailto:victor.chukalovskiy at gmail.com>
> >     <mailto:victor.chukalovskiy at gmail.com
> >     <mailto:victor.chukalovskiy at gmail.com>>>
> >     > wrote:
> >     >
> >     > So am I as well as another ClueCon attendee I spoke to.
> >     >
> >     > What is the usage scenario you are looking at?
> >     >
> >     > On 13-08-07 06:46 PM, Dmitry Lysenko wrote:
> >     >> Yes. I am interested.
> >     >>
> >     >> Dmitry.
> >     >>
> >     >>
> >     >> 2013/8/8 William King <william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>
> >     >> <mailto:william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>>>
> >     >>
> >     >> Is anyone on this list interested in FreeSWITCH support for
> >     >> SMPP for SMS
> >     >> messages?
> >     >>
> >     >> For more information about the SMPP protocol checkout:
> >     >> http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
> >     >>
> >     >> If so feel free to contact me on or off this list.
> >     >> --
> >     >> William King
> >     >> Senior Engineer
> >     >> Quentus Technologies, INC
> >     >> 1037 NE 65th St Suite 273
> >     >> Seattle, WA 98115
> >     >> Main: (877) 211-9337 <tel:%28877%29%20211-9337>
> >     >> Office: (206) 388-4772 <tel:%28206%29%20388-4772>
> >     >> Cell: (253) 686-5518 <tel:%28253%29%20686-5518>
> >     >> william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>
> >     <mailto:william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>>
> >     >>
> >     >>
> >
> _________________________________________________________________________
> >     >> Professional FreeSWITCH Consulting Services:
> >     >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
> >>
> >     >> http://www.freeswitchsolutions.com
> >     <http://www.freeswitchsolutions.com/>
> >     >>
> >     >> 
> >     >>  </>
> >     >>
> >     >> Official FreeSWITCH Sites
> >     >> http://www.freeswitch.org <http://www.freeswitch.org/>
> >     >> http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
> >     >> http://www.cluecon.com <http://www.cluecon.com/>
> >     >>
> >     >> FreeSWITCH-users mailing list
> >     >> FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >> <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >>
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >> http://www.freeswitch.org <http://www.freeswitch.org/>
> >     >>
> >     >>
> >     >>
> >     >>
> >     >>
> >
> _________________________________________________________________________
> >     >> Professional FreeSWITCH Consulting Services:
> >     >> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
> >>
> >     >> http://www.freeswitchsolutions.com
> >     <http://www.freeswitchsolutions.com/>
> >     >>
> >     >> 
> >     >>  </>
> >     >>
> >     >> Official FreeSWITCH Sites
> >     >> http://www.freeswitch.org <http://www.freeswitch.org/>
> >     >> http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
> >     >> http://www.cluecon.com <http://www.cluecon.com/>
> >     >>
> >     >> FreeSWITCH-users mailing list
> >     >> FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >>
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >> http://www.freeswitch.org <http://www.freeswitch.org/>
> >     >
> >     >
> >     >
> >
> _________________________________________________________________________
> >     > Professional FreeSWITCH Consulting Services:
> >     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
> >>
> >     > http://www.freeswitchsolutions.com
> >     <http://www.freeswitchsolutions.com/>
> >     >
> >     > 
> >     >  </>
> >     >
> >     > Official FreeSWITCH Sites
> >     > http://www.freeswitch.org <http://www.freeswitch.org/>
> >     > http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
> >     > http://www.cluecon.com <http://www.cluecon.com/>
> >     >
> >     > FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     > <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     > http://www.freeswitch.org <http://www.freeswitch.org/>
> >     >
> >     >
> >     >
> >     >
> >
> _________________________________________________________________________
> >     > Professional FreeSWITCH Consulting Services:
> >     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     > http://www.freeswitchsolutions.com
> >     <http://www.freeswitchsolutions.com/>
> >     >
> >     > 
> >     >  </>
> >     >
> >     > Official FreeSWITCH Sites
> >     > http://www.freeswitch.org <http://www.freeswitch.org/>
> >     > http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
> >     > http://www.cluecon.com <http://www.cluecon.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 <http://www.freeswitch.org/>
> >     >
> >
> >
> _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com <
> http://www.freeswitchsolutions.com/>
> >
> >     
> >      </>
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org <http://www.freeswitch.org/>
> >     http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
> >     http://www.cluecon.com <http://www.cluecon.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 <http://www.freeswitch.org/>
> >
> >
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveu at coppice.org  Thu Aug 15 19:14:02 2013
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 15 Aug 2013 23:14:02 +0800
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <CAKbxfG8X3G4t+swgC2e8k-y+kaYCx_ukBUi4mKK+oDh=snB9LQ@mail.gmail.com>
References: <20130812053755.96624167@mail.tritonwest.net>
	<520C8D49.3000402@quentustech.com>
	<CAKbxfG8X3G4t+swgC2e8k-y+kaYCx_ukBUi4mKK+oDh=snB9LQ@mail.gmail.com>
Message-ID: <520CF03A.5070206@coppice.org>

Logica released a Java SMPP platform. I don't know how well that works. 
Like most SMPP stuff, it is long since abandoned. The SMS forum was 
abandoned in something like 2007. I know SMS is dying now, but its 
amazing how much infrastructure tools have been abandoned.

I would need to do some work to release my code, as there is some 
customer specific bits mixed in with it that would need sanitising.

Regards,
Steve

On 08/15/2013 10:40 PM, Anthony Minessale wrote:
>
> Didn't Steve say he has an old implementation to dust off earlier in 
> this thread?
>
> On Aug 15, 2013 3:17 AM, "William King" <william.king at quentustech.com 
> <mailto:william.king at quentustech.com>> wrote:
>
>     Dave,
>
>     Know of any current implementations? or would this be entirely new
>     functionality?
>
>     William King
>     Senior Engineer
>     Quentus Technologies, INC
>     1037 NE 65th St Suite 273
>     Seattle, WA 98115
>     Main: (877) 211-9337 <tel:%28877%29%20211-9337>
>     Office: (206) 388-4772 <tel:%28206%29%20388-4772>
>     Cell: (253) 686-5518 <tel:%28253%29%20686-5518>
>     william.king at quentustech.com <mailto:william.king at quentustech.com>
>
>     On 08/11/2013 10:37 PM, Dave R. Kompel wrote:
>     > William,
>     >
>     > The business case is for those of us that use FreeSWITCH as a main
>     > switch at a CLEC, where we have our own prefix assignments. In
>     order to
>     > do SMS on those numbers we need to interconnect with a SMS backbone
>     > provider. SMPP is you're only option for that...
>     >
>     > --Dave
>     >
>     >
>     ------------------------------------------------------------------------
>     >     *From:* William King [mailto:william.king at quentustech.com
>     <mailto:william.king at quentustech.com>]
>     >     *To:* freeswitch-users at lists.freeswitch.org
>     <mailto:freeswitch-users at lists.freeswitch.org>
>     >     *Sent:* Fri, 09 Aug 2013 00:00:33 -0700
>     >     *Subject:* Re: [Freeswitch-users] Potential FreeSWITCH
>     support for SMPP
>     >
>     >     >From what I've seen, retail level SMS carriers(not sms
>     aggregators)
>     >     usually offer a mix of HTTP, XMPP, SIP SIMPLE, or SMPP
>     interfaces. I've
>     >     seen the higher volume SMS connections prefer SMPP or HTTP.
>     >
>     >     I'm exploring to see if there is a business case for
>     FreeSWITCH support
>     >     for SMPP(either direct support, or interface support through an
>     >     independent application) that is not already covered by the
>     current FS
>     >     feature set.
>     >
>     >     William King
>     >     Senior Engineer
>     >     Quentus Technologies, INC
>     >     1037 NE 65th St Suite 273
>     >     Seattle, WA 98115
>     >     Main: (877) 211-9337
>     >     Office: (206) 388-4772
>     >     Cell: (253) 686-5518
>     > william.king at quentustech.com
>     <mailto:william.king at quentustech.com>
>     <mailto:william.king at quentustech.com
>     <mailto:william.king at quentustech.com>>
>     >
>     >     On 08/08/2013 04:23 AM, Raymond Chandler wrote:
>     >     > So, to the best of my knowledge, SMPP is strictly for SMS
>     so you can
>     >     > route SMS to your clients via FS to SIMPLE / dingaling /
>     etc clients
>     >     >
>     >     > -Ray
>     >     >
>     >     > On Aug 7, 2013 11:05 PM, "Victor Chukalovskiy"
>     >     > <victor.chukalovskiy at gmail.com
>     <mailto:victor.chukalovskiy at gmail.com>
>     >     <mailto:victor.chukalovskiy at gmail.com
>     <mailto:victor.chukalovskiy at gmail.com>>
>     >     <mailto:victor.chukalovskiy at gmail.com
>     <mailto:victor.chukalovskiy at gmail.com>
>     >     <mailto:victor.chukalovskiy at gmail.com
>     <mailto:victor.chukalovskiy at gmail.com>>>>
>     >     > wrote:
>     >     >
>     >     > So am I as well as another ClueCon attendee I spoke to.
>     >     >
>     >     > What is the usage scenario you are looking at?
>     >     >
>     >     > On 13-08-07 06:46 PM, Dmitry Lysenko wrote:
>     >     >> Yes. I am interested.
>     >     >>
>     >     >> Dmitry.
>     >     >>
>     >     >>
>     >     >> 2013/8/8 William King <william.king at quentustech.com
>     <mailto:william.king at quentustech.com>
>     >     <mailto:william.king at quentustech.com
>     <mailto:william.king at quentustech.com>>
>     >     >> <mailto:william.king at quentustech.com
>     <mailto:william.king at quentustech.com>
>     >     <mailto:william.king at quentustech.com
>     <mailto:william.king at quentustech.com>>>>
>     >     >>
>     >     >> Is anyone on this list interested in FreeSWITCH support for
>     >     >> SMPP for SMS
>     >     >> messages?
>     >     >>
>     >     >> For more information about the SMPP protocol checkout:
>     >     >> http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
>     >     >>
>     >     >> If so feel free to contact me on or off this list.
>     >     >> --
>     >     >> William King
>     >     >> Senior Engineer
>     >     >> Quentus Technologies, INC
>     >     >> 1037 NE 65th St Suite 273
>     >     >> Seattle, WA 98115
>     >     >> Main: (877) 211-9337 <tel:%28877%29%20211-9337>
>     <tel:%28877%29%20211-9337>
>     >     >> Office: (206) 388-4772 <tel:%28206%29%20388-4772>
>     <tel:%28206%29%20388-4772>
>     >     >> Cell: (253) 686-5518 <tel:%28253%29%20686-5518>
>     >     >> william.king at quentustech.com
>     <mailto:william.king at quentustech.com>
>     >     <mailto:william.king at quentustech.com
>     <mailto:william.king at quentustech.com>>
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>     <mailto:william.king at quentustech.com>
>     >     <mailto:william.king at quentustech.com
>     <mailto:william.king at quentustech.com>>>
>     >     >>
>     >     >>
>     >
>     _________________________________________________________________________
>     >     >> Professional FreeSWITCH Consulting Services:
>     >     >> consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     >     <mailto:consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>>
>     >     >> http://www.freeswitchsolutions.com
>     >     <http://www.freeswitchsolutions.com/>
>     >     >>
>     >     >> 
>     >     >>  </>
>     >     >>
>     >     >> Official FreeSWITCH Sites
>     >     >> http://www.freeswitch.org <http://www.freeswitch.org/>
>     >     >> http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
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>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
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>     >     >>
>     >    
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>     >     >>
>     >     >>
>     >     >>
>     >     >>
>     >     >>
>     >
>     _________________________________________________________________________
>     >     >> Professional FreeSWITCH Consulting Services:
>     >     >> consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     >     <mailto:consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>>
>     >     >> http://www.freeswitchsolutions.com
>     >     <http://www.freeswitchsolutions.com/>
>     >     >>
>     >     >> 
>     >     >>  </>
>     >     >>
>     >     >> Official FreeSWITCH Sites
>     >     >> http://www.freeswitch.org <http://www.freeswitch.org/>
>     >     >> http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>     >     >> http://www.cluecon.com <http://www.cluecon.com/>
>     >     >>
>     >     >> FreeSWITCH-users mailing list
>     >     >> FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
>     >     >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >     >>
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>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>     >     >
>     >     >
>     >     >
>     >
>     _________________________________________________________________________
>     >     > Professional FreeSWITCH Consulting Services:
>     >     > consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     >     <mailto:consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>>
>     >     > http://www.freeswitchsolutions.com
>     >     <http://www.freeswitchsolutions.com/>
>     >     >
>     >     > 
>     >     >  </>
>     >     >
>     >     > Official FreeSWITCH Sites
>     >     > http://www.freeswitch.org <http://www.freeswitch.org/>
>     >     > http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>     >     > http://www.cluecon.com <http://www.cluecon.com/>
>     >     >
>     >     > FreeSWITCH-users mailing list
>     >     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>     >     > <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
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>     >     >
>     >     >
>     >     >
>     >
>     _________________________________________________________________________
>     >     > Professional FreeSWITCH Consulting Services:
>     >     > consulting at freeswitch.org
>     <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     >     > http://www.freeswitchsolutions.com
>     >     <http://www.freeswitchsolutions.com/>
>     >     >
>     >     > 
>     >     >  </>
>     >     >
>     >     > Official FreeSWITCH Sites
>     >     > http://www.freeswitch.org <http://www.freeswitch.org/>
>     >     > http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>     >     > http://www.cluecon.com <http://www.cluecon.com/>
>     >     >
>     >     > FreeSWITCH-users mailing list
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>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
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>     >
>     >
>     _________________________________________________________________________
>     >     Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org>>
>     > http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com/>
>     >
>     >     
>     >  </>
>     >
>     >     Official FreeSWITCH Sites
>     > http://www.freeswitch.org <http://www.freeswitch.org/>
>     > http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>     > http://www.cluecon.com <http://www.cluecon.com/>
>     >
>     >     FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
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>     > http://www.freeswitch.org <http://www.freeswitch.org/>
>     >
>     >
>     >
>     >
>     >
>     >
>     _________________________________________________________________________
>     > Professional FreeSWITCH Consulting Services:
>     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     > http://www.freeswitchsolutions.com
>     >
>     > 
>     > 
>     >
>     > Official FreeSWITCH Sites
>     > http://www.freeswitch.org
>     > http://wiki.freeswitch.org
>     > http://www.cluecon.com
>     >
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>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
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>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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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From mishehu at freeswitch.org  Thu Aug 15 20:23:12 2013
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Thu, 15 Aug 2013 11:23:12 -0500
Subject: [Freeswitch-users] Auto-detecting Local IP when no gateway set
In-Reply-To: <CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
Message-ID: <520D0070.2010008@freeswitch.org>

I'm not sure this will work since Siri's customer appears to already use 
multiple sip profiles on the system.  I get the nagging feeling that a 
technical problem is trying to solve a social problem here:  maybe the 
easy solution is for Siri to tell the customer that if they change the 
IP addresses on the server that they are equally responsible for 
updating the FreeSWITCH configs.

-Yossi

On 08/15/2013 07:14 AM, Raymond Chandler wrote:
>
> Alternatively, you could use xml_curl to serve your configs so that 
> you don't actually need local_ip_v4 as you can figure it out based on 
> the IP hitting the web server. Doing that will allow you to set the IP 
> in all of the configs that need it using functionality that already 
> exists in FreeSWITCH today.
>
> -Ray
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jorgefren12 at gmail.com  Thu Aug 15 21:07:35 2013
From: jorgefren12 at gmail.com (=?ISO-8859-1?Q?Jorge_N=FA=F1ez?=)
Date: Thu, 15 Aug 2013 12:07:35 -0500
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <0E0D5ED7-E97F-4F17-8290-2D893649A8F8@jerris.com>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>
	<CE311BFC.78EA0%krice@freeswitch.org>
	<045801ce990e$205b4f90$6111eeb0$@207me.com>
	<CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>
	<049101ce9915$e1cbb5f0$a56321d0$@207me.com>
	<CAOP4xcy6NZG0JKPMYhJp_wxaLqnAjsEkGMgTD5itvBPNdSCPzQ@mail.gmail.com>
	<050301ce993b$4dc37b60$e94a7220$@207me.com>
	<CAKbxfG80vr3b+BULjn5ZhKW6ZjxY6Khy4Ymhk8RWGVFLpuSKoA@mail.gmail.com>
	<0E0D5ED7-E97F-4F17-8290-2D893649A8F8@jerris.com>
Message-ID: <CAOP4xcwBNPxaxsrFBr48-02QriO_baDv-=o+QhoDsiMKUnq4ug@mail.gmail.com>

Anthony and just for test how can I reduce the latency? and thanks Michael
for use webrtc Do I need to use Sipml5 or JSSIP o are there any othe way to
use because I was trying to find some information about mod_html5 but I
didn?t find

thanks

Jorge


2013/8/15 Michael Jerris <mike at jerris.com>

> The alternative if you want in sync audio would be to use webrtc endpoints
> to listen.
>
> On Aug 14, 2013, at 7:19 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> Icecast is not designed for low-latency and realtime audio.  Its designed
> for higher quality reliable audio.  The servers and the client libs both
> use latency and buffering to guarantee this.  There are some techniques to
> reduce the buffering on at least the FS side but then you will start
> getting skips and resets if you miss any packets.
>
> This comes up all the time when people first try doing this.  Either you
> need to just accept the delay since most people will not even know its
> there or use some other method.
>
>
>
>
> On Wed, Aug 14, 2013 at 5:12 PM, Stephen Dame <sdame at 207me.com> wrote:
>
>> Did you restart the freeswitch server after changes to the config
>> files,   I?m not sure 100% if reloadxml resets the conference settings. *
>> ***
>>
>> ** **
>>
>> Also I use some flash based players, I think there where client settiings
>> for buffering.   Can you set buffer in HTML tag?****
>>
>> ** **
>>
>> Regards,****
>>
>> Stephen****
>>
>> ** **
>>
>> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jorge N??ez
>> *Sent:* Wednesday, August 14, 2013 4:18 PM
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big delay*
>> ***
>>
>> ** **
>>
>> Hi thanks, I set your configuration but nothing changed, I reduced the
>> burst size and it takes me just 11s and also I tried with 8k, 16k and 32k
>> and nothing change****
>>
>> ** **
>>
>>     <limits>****
>>
>>         <clients>100</clients>****
>>
>>         <sources>10</sources>****
>>
>>         <threadpool>5</threadpool>****
>>
>>         <queue-size>524288</queue-size>****
>>
>>         <!-- <queue-size>524288</queue-size> -->****
>>
>>         <client-timeout>30</client-timeout>****
>>
>>         <header-timeout>15</header-timeout>****
>>
>>         <source-timeout>10</source-timeout>****
>>
>>         <!-- If enabled, this will provide a burst of data when a client
>> ****
>>
>>              first connects, thereby significantly reducing the startup *
>> ***
>>
>>              time for listeners that do substantial buffering. However,**
>> **
>>
>>              it also significantly increases latency between the source**
>> **
>>
>>              client and listening client.  For low-latency setups, you***
>> *
>>
>>              might want to disable this. -->****
>>
>>         <burst-on-connect>0</burst-on-connect>****
>>
>>         <!-- same as burst-on-connect, but this allows for being more****
>>
>>              specific on how much to burst. Most people won't need to****
>>
>>              change from the default 64k. Applies to all mountpoints ****
>>
>> ** **
>>
>>  -->****
>>
>>         <burst-size>4096</burst-size>****
>>
>>     </limits>****
>>
>> ** **
>>
>> Regards****
>>
>> ** **
>>
>> Jorge****
>>
>> ** **
>>
>> ** **
>>
>> 2013/8/14 Stephen Dame <sdame at 207me.com>****
>>
>> Jorge, ****
>>
>>  ****
>>
>> Play around with the burst size and queue size?  here is my xml config
>> around 3-4 second delay from 16k freeswitch conference. ****
>>
>> To small a buffer and the players disconnect? Im also running icecast on
>> same server.****
>>
>>  ****
>>
>> <icecast>****
>>
>>     <limits>****
>>
>>         <clients>100</clients>****
>>
>>         <sources>10</sources>****
>>
>>         <threadpool>5</threadpool>****
>>
>>         <queue-size>524288</queue-size>****
>>
>>         <client-timeout>30</client-timeout>****
>>
>>         <header-timeout>15</header-timeout>****
>>
>>         <source-timeout>10</source-timeout>****
>>
>>         <burst-on-connect>1</burst-on-connect>****
>>
>>         <!-- same as burst-on-connect, but this allows for being more****
>>
>>              specific on how much to burst. Most people won't need to****
>>
>>              change from the default 64k. Applies to all mountpoints  -->
>> ****
>>
>>         <burst-size>65535</burst-size>****
>>
>>     </limits>****
>>
>>  ****
>>
>> Regards,****
>>
>> Stephen****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice
>> *Sent:* Wednesday, August 14, 2013 12:35 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big delay*
>> ***
>>
>>  ****
>>
>> You have to keep in mind that icecast itself has a fair bit of buffering
>> involved with it and theres not a lot you can do about that
>>
>>
>> On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com> wrote:****
>>
>> Hi I am using mod_shout to send a conference call to icecast and hear it
>> from a tag audio in html5 in realtime, but it has a big delay like 12
>> seconds. How can I reduce the latency of the audio sent from freeswitch or
>> what can I do to improve this.
>>
>> Regards
>>
>> Jorge****
>> ------------------------------
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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****
>>
>>
>> --
>> Ken
>> *http://www.FreeSWITCH.org <http://www.freeswitch.org/>
>> http://www.ClueCon.com <http://www.cluecon.com/>
>> http://www.OSTAG.org <http://www.ostag.org/>
>> *irc.freenode.net #freeswitch****
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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****
>>
>> ** **
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
-------------- next part --------------
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From anthony.minessale at gmail.com  Thu Aug 15 21:34:55 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Aug 2013 12:34:55 -0500
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <CAOP4xcwBNPxaxsrFBr48-02QriO_baDv-=o+QhoDsiMKUnq4ug@mail.gmail.com>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>
	<CE311BFC.78EA0%krice@freeswitch.org>
	<045801ce990e$205b4f90$6111eeb0$@207me.com>
	<CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>
	<049101ce9915$e1cbb5f0$a56321d0$@207me.com>
	<CAOP4xcy6NZG0JKPMYhJp_wxaLqnAjsEkGMgTD5itvBPNdSCPzQ@mail.gmail.com>
	<050301ce993b$4dc37b60$e94a7220$@207me.com>
	<CAKbxfG80vr3b+BULjn5ZhKW6ZjxY6Khy4Ymhk8RWGVFLpuSKoA@mail.gmail.com>
	<0E0D5ED7-E97F-4F17-8290-2D893649A8F8@jerris.com>
	<CAOP4xcwBNPxaxsrFBr48-02QriO_baDv-=o+QhoDsiMKUnq4ug@mail.gmail.com>
Message-ID: <CAKbxfG8zu0+_AM4zzmAq01BFHxpLygb8-2JdXNujOw6X6UMesw@mail.gmail.com>

enable_file_write_buffering=false
stream_prebuffer=0

On the leg doing the recording.  It does't stop the iceast server or
libshout from buffering it more.





On Thu, Aug 15, 2013 at 12:07 PM, Jorge N??ez <jorgefren12 at gmail.com> wrote:

> Anthony and just for test how can I reduce the latency? and thanks Michael
> for use webrtc Do I need to use Sipml5 or JSSIP o are there any othe way to
> use because I was trying to find some information about mod_html5 but I
> didn?t find
>
> thanks
>
> Jorge
>
>
> 2013/8/15 Michael Jerris <mike at jerris.com>
>
>> The alternative if you want in sync audio would be to use webrtc
>> endpoints to listen.
>>
>> On Aug 14, 2013, at 7:19 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>> Icecast is not designed for low-latency and realtime audio.  Its designed
>> for higher quality reliable audio.  The servers and the client libs both
>> use latency and buffering to guarantee this.  There are some techniques to
>> reduce the buffering on at least the FS side but then you will start
>> getting skips and resets if you miss any packets.
>>
>> This comes up all the time when people first try doing this.  Either you
>> need to just accept the delay since most people will not even know its
>> there or use some other method.
>>
>>
>>
>>
>> On Wed, Aug 14, 2013 at 5:12 PM, Stephen Dame <sdame at 207me.com> wrote:
>>
>>> Did you restart the freeswitch server after changes to the config
>>> files,   I?m not sure 100% if reloadxml resets the conference settings.
>>> ****
>>>
>>> ** **
>>>
>>> Also I use some flash based players, I think there where client
>>> settiings for buffering.   Can you set buffer in HTML tag?****
>>>
>>> ** **
>>>
>>> Regards,****
>>>
>>> Stephen****
>>>
>>> ** **
>>>
>>> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>>>
>>> ** **
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jorge
>>> N??ez
>>> *Sent:* Wednesday, August 14, 2013 4:18 PM
>>>
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big delay
>>> ****
>>>
>>> ** **
>>>
>>> Hi thanks, I set your configuration but nothing changed, I reduced the
>>> burst size and it takes me just 11s and also I tried with 8k, 16k and 32k
>>> and nothing change****
>>>
>>> ** **
>>>
>>>     <limits>****
>>>
>>>         <clients>100</clients>****
>>>
>>>         <sources>10</sources>****
>>>
>>>         <threadpool>5</threadpool>****
>>>
>>>         <queue-size>524288</queue-size>****
>>>
>>>         <!-- <queue-size>524288</queue-size> -->****
>>>
>>>         <client-timeout>30</client-timeout>****
>>>
>>>         <header-timeout>15</header-timeout>****
>>>
>>>         <source-timeout>10</source-timeout>****
>>>
>>>         <!-- If enabled, this will provide a burst of data when a client
>>> ****
>>>
>>>              first connects, thereby significantly reducing the startup
>>> ****
>>>
>>>              time for listeners that do substantial buffering. However,*
>>> ***
>>>
>>>              it also significantly increases latency between the source*
>>> ***
>>>
>>>              client and listening client.  For low-latency setups, you**
>>> **
>>>
>>>              might want to disable this. -->****
>>>
>>>         <burst-on-connect>0</burst-on-connect>****
>>>
>>>         <!-- same as burst-on-connect, but this allows for being more***
>>> *
>>>
>>>              specific on how much to burst. Most people won't need to***
>>> *
>>>
>>>              change from the default 64k. Applies to all mountpoints ***
>>> *
>>>
>>> ** **
>>>
>>>  -->****
>>>
>>>         <burst-size>4096</burst-size>****
>>>
>>>     </limits>****
>>>
>>> ** **
>>>
>>> Regards****
>>>
>>> ** **
>>>
>>> Jorge****
>>>
>>> ** **
>>>
>>> ** **
>>>
>>> 2013/8/14 Stephen Dame <sdame at 207me.com>****
>>>
>>> Jorge, ****
>>>
>>>  ****
>>>
>>> Play around with the burst size and queue size?  here is my xml config
>>> around 3-4 second delay from 16k freeswitch conference. ****
>>>
>>> To small a buffer and the players disconnect? Im also running icecast on
>>> same server.****
>>>
>>>  ****
>>>
>>> <icecast>****
>>>
>>>     <limits>****
>>>
>>>         <clients>100</clients>****
>>>
>>>         <sources>10</sources>****
>>>
>>>         <threadpool>5</threadpool>****
>>>
>>>         <queue-size>524288</queue-size>****
>>>
>>>         <client-timeout>30</client-timeout>****
>>>
>>>         <header-timeout>15</header-timeout>****
>>>
>>>         <source-timeout>10</source-timeout>****
>>>
>>>         <burst-on-connect>1</burst-on-connect>****
>>>
>>>         <!-- same as burst-on-connect, but this allows for being more***
>>> *
>>>
>>>              specific on how much to burst. Most people won't need to***
>>> *
>>>
>>>              change from the default 64k. Applies to all mountpoints  -->
>>> ****
>>>
>>>         <burst-size>65535</burst-size>****
>>>
>>>     </limits>****
>>>
>>>  ****
>>>
>>> Regards,****
>>>
>>> Stephen****
>>>
>>>  ****
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice
>>> *Sent:* Wednesday, August 14, 2013 12:35 PM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big delay
>>> ****
>>>
>>>  ****
>>>
>>> You have to keep in mind that icecast itself has a fair bit of buffering
>>> involved with it and theres not a lot you can do about that
>>>
>>>
>>> On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com> wrote:****
>>>
>>> Hi I am using mod_shout to send a conference call to icecast and hear it
>>> from a tag audio in html5 in realtime, but it has a big delay like 12
>>> seconds. How can I reduce the latency of the audio sent from freeswitch or
>>> what can I do to improve this.
>>>
>>> Regards
>>>
>>> Jorge****
>>> ------------------------------
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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****
>>>
>>>
>>> --
>>> Ken
>>> *http://www.FreeSWITCH.org <http://www.freeswitch.org/>
>>> http://www.ClueCon.com <http://www.cluecon.com/>
>>> http://www.OSTAG.org <http://www.ostag.org/>
>>> *irc.freenode.net #freeswitch****
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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****
>>>
>>> ** **
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
-------------- next part --------------
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From jleung at v10networks.ca  Thu Aug 15 21:40:15 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Thu, 15 Aug 2013 10:40:15 -0700
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <CAKbxfG8zu0+_AM4zzmAq01BFHxpLygb8-2JdXNujOw6X6UMesw@mail.gmail.com>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>
	<CE311BFC.78EA0%krice@freeswitch.org>
	<045801ce990e$205b4f90$6111eeb0$@207me.com>
	<CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>
	<049101ce9915$e1cbb5f0$a56321d0$@207me.com>
	<CAOP4xcy6NZG0JKPMYhJp_wxaLqnAjsEkGMgTD5itvBPNdSCPzQ@mail.gmail.com>
	<050301ce993b$4dc37b60$e94a7220$@207me.com>
	<CAKbxfG80vr3b+BULjn5ZhKW6ZjxY6Khy4Ymhk8RWGVFLpuSKoA@mail.gmail.com>
	<0E0D5ED7-E97F-4F17-8290-2D893649A8F8@jerris.com>
	<CAOP4xcwBNPxaxsrFBr48-02QriO_baDv-=o+QhoDsiMKUnq4ug@mail.gmail.com>
	<CAKbxfG8zu0+_AM4zzmAq01BFHxpLygb8-2JdXNujOw6X6UMesw@mail.gmail.com>
Message-ID: <CADk4Sr1g2c3toiU7+ikM_WByfUzvxwXKQwLrFXF8KnrtZoXWnQ@mail.gmail.com>

Icecast is known to have a relatively high latency for audio.


On Thu, Aug 15, 2013 at 10:34 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> enable_file_write_buffering=false
> stream_prebuffer=0
>
> On the leg doing the recording.  It does't stop the iceast server or
> libshout from buffering it more.
>
>
>
>
>
> On Thu, Aug 15, 2013 at 12:07 PM, Jorge N??ez <jorgefren12 at gmail.com>wrote:
>
>> Anthony and just for test how can I reduce the latency? and thanks
>> Michael for use webrtc Do I need to use Sipml5 or JSSIP o are there any
>> othe way to use because I was trying to find some information about
>> mod_html5 but I didn?t find
>>
>> thanks
>>
>> Jorge
>>
>>
>> 2013/8/15 Michael Jerris <mike at jerris.com>
>>
>>> The alternative if you want in sync audio would be to use webrtc
>>> endpoints to listen.
>>>
>>> On Aug 14, 2013, at 7:19 PM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>> Icecast is not designed for low-latency and realtime audio.  Its
>>> designed for higher quality reliable audio.  The servers and the client
>>> libs both use latency and buffering to guarantee this.  There are some
>>> techniques to reduce the buffering on at least the FS side but then you
>>> will start getting skips and resets if you miss any packets.
>>>
>>> This comes up all the time when people first try doing this.  Either you
>>> need to just accept the delay since most people will not even know its
>>> there or use some other method.
>>>
>>>
>>>
>>>
>>> On Wed, Aug 14, 2013 at 5:12 PM, Stephen Dame <sdame at 207me.com> wrote:
>>>
>>>> Did you restart the freeswitch server after changes to the config
>>>> files,   I?m not sure 100% if reloadxml resets the conference settings.
>>>> ****
>>>>
>>>> ** **
>>>>
>>>> Also I use some flash based players, I think there where client
>>>> settiings for buffering.   Can you set buffer in HTML tag?****
>>>>
>>>> ** **
>>>>
>>>> Regards,****
>>>>
>>>> Stephen****
>>>>
>>>> ** **
>>>>
>>>> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>>>>
>>>> ** **
>>>>
>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jorge
>>>> N??ez
>>>> *Sent:* Wednesday, August 14, 2013 4:18 PM
>>>>
>>>> *To:* FreeSWITCH Users Help
>>>> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big
>>>> delay****
>>>>
>>>> ** **
>>>>
>>>> Hi thanks, I set your configuration but nothing changed, I reduced the
>>>> burst size and it takes me just 11s and also I tried with 8k, 16k and 32k
>>>> and nothing change****
>>>>
>>>> ** **
>>>>
>>>>     <limits>****
>>>>
>>>>         <clients>100</clients>****
>>>>
>>>>         <sources>10</sources>****
>>>>
>>>>         <threadpool>5</threadpool>****
>>>>
>>>>         <queue-size>524288</queue-size>****
>>>>
>>>>         <!-- <queue-size>524288</queue-size> -->****
>>>>
>>>>         <client-timeout>30</client-timeout>****
>>>>
>>>>         <header-timeout>15</header-timeout>****
>>>>
>>>>         <source-timeout>10</source-timeout>****
>>>>
>>>>         <!-- If enabled, this will provide a burst of data when a
>>>> client ****
>>>>
>>>>              first connects, thereby significantly reducing the startup
>>>> ****
>>>>
>>>>              time for listeners that do substantial buffering. However,
>>>> ****
>>>>
>>>>              it also significantly increases latency between the source
>>>> ****
>>>>
>>>>              client and listening client.  For low-latency setups, you*
>>>> ***
>>>>
>>>>              might want to disable this. -->****
>>>>
>>>>         <burst-on-connect>0</burst-on-connect>****
>>>>
>>>>         <!-- same as burst-on-connect, but this allows for being more**
>>>> **
>>>>
>>>>              specific on how much to burst. Most people won't need to**
>>>> **
>>>>
>>>>              change from the default 64k. Applies to all mountpoints **
>>>> **
>>>>
>>>> ** **
>>>>
>>>>  -->****
>>>>
>>>>         <burst-size>4096</burst-size>****
>>>>
>>>>     </limits>****
>>>>
>>>> ** **
>>>>
>>>> Regards****
>>>>
>>>> ** **
>>>>
>>>> Jorge****
>>>>
>>>> ** **
>>>>
>>>> ** **
>>>>
>>>> 2013/8/14 Stephen Dame <sdame at 207me.com>****
>>>>
>>>> Jorge, ****
>>>>
>>>>  ****
>>>>
>>>> Play around with the burst size and queue size?  here is my xml config
>>>> around 3-4 second delay from 16k freeswitch conference. ****
>>>>
>>>> To small a buffer and the players disconnect? Im also running icecast
>>>> on same server.****
>>>>
>>>>  ****
>>>>
>>>> <icecast>****
>>>>
>>>>     <limits>****
>>>>
>>>>         <clients>100</clients>****
>>>>
>>>>         <sources>10</sources>****
>>>>
>>>>         <threadpool>5</threadpool>****
>>>>
>>>>         <queue-size>524288</queue-size>****
>>>>
>>>>         <client-timeout>30</client-timeout>****
>>>>
>>>>         <header-timeout>15</header-timeout>****
>>>>
>>>>         <source-timeout>10</source-timeout>****
>>>>
>>>>         <burst-on-connect>1</burst-on-connect>****
>>>>
>>>>         <!-- same as burst-on-connect, but this allows for being more**
>>>> **
>>>>
>>>>              specific on how much to burst. Most people won't need to**
>>>> **
>>>>
>>>>              change from the default 64k. Applies to all mountpoints
>>>> -->****
>>>>
>>>>         <burst-size>65535</burst-size>****
>>>>
>>>>     </limits>****
>>>>
>>>>  ****
>>>>
>>>> Regards,****
>>>>
>>>> Stephen****
>>>>
>>>>  ****
>>>>
>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice
>>>> *Sent:* Wednesday, August 14, 2013 12:35 PM
>>>> *To:* FreeSWITCH Users Help
>>>> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big
>>>> delay****
>>>>
>>>>  ****
>>>>
>>>> You have to keep in mind that icecast itself has a fair bit of
>>>> buffering involved with it and theres not a lot you can do about that
>>>>
>>>>
>>>> On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com> wrote:****
>>>>
>>>> Hi I am using mod_shout to send a conference call to icecast and hear
>>>> it from a tag audio in html5 in realtime, but it has a big delay like 12
>>>> seconds. How can I reduce the latency of the audio sent from freeswitch or
>>>> what can I do to improve this.
>>>>
>>>> Regards
>>>>
>>>> Jorge****
>>>> ------------------------------
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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****
>>>>
>>>>
>>>> --
>>>> Ken
>>>> *http://www.FreeSWITCH.org <http://www.freeswitch.org/>
>>>> http://www.ClueCon.com <http://www.cluecon.com/>
>>>> http://www.OSTAG.org <http://www.ostag.org/>
>>>> *irc.freenode.net #freeswitch****
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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****
>>>>
>>>> ** **
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Thu Aug 15 22:39:46 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Aug 2013 13:39:46 -0500
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <520CF03A.5070206@coppice.org>
References: <20130812053755.96624167@mail.tritonwest.net>
	<520C8D49.3000402@quentustech.com>
	<CAKbxfG8X3G4t+swgC2e8k-y+kaYCx_ukBUi4mKK+oDh=snB9LQ@mail.gmail.com>
	<520CF03A.5070206@coppice.org>
Message-ID: <CAKbxfG99ZWAG=gmoTxG+ccE6WBGCuWXt9g1+U-QVKtCjcWZ93w@mail.gmail.com>

It seems to be dying less when it comes to enabling it to SIP or XMPP.
 I've watched the topic rise for the last few years and we seem on the
verge of many SMS enabled SIP accounts from ITSP.  I think it will have a
reprise when people start developing SMS based apps that can use a platform
like FS to make the app design easier.

I'm sure before too long when everyone has LTE, we will just learn to send
data messages to each other over IP but the legacy stuff always seems to
hang around doesn't it....


On Thu, Aug 15, 2013 at 10:14 AM, Steve Underwood <steveu at coppice.org>wrote:

> Logica released a Java SMPP platform. I don't know how well that works.
> Like most SMPP stuff, it is long since abandoned. The SMS forum was
> abandoned in something like 2007. I know SMS is dying now, but its
> amazing how much infrastructure tools have been abandoned.
>
> I would need to do some work to release my code, as there is some
> customer specific bits mixed in with it that would need sanitising.
>
> Regards,
> Steve
>
> On 08/15/2013 10:40 PM, Anthony Minessale wrote:
> >
> > Didn't Steve say he has an old implementation to dust off earlier in
> > this thread?
> >
> > On Aug 15, 2013 3:17 AM, "William King" <william.king at quentustech.com
> > <mailto:william.king at quentustech.com>> wrote:
> >
> >     Dave,
> >
> >     Know of any current implementations? or would this be entirely new
> >     functionality?
> >
> >     William King
> >     Senior Engineer
> >     Quentus Technologies, INC
> >     1037 NE 65th St Suite 273
> >     Seattle, WA 98115
> >     Main: (877) 211-9337 <tel:%28877%29%20211-9337>
> >     Office: (206) 388-4772 <tel:%28206%29%20388-4772>
> >     Cell: (253) 686-5518 <tel:%28253%29%20686-5518>
> >     william.king at quentustech.com <mailto:william.king at quentustech.com>
> >
> >     On 08/11/2013 10:37 PM, Dave R. Kompel wrote:
> >     > William,
> >     >
> >     > The business case is for those of us that use FreeSWITCH as a main
> >     > switch at a CLEC, where we have our own prefix assignments. In
> >     order to
> >     > do SMS on those numbers we need to interconnect with a SMS backbone
> >     > provider. SMPP is you're only option for that...
> >     >
> >     > --Dave
> >     >
> >     >
> >
> ------------------------------------------------------------------------
> >     >     *From:* William King [mailto:william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>]
> >     >     *To:* freeswitch-users at lists.freeswitch.org
> >     <mailto:freeswitch-users at lists.freeswitch.org>
> >     >     *Sent:* Fri, 09 Aug 2013 00:00:33 -0700
> >     >     *Subject:* Re: [Freeswitch-users] Potential FreeSWITCH
> >     support for SMPP
> >     >
> >     >     >From what I've seen, retail level SMS carriers(not sms
> >     aggregators)
> >     >     usually offer a mix of HTTP, XMPP, SIP SIMPLE, or SMPP
> >     interfaces. I've
> >     >     seen the higher volume SMS connections prefer SMPP or HTTP.
> >     >
> >     >     I'm exploring to see if there is a business case for
> >     FreeSWITCH support
> >     >     for SMPP(either direct support, or interface support through an
> >     >     independent application) that is not already covered by the
> >     current FS
> >     >     feature set.
> >     >
> >     >     William King
> >     >     Senior Engineer
> >     >     Quentus Technologies, INC
> >     >     1037 NE 65th St Suite 273
> >     >     Seattle, WA 98115
> >     >     Main: (877) 211-9337
> >     >     Office: (206) 388-4772
> >     >     Cell: (253) 686-5518
> >     > william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>
> >     <mailto:william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>>
> >     >
> >     >     On 08/08/2013 04:23 AM, Raymond Chandler wrote:
> >     >     > So, to the best of my knowledge, SMPP is strictly for SMS
> >     so you can
> >     >     > route SMS to your clients via FS to SIMPLE / dingaling /
> >     etc clients
> >     >     >
> >     >     > -Ray
> >     >     >
> >     >     > On Aug 7, 2013 11:05 PM, "Victor Chukalovskiy"
> >     >     > <victor.chukalovskiy at gmail.com
> >     <mailto:victor.chukalovskiy at gmail.com>
> >     >     <mailto:victor.chukalovskiy at gmail.com
> >     <mailto:victor.chukalovskiy at gmail.com>>
> >     >     <mailto:victor.chukalovskiy at gmail.com
> >     <mailto:victor.chukalovskiy at gmail.com>
> >     >     <mailto:victor.chukalovskiy at gmail.com
> >     <mailto:victor.chukalovskiy at gmail.com>>>>
> >     >     > wrote:
> >     >     >
> >     >     > So am I as well as another ClueCon attendee I spoke to.
> >     >     >
> >     >     > What is the usage scenario you are looking at?
> >     >     >
> >     >     > On 13-08-07 06:46 PM, Dmitry Lysenko wrote:
> >     >     >> Yes. I am interested.
> >     >     >>
> >     >     >> Dmitry.
> >     >     >>
> >     >     >>
> >     >     >> 2013/8/8 William King <william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>
> >     >     <mailto:william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>>
> >     >     >> <mailto:william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>
> >     >     <mailto:william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>>>>
> >     >     >>
> >     >     >> Is anyone on this list interested in FreeSWITCH support for
> >     >     >> SMPP for SMS
> >     >     >> messages?
> >     >     >>
> >     >     >> For more information about the SMPP protocol checkout:
> >     >     >> http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
> >     >     >>
> >     >     >> If so feel free to contact me on or off this list.
> >     >     >> --
> >     >     >> William King
> >     >     >> Senior Engineer
> >     >     >> Quentus Technologies, INC
> >     >     >> 1037 NE 65th St Suite 273
> >     >     >> Seattle, WA 98115
> >     >     >> Main: (877) 211-9337 <tel:%28877%29%20211-9337>
> >     <tel:%28877%29%20211-9337>
> >     >     >> Office: (206) 388-4772 <tel:%28206%29%20388-4772>
> >     <tel:%28206%29%20388-4772>
> >     >     >> Cell: (253) 686-5518 <tel:%28253%29%20686-5518>
> >     >     >> william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>
> >     >     <mailto:william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>>
> >     >     <mailto:william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>
> >     >     <mailto:william.king at quentustech.com
> >     <mailto:william.king at quentustech.com>>>
> >     >     >>
> >     >     >>
> >     >
> >
> _________________________________________________________________________
> >     >     >> Professional FreeSWITCH Consulting Services:
> >     >     >> consulting at freeswitch.org
> >     <mailto:consulting at freeswitch.org>
> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
> >>
> >     >     <mailto:consulting at freeswitch.org
> >     <mailto:consulting at freeswitch.org>
> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
> >>>
> >     >     >> http://www.freeswitchsolutions.com
> >     >     <http://www.freeswitchsolutions.com/>
> >     >     >>
> >     >     >> 
> >     >     >>  </>
> >     >     >>
> >     >     >> Official FreeSWITCH Sites
> >     >     >> http://www.freeswitch.org <http://www.freeswitch.org/>
> >     >     >> http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
> >     >     >> http://www.cluecon.com <http://www.cluecon.com/>
> >     >     >>
> >     >     >> FreeSWITCH-users mailing list
> >     >     >> FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     >     >> <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
> >     >     >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >     >>
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >     >> http://www.freeswitch.org <http://www.freeswitch.org/>
> >     >     >>
> >     >     >>
> >     >     >>
> >     >     >>
> >     >     >>
> >     >
> >
> _________________________________________________________________________
> >     >     >> Professional FreeSWITCH Consulting Services:
> >     >     >> consulting at freeswitch.org
> >     <mailto:consulting at freeswitch.org>
> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
> >>
> >     >     <mailto:consulting at freeswitch.org
> >     <mailto:consulting at freeswitch.org>
> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
> >>>
> >     >     >> http://www.freeswitchsolutions.com
> >     >     <http://www.freeswitchsolutions.com/>
> >     >     >>
> >     >     >> 
> >     >     >>  </>
> >     >     >>
> >     >     >> Official FreeSWITCH Sites
> >     >     >> http://www.freeswitch.org <http://www.freeswitch.org/>
> >     >     >> http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
> >     >     >> http://www.cluecon.com <http://www.cluecon.com/>
> >     >     >>
> >     >     >> FreeSWITCH-users mailing list
> >     >     >> FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
> >     >     >>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >     >>
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >     >> http://www.freeswitch.org <http://www.freeswitch.org/>
> >     >     >
> >     >     >
> >     >     >
> >     >
> >
> _________________________________________________________________________
> >     >     > Professional FreeSWITCH Consulting Services:
> >     >     > consulting at freeswitch.org
> >     <mailto:consulting at freeswitch.org>
> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
> >>
> >     >     <mailto:consulting at freeswitch.org
> >     <mailto:consulting at freeswitch.org>
> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
> >>>
> >     >     > http://www.freeswitchsolutions.com
> >     >     <http://www.freeswitchsolutions.com/>
> >     >     >
> >     >     > 
> >     >     >  </>
> >     >     >
> >     >     > Official FreeSWITCH Sites
> >     >     > http://www.freeswitch.org <http://www.freeswitch.org/>
> >     >     > http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
> >     >     > http://www.cluecon.com <http://www.cluecon.com/>
> >     >     >
> >     >     > FreeSWITCH-users mailing list
> >     >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     >     > <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
> >     >     >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >     >
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >     > http://www.freeswitch.org <http://www.freeswitch.org/>
> >     >     >
> >     >     >
> >     >     >
> >     >     >
> >     >
> >
> _________________________________________________________________________
> >     >     > Professional FreeSWITCH Consulting Services:
> >     >     > consulting at freeswitch.org
> >     <mailto:consulting at freeswitch.org>
> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
> >>
> >     >     > http://www.freeswitchsolutions.com
> >     >     <http://www.freeswitchsolutions.com/>
> >     >     >
> >     >     > 
> >     >     >  </>
> >     >     >
> >     >     > Official FreeSWITCH Sites
> >     >     > http://www.freeswitch.org <http://www.freeswitch.org/>
> >     >     > http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
> >     >     > http://www.cluecon.com <http://www.cluecon.com/>
> >     >     >
> >     >     > FreeSWITCH-users mailing list
> >     >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     >     >
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >     >
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     >     > http://www.freeswitch.org <http://www.freeswitch.org/>
> >     >     >
> >     >
> >     >
> >
> _________________________________________________________________________
> >     >     Professional FreeSWITCH Consulting Services:
> >     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
> >>
> >     > http://www.freeswitchsolutions.com
> >     <http://www.freeswitchsolutions.com/>
> >     >
> >     >     
> >     >  </>
> >     >
> >     >     Official FreeSWITCH Sites
> >     > http://www.freeswitch.org <http://www.freeswitch.org/>
> >     > http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
> >     > http://www.cluecon.com <http://www.cluecon.com/>
> >     >
> >     >     FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     > http://www.freeswitch.org <http://www.freeswitch.org/>
> >     >
> >     >
> >     >
> >     >
> >     >
> >     >
> >
> _________________________________________________________________________
> >     > Professional FreeSWITCH Consulting Services:
> >     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     > http://www.freeswitchsolutions.com
> >     >
> >     > 
> >     > 
> >     >
> >     > Official FreeSWITCH Sites
> >     > http://www.freeswitch.org
> >     > http://wiki.freeswitch.org
> >     > http://www.cluecon.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
> >     >
> >
> >
> _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com
> >
> >     
> >     
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org
> >     http://wiki.freeswitch.org
> >     http://www.cluecon.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
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From jmesquita at freeswitch.org  Thu Aug 15 22:55:16 2013
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 15 Aug 2013 15:55:16 -0300
Subject: [Freeswitch-users] Potential FreeSWITCH support for SMPP
In-Reply-To: <CAKbxfG99ZWAG=gmoTxG+ccE6WBGCuWXt9g1+U-QVKtCjcWZ93w@mail.gmail.com>
References: <20130812053755.96624167@mail.tritonwest.net>
	<520C8D49.3000402@quentustech.com>
	<CAKbxfG8X3G4t+swgC2e8k-y+kaYCx_ukBUi4mKK+oDh=snB9LQ@mail.gmail.com>
	<520CF03A.5070206@coppice.org>
	<CAKbxfG99ZWAG=gmoTxG+ccE6WBGCuWXt9g1+U-QVKtCjcWZ93w@mail.gmail.com>
Message-ID: <CABH8QX2+GpyRuNzfH1ckHQ-Es_nJ-74XUY5prqCsc6MoMA1tew@mail.gmail.com>

SMS in South America is still a huge business. I am assuming Africa as well?

Jo?o Mesquita
FreeSWITCH? Solutions


On Thu, Aug 15, 2013 at 3:39 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> It seems to be dying less when it comes to enabling it to SIP or XMPP.
>  I've watched the topic rise for the last few years and we seem on the
> verge of many SMS enabled SIP accounts from ITSP.  I think it will have a
> reprise when people start developing SMS based apps that can use a platform
> like FS to make the app design easier.
>
> I'm sure before too long when everyone has LTE, we will just learn to send
> data messages to each other over IP but the legacy stuff always seems to
> hang around doesn't it....
>
>
> On Thu, Aug 15, 2013 at 10:14 AM, Steve Underwood <steveu at coppice.org>wrote:
>
>> Logica released a Java SMPP platform. I don't know how well that works.
>> Like most SMPP stuff, it is long since abandoned. The SMS forum was
>> abandoned in something like 2007. I know SMS is dying now, but its
>> amazing how much infrastructure tools have been abandoned.
>>
>> I would need to do some work to release my code, as there is some
>> customer specific bits mixed in with it that would need sanitising.
>>
>> Regards,
>> Steve
>>
>> On 08/15/2013 10:40 PM, Anthony Minessale wrote:
>> >
>> > Didn't Steve say he has an old implementation to dust off earlier in
>> > this thread?
>> >
>> > On Aug 15, 2013 3:17 AM, "William King" <william.king at quentustech.com
>> > <mailto:william.king at quentustech.com>> wrote:
>> >
>> >     Dave,
>> >
>> >     Know of any current implementations? or would this be entirely new
>> >     functionality?
>> >
>> >     William King
>> >     Senior Engineer
>> >     Quentus Technologies, INC
>> >     1037 NE 65th St Suite 273
>> >     Seattle, WA 98115
>> >     Main: (877) 211-9337 <tel:%28877%29%20211-9337>
>> >     Office: (206) 388-4772 <tel:%28206%29%20388-4772>
>> >     Cell: (253) 686-5518 <tel:%28253%29%20686-5518>
>> >     william.king at quentustech.com <mailto:william.king at quentustech.com>
>> >
>> >     On 08/11/2013 10:37 PM, Dave R. Kompel wrote:
>> >     > William,
>> >     >
>> >     > The business case is for those of us that use FreeSWITCH as a main
>> >     > switch at a CLEC, where we have our own prefix assignments. In
>> >     order to
>> >     > do SMS on those numbers we need to interconnect with a SMS
>> backbone
>> >     > provider. SMPP is you're only option for that...
>> >     >
>> >     > --Dave
>> >     >
>> >     >
>> >
>> ------------------------------------------------------------------------
>> >     >     *From:* William King [mailto:william.king at quentustech.com
>> >     <mailto:william.king at quentustech.com>]
>> >     >     *To:* freeswitch-users at lists.freeswitch.org
>> >     <mailto:freeswitch-users at lists.freeswitch.org>
>> >     >     *Sent:* Fri, 09 Aug 2013 00:00:33 -0700
>> >     >     *Subject:* Re: [Freeswitch-users] Potential FreeSWITCH
>> >     support for SMPP
>> >     >
>> >     >     >From what I've seen, retail level SMS carriers(not sms
>> >     aggregators)
>> >     >     usually offer a mix of HTTP, XMPP, SIP SIMPLE, or SMPP
>> >     interfaces. I've
>> >     >     seen the higher volume SMS connections prefer SMPP or HTTP.
>> >     >
>> >     >     I'm exploring to see if there is a business case for
>> >     FreeSWITCH support
>> >     >     for SMPP(either direct support, or interface support through
>> an
>> >     >     independent application) that is not already covered by the
>> >     current FS
>> >     >     feature set.
>> >     >
>> >     >     William King
>> >     >     Senior Engineer
>> >     >     Quentus Technologies, INC
>> >     >     1037 NE 65th St Suite 273
>> >     >     Seattle, WA 98115
>> >     >     Main: (877) 211-9337
>> >     >     Office: (206) 388-4772
>> >     >     Cell: (253) 686-5518
>> >     > william.king at quentustech.com
>> >     <mailto:william.king at quentustech.com>
>> >     <mailto:william.king at quentustech.com
>> >     <mailto:william.king at quentustech.com>>
>> >     >
>> >     >     On 08/08/2013 04:23 AM, Raymond Chandler wrote:
>> >     >     > So, to the best of my knowledge, SMPP is strictly for SMS
>> >     so you can
>> >     >     > route SMS to your clients via FS to SIMPLE / dingaling /
>> >     etc clients
>> >     >     >
>> >     >     > -Ray
>> >     >     >
>> >     >     > On Aug 7, 2013 11:05 PM, "Victor Chukalovskiy"
>> >     >     > <victor.chukalovskiy at gmail.com
>> >     <mailto:victor.chukalovskiy at gmail.com>
>> >     >     <mailto:victor.chukalovskiy at gmail.com
>> >     <mailto:victor.chukalovskiy at gmail.com>>
>> >     >     <mailto:victor.chukalovskiy at gmail.com
>> >     <mailto:victor.chukalovskiy at gmail.com>
>> >     >     <mailto:victor.chukalovskiy at gmail.com
>> >     <mailto:victor.chukalovskiy at gmail.com>>>>
>> >     >     > wrote:
>> >     >     >
>> >     >     > So am I as well as another ClueCon attendee I spoke to.
>> >     >     >
>> >     >     > What is the usage scenario you are looking at?
>> >     >     >
>> >     >     > On 13-08-07 06:46 PM, Dmitry Lysenko wrote:
>> >     >     >> Yes. I am interested.
>> >     >     >>
>> >     >     >> Dmitry.
>> >     >     >>
>> >     >     >>
>> >     >     >> 2013/8/8 William King <william.king at quentustech.com
>> >     <mailto:william.king at quentustech.com>
>> >     >     <mailto:william.king at quentustech.com
>> >     <mailto:william.king at quentustech.com>>
>> >     >     >> <mailto:william.king at quentustech.com
>> >     <mailto:william.king at quentustech.com>
>> >     >     <mailto:william.king at quentustech.com
>> >     <mailto:william.king at quentustech.com>>>>
>> >     >     >>
>> >     >     >> Is anyone on this list interested in FreeSWITCH support for
>> >     >     >> SMPP for SMS
>> >     >     >> messages?
>> >     >     >>
>> >     >     >> For more information about the SMPP protocol checkout:
>> >     >     >> http://en.wikipedia.org/wiki/Short_Message_Peer-to-Peer
>> >     >     >>
>> >     >     >> If so feel free to contact me on or off this list.
>> >     >     >> --
>> >     >     >> William King
>> >     >     >> Senior Engineer
>> >     >     >> Quentus Technologies, INC
>> >     >     >> 1037 NE 65th St Suite 273
>> >     >     >> Seattle, WA 98115
>> >     >     >> Main: (877) 211-9337 <tel:%28877%29%20211-9337>
>> >     <tel:%28877%29%20211-9337>
>> >     >     >> Office: (206) 388-4772 <tel:%28206%29%20388-4772>
>> >     <tel:%28206%29%20388-4772>
>> >     >     >> Cell: (253) 686-5518 <tel:%28253%29%20686-5518>
>> >     >     >> william.king at quentustech.com
>> >     <mailto:william.king at quentustech.com>
>> >     >     <mailto:william.king at quentustech.com
>> >     <mailto:william.king at quentustech.com>>
>> >     >     <mailto:william.king at quentustech.com
>> >     <mailto:william.king at quentustech.com>
>> >     >     <mailto:william.king at quentustech.com
>> >     <mailto:william.king at quentustech.com>>>
>> >     >     >>
>> >     >     >>
>> >     >
>> >
>> _________________________________________________________________________
>> >     >     >> Professional FreeSWITCH Consulting Services:
>> >     >     >> consulting at freeswitch.org
>> >     <mailto:consulting at freeswitch.org>
>> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
>> >>
>> >     >     <mailto:consulting at freeswitch.org
>> >     <mailto:consulting at freeswitch.org>
>> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
>> >>>
>> >     >     >> http://www.freeswitchsolutions.com
>> >     >     <http://www.freeswitchsolutions.com/>
>> >     >     >>
>> >     >     >> 
>> >     >     >>  </>
>> >     >     >>
>> >     >     >> Official FreeSWITCH Sites
>> >     >     >> http://www.freeswitch.org <http://www.freeswitch.org/>
>> >     >     >> http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>> >     >     >> http://www.cluecon.com <http://www.cluecon.com/>
>> >     >     >>
>> >     >     >> FreeSWITCH-users mailing list
>> >     >     >> FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>> >     >     >> <mailto:FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
>> >     >     >>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >     >     >>
>> >     >
>> >     UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >     >     >> http://www.freeswitch.org <http://www.freeswitch.org/>
>> >     >     >>
>> >     >     >>
>> >     >     >>
>> >     >     >>
>> >     >     >>
>> >     >
>> >
>> _________________________________________________________________________
>> >     >     >> Professional FreeSWITCH Consulting Services:
>> >     >     >> consulting at freeswitch.org
>> >     <mailto:consulting at freeswitch.org>
>> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
>> >>
>> >     >     <mailto:consulting at freeswitch.org
>> >     <mailto:consulting at freeswitch.org>
>> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
>> >>>
>> >     >     >> http://www.freeswitchsolutions.com
>> >     >     <http://www.freeswitchsolutions.com/>
>> >     >     >>
>> >     >     >> 
>> >     >     >>  </>
>> >     >     >>
>> >     >     >> Official FreeSWITCH Sites
>> >     >     >> http://www.freeswitch.org <http://www.freeswitch.org/>
>> >     >     >> http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>> >     >     >> http://www.cluecon.com <http://www.cluecon.com/>
>> >     >     >>
>> >     >     >> FreeSWITCH-users mailing list
>> >     >     >> FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
>> >     >     >>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >     >     >>
>> >     >
>> >     UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >     >     >> http://www.freeswitch.org <http://www.freeswitch.org/>
>> >     >     >
>> >     >     >
>> >     >     >
>> >     >
>> >
>> _________________________________________________________________________
>> >     >     > Professional FreeSWITCH Consulting Services:
>> >     >     > consulting at freeswitch.org
>> >     <mailto:consulting at freeswitch.org>
>> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
>> >>
>> >     >     <mailto:consulting at freeswitch.org
>> >     <mailto:consulting at freeswitch.org>
>> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
>> >>>
>> >     >     > http://www.freeswitchsolutions.com
>> >     >     <http://www.freeswitchsolutions.com/>
>> >     >     >
>> >     >     > 
>> >     >     >  </>
>> >     >     >
>> >     >     > Official FreeSWITCH Sites
>> >     >     > http://www.freeswitch.org <http://www.freeswitch.org/>
>> >     >     > http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>> >     >     > http://www.cluecon.com <http://www.cluecon.com/>
>> >     >     >
>> >     >     > FreeSWITCH-users mailing list
>> >     >     > FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>> >     >     > <mailto:FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>>
>> >     >     >
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >     >     >
>> >     >
>> >     UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >     >     > http://www.freeswitch.org <http://www.freeswitch.org/>
>> >     >     >
>> >     >     >
>> >     >     >
>> >     >     >
>> >     >
>> >
>> _________________________________________________________________________
>> >     >     > Professional FreeSWITCH Consulting Services:
>> >     >     > consulting at freeswitch.org
>> >     <mailto:consulting at freeswitch.org>
>> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
>> >>
>> >     >     > http://www.freeswitchsolutions.com
>> >     >     <http://www.freeswitchsolutions.com/>
>> >     >     >
>> >     >     > 
>> >     >     >  </>
>> >     >     >
>> >     >     > Official FreeSWITCH Sites
>> >     >     > http://www.freeswitch.org <http://www.freeswitch.org/>
>> >     >     > http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>> >     >     > http://www.cluecon.com <http://www.cluecon.com/>
>> >     >     >
>> >     >     > FreeSWITCH-users mailing list
>> >     >     > FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>> >     >     >
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >     >     >
>> >     >
>> >     UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >     >     > http://www.freeswitch.org <http://www.freeswitch.org/>
>> >     >     >
>> >     >
>> >     >
>> >
>> _________________________________________________________________________
>> >     >     Professional FreeSWITCH Consulting Services:
>> >     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> >     <mailto:consulting at freeswitch.org <mailto:consulting at freeswitch.org
>> >>
>> >     > http://www.freeswitchsolutions.com
>> >     <http://www.freeswitchsolutions.com/>
>> >     >
>> >     >     
>> >     >  </>
>> >     >
>> >     >     Official FreeSWITCH Sites
>> >     > http://www.freeswitch.org <http://www.freeswitch.org/>
>> >     > http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>> >     > http://www.cluecon.com <http://www.cluecon.com/>
>> >     >
>> >     >     FreeSWITCH-users mailing list
>> >     > FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     >     <mailto:FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>>
>> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >     >
>> >     UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >     > http://www.freeswitch.org <http://www.freeswitch.org/>
>> >     >
>> >     >
>> >     >
>> >     >
>> >     >
>> >     >
>> >
>> _________________________________________________________________________
>> >     > Professional FreeSWITCH Consulting Services:
>> >     > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> >     > http://www.freeswitchsolutions.com
>> >     >
>> >     > 
>> >     > 
>> >     >
>> >     > Official FreeSWITCH Sites
>> >     > http://www.freeswitch.org
>> >     > http://wiki.freeswitch.org
>> >     > http://www.cluecon.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
>> >     >
>> >
>> >
>> _________________________________________________________________________
>> >     Professional FreeSWITCH Consulting Services:
>> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> >     http://www.freeswitchsolutions.com
>> >
>> >     
>> >     
>> >
>> >     Official FreeSWITCH Sites
>> >     http://www.freeswitch.org
>> >     http://wiki.freeswitch.org
>> >     http://www.cluecon.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
>> >
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveayre at gmail.com  Thu Aug 15 23:00:54 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 15 Aug 2013 20:00:54 +0100
Subject: [Freeswitch-users] Auto-detecting Local IP when no gateway set
In-Reply-To: <CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
Message-ID: <CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>

I was going to suggest this too, but if it's on a LAN I'm guessing there
won't be a server and serving it from httpd on the localhost would only see
127.0.0.1

...although you could lookup the current IP of the LAN interface from a
script on the local webserver.

-Steve


On 15 August 2013 13:14, Raymond Chandler <intralanman at freeswitch.org>wrote:

> Alternatively, you could use xml_curl to serve your configs so that you
> don't actually need local_ip_v4 as you can figure it out based on the IP
> hitting the web server. Doing that will allow you to set the IP in all of
> the configs that need it using functionality that already exists in
> FreeSWITCH today.
>
> -Ray
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveayre at gmail.com  Thu Aug 15 23:03:19 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 15 Aug 2013 20:03:19 +0100
Subject: [Freeswitch-users] Auto-detecting Local IP when no gateway set
In-Reply-To: <CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
Message-ID: <CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>

Personally I think allowing something like <param name="sip-ip"
value="interface:eth0"/> to pick up the primary IP of that interface might
work best.

Anthm, can you clarify if FS already has any such functionality? Or time
for a wishlist Jira?

Doing so could work alongside auto-restart=true and therefore pick up IP
changes such as when using DHCP.



On 15 August 2013 20:00, Steven Ayre <steveayre at gmail.com> wrote:

> I was going to suggest this too, but if it's on a LAN I'm guessing there
> won't be a server and serving it from httpd on the localhost would only see
> 127.0.0.1
>
> ...although you could lookup the current IP of the LAN interface from a
> script on the local webserver.
>
> -Steve
>
>
> On 15 August 2013 13:14, Raymond Chandler <intralanman at freeswitch.org>wrote:
>
>> Alternatively, you could use xml_curl to serve your configs so that you
>> don't actually need local_ip_v4 as you can figure it out based on the IP
>> hitting the web server. Doing that will allow you to set the IP in all of
>> the configs that need it using functionality that already exists in
>> FreeSWITCH today.
>>
>> -Ray
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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  Thu Aug 15 23:15:30 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 15 Aug 2013 14:15:30 -0500
Subject: [Freeswitch-users] Auto-detecting Local IP when no gateway set
In-Reply-To: <CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
	<CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>
Message-ID: <CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>

No such functionality, meanwhile you could use the #exec config directive
similar to SSI in apache to manually run a shell command every time the
config is loaded to call a command that prints the ip.
On Aug 15, 2013 2:08 PM, "Steven Ayre" <steveayre at gmail.com> wrote:

> Personally I think allowing something like <param name="sip-ip"
> value="interface:eth0"/> to pick up the primary IP of that interface might
> work best.
>
> Anthm, can you clarify if FS already has any such functionality? Or time
> for a wishlist Jira?
>
> Doing so could work alongside auto-restart=true and therefore pick up IP
> changes such as when using DHCP.
>
>
>
> On 15 August 2013 20:00, Steven Ayre <steveayre at gmail.com> wrote:
>
>> I was going to suggest this too, but if it's on a LAN I'm guessing there
>> won't be a server and serving it from httpd on the localhost would only see
>> 127.0.0.1
>>
>> ...although you could lookup the current IP of the LAN interface from a
>> script on the local webserver.
>>
>> -Steve
>>
>>
>> On 15 August 2013 13:14, Raymond Chandler <intralanman at freeswitch.org>wrote:
>>
>>> Alternatively, you could use xml_curl to serve your configs so that you
>>> don't actually need local_ip_v4 as you can figure it out based on the IP
>>> hitting the web server. Doing that will allow you to set the IP in all of
>>> the configs that need it using functionality that already exists in
>>> FreeSWITCH today.
>>>
>>> -Ray
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 karl at xtronics.com  Fri Aug 16 01:14:35 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Thu, 15 Aug 2013 16:14:35 -0500
Subject: [Freeswitch-users] freetdm rxgain txgain
In-Reply-To: <CAHKqMb7uoZKy0O4NAdSnFxdCzxv5sTJ5S4rJYDncv6dgyjPNEA@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
	<CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
	<CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
	<520C4938.4010202@xtronics.com>
	<CAHKqMb7uoZKy0O4NAdSnFxdCzxv5sTJ5S4rJYDncv6dgyjPNEA@mail.gmail.com>
Message-ID: <520D44BB.5050805@xtronics.com>

On 08/15/2013 04:59 AM, Nandy Dagondon wrote:
> Hi Karl,
>
> Did you test the milliwatt tone using the FXS port? I asked because you said it's not sending the
> ringing signal. Do you mean the phones did not ring. The Digium cards require a separate power
> supply feed. Without it - it can't generate the 100V ringing voltage. This is very obvious but let's
> not discounting this possibility. :-)
>
> /Nandy

I used the tone on both fxo and fxs ports - the extra power connector is there - for some reason the 
gain settings in /etc/freeswitch/freetdm.conf don't seem to get applied.

The key test is the output of
  $ dahdi_diag

I just ran it again while is active to see if it makes a difference.

I looks like it wants to see settings in zt.conf

Which config file should this be using - zt.conf or freetdm.conf ? I can find documentation that 
points to both?




--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

Underlying most arguments against the free market
is a lack of belief in freedom itself.
- Milton Freidman
--------------------------------------------------------------------------------


From ryangard at gmail.com  Fri Aug 16 01:23:39 2013
From: ryangard at gmail.com (Ryan Gard)
Date: Thu, 15 Aug 2013 17:23:39 -0400
Subject: [Freeswitch-users] deny_refer_requests failure
In-Reply-To: <CAPEdYTh5DUjxfGtyFk+QfT2HaaFaZZnpyyJfX5sgoesp+dxbyw@mail.gmail.com>
References: <CAAQxoQH9R_nb5hv=nMqAknwL+ZsSaEiL3EcUF5jKnb=-YMva4w@mail.gmail.com>
	<CAPEdYTh5DUjxfGtyFk+QfT2HaaFaZZnpyyJfX5sgoesp+dxbyw@mail.gmail.com>
Message-ID: <CAAQxoQHDHCJ6mPb=vVCqfQaS8VnB9Gt5JBX1CjskbNHDA5r62g@mail.gmail.com>

Correct: attempting to use it to block blind transfers.

I'll toss it into JIRA.


On Thu, Aug 15, 2013 at 7:01 AM, Peter Olsson <peter at olssononline.se> wrote:

> I believe it only works when using REFER with replaces headers set. Are
> you trying REFER as a blind transfer maybe? In that case it won't work.
> However, it probably should be implemented to work in that case as well, so
> please file a Jira for it.
>
> /Peter
>
>
> 2013/8/14 Ryan Gard <ryangard at gmail.com>
>
>> Hey,
>>
>> Running into issues with deny_refer_requests. I have verified that it is
>> setting the variable appropriately on both the a-leg and b-leg of the call,
>> but it still processes the refer request without a second thought upon
>> receiving it.
>>
>> Is there anything specific on top of setting the variable to 'true' I
>> should be keeping tabs on? It doesn't seem to have too much in the way of
>> documentation, but the references I have seen say that it should work
>> without issue as long as one of the legs (both legs have it set) has it set
>> to true.
>>
>> Thanks :)
>>
>> --
>> Ryan Gard
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Ryan Gard
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From eidevm5 at gmail.com  Fri Aug 16 03:18:18 2013
From: eidevm5 at gmail.com (Peter)
Date: Fri, 16 Aug 2013 09:18:18 +1000
Subject: [Freeswitch-users] Establishing SRTP from SBC to endpoint
In-Reply-To: <3BF36541-3D31-4460-AF01-5564BB0FD4CC@jerris.com>
References: <CABkS=xqD0mukebja+w3aurrhcUJ4Gp=DcOpkh+BDnCpOKr32CA@mail.gmail.com>
	<CAAYhoXLzsB-o=5VNBaRsR9xdS8bm=j5zKxe8ideg5-jmKU=fKw@mail.gmail.com>
	<CABkS=xrLBX2A1ZLO9gx7mW2yQ2e9LTjyDaNOqJXH2DFADS4KYg@mail.gmail.com>
	<CABkS=xp5XcskU6uJoM-7iHy3taPm3M=QiD29UO9EfToDTR-Fug@mail.gmail.com>
	<CABkS=xqFUnp0D1SFNZh+n8XNUbxTKYACYQ__+tTW49ffL_uPbQ@mail.gmail.com>
	<3BF36541-3D31-4460-AF01-5564BB0FD4CC@jerris.com>
Message-ID: <CABkS=xrTbvQ+yMj914vOVN+DRkvwkygu3gkBQXxzSndTpqTfiQ@mail.gmail.com>

Might well be the case.   I was just going off the release notes at:

https://wiki.freeswitch.org/wiki/Release_Notes

which says it was introduced in  Stable 1.2.9


On Thu, Aug 15, 2013 at 11:29 PM, Michael Jerris <mike at jerris.com> wrote:

> I thought the change from sip_ to rtp_ for some variables was only in the
> 1.4 branch, not in 1.2.9.
>
> On Aug 15, 2013, at 1:57 AM, Peter <eidevm5 at gmail.com> wrote:
>
> Let me correct my last email.
>
> If I use rtp_secure_media instead of sip_secure_media, the outgoing call
> uses RTP and not SRTP.
>
> rtp_secure_media was supposed to have been introduced in 1.2.9, so I
> wonder what the difference is?
>
>
> On Thu, Aug 15, 2013 at 3:52 PM, Peter <eidevm5 at gmail.com> wrote:
>
>> Finally got it going.  I don't know how many combinations I tried.
>>
>> All I needed was the sip_secure_media (or rtp_secure_media, which is the
>> new name) set to true in the dialplan on the SBC.
>>
>>
>> On Wed, Aug 14, 2013 at 11:42 AM, Peter <eidevm5 at gmail.com> wrote:
>>
>>> Hi Carlos.
>>>
>>> Didn't realise rtp_secure_media existed.  After searching I saw:
>>>
>>>
>>> https://wiki.freeswitch.org/wiki/Release_Notes#rtp_secure_media_.28was_sip_secure_media.29
>>>
>>> which says it was introduced in 1.2.9
>>>
>>> However, it's a little ambiguous as to whether sip_secure_media was
>>> deprecated.
>>>
>>> Anyway, I tried using rtp_secure_media instead, but I still can't get
>>> SRTP working.
>>>
>>>
>>> I did some testing with some other SIP clients.   In particular,
>>> csipsimple.  On the client, if I set SRTP to be optional, the media stream
>>> uses RTP.   However, if I set SRTP to be mandatory, when I try to call it,
>>> Freeswitch receives:
>>>
>>>    SIP/2.0 488 Not Acceptable Here
>>>
>>> Which seems to indicate that something is not is not right with the SRTP
>>> setup.
>>>
>>> There's a full debug from the FS1 (the freeswitch server where the
>>> csipsimple client is registered to) at:
>>>
>>> http://pastebin.freeswitch.org/21295
>>>
>>> Note in the debug I have sdp_secure_savp_only set to true.   I've tried
>>> disabling this setting, but get the same result.
>>>
>>> Thanks
>>>
>>> Peter
>>>
>>>
>>>
>>>
>>>
>>> On Tue, Aug 13, 2013 at 11:06 PM, Carlos Flor <jackal at cybershroud.net>wrote:
>>>
>>>> Try using rtp_secure_media=true instead of sip_secure_media.  If you
>>>> are trying to set it on the b-leg, you probably want to use export instead
>>>> of set, or use nolocal:rtp_secure_media.
>>>>
>>>> Hope that helps.
>>>>
>>>>
>>>> On Mon, Aug 12, 2013 at 10:26 PM, Peter <eidevm5 at gmail.com> wrote:
>>>>
>>>>> In my environment, I have the following (simplified) setup:
>>>>>
>>>>> FS1  ----  FS SBC ---  FS2
>>>>>
>>>>> Phones registered to FS1 (100x) use TLS/SRTP and phones registered to
>>>>> FS2 (200x) use SIP/RTP
>>>>>
>>>>> FS1 has inbound-bypass-media set to true to allow SRTP peer to peer
>>>>> and direct to the SBC.
>>>>>
>>>>> If I make an inbound call (eg: 1000 to 2000), SRTP is correctly
>>>>> established between the phone and SBC with RTP on the other side of the SBC
>>>>> to the internal phone.
>>>>>
>>>>> However, when I try it the other way, I can't get SRTP established
>>>>> from the SBC to the external phone.
>>>>>
>>>>> I've been using https://wiki.freeswitch.org/wiki/Secure_RTP as a
>>>>> guide.
>>>>>
>>>>> I've even tried explicitly setting sip_secure_media to true on the SBC
>>>>> and FS1.
>>>>>
>>>>> The dialplan on the SBC has:
>>>>>
>>>>>   <extension name="outgoing">
>>>>>         <condition field="destination_number"
>>>>> expression="^(10[0-9][0-9])$">
>>>>>             <action application="set" data="sip_secure_media=true"/>
>>>>>             <action application="bridge" data="sofia/external/${
>>>>> destination_number}@10.1.1.204"/>
>>>>>         </condition>
>>>>>   </extension>
>>>>>
>>>>>
>>>>> And on FS1, the dialplan has:
>>>>>
>>>>>    <extension name="Local-Numbers">
>>>>>       <condition field="destination_number"
>>>>> expression="^(10[01][0-9])$">
>>>>>         <action application="export" data="dialed_extension=$1"/>
>>>>>         <action application="set" data="sip_secure_media=true"/>
>>>>>         <action application="bridge" data="user/${dialed_extension}@
>>>>> ${domain_name}"/>
>>>>>       </condition>
>>>>>     </extension>
>>>>>
>>>>>
>>>>> Note that I've been testing this against two phones with SRTP enabled,
>>>>> but only one that is using TLS.  I get the same result calling each phone.
>>>>>
>>>>> On a related point, what it the step required for a TLS connection
>>>>> from the SBC to the phone?   I'm assume the phone just needs the CA cert
>>>>> from the SBC.  Correct?
>>>>>
>>>>> Any information as to where I'm going wrong will be gratefully
>>>>> accepted.
>>>>>
>>>>> Thanks
>>>>>
>>>>> Peter
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Fri Aug 16 03:19:29 2013
From: eidevm5 at gmail.com (Peter)
Date: Fri, 16 Aug 2013 09:19:29 +1000
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CAMvhjJut4W_hwnAG45o2wFWCOKknwPzE8vFGrkNv6JoZZA8o1Q@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
	<CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
	<CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
	<CALn233ovuLR8rgD==bqiOi+Fa=yDwS8azsrD3cWkBz9sz7Mw7w@mail.gmail.com>
	<CAMvhjJut4W_hwnAG45o2wFWCOKknwPzE8vFGrkNv6JoZZA8o1Q@mail.gmail.com>
Message-ID: <CABkS=xorE-TTwdKb=m0pO-C4stk4JC5KBJGcpbrhpFhdeMYKbQ@mail.gmail.com>

Of course you can use a softphone on the freeswitch server.   It would
eliminate any network issues.


On Thu, Aug 15, 2013 at 7:12 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> I also think that there may be some network problem as my windows pc is
> not showing that some linux machine is connected but the linux machine
> shows the windows system connected to it. Also can i install the softphone
> on the same linux machine on which my freeswitch runs and can check whether
> it gets connected or not...???
>  On Aug 15, 2013 5:10 AM, "Mitch Capper" <mitch.capper at gmail.com> wrote:
>
>> You may want to try FSClient its a freeswitch based windows soft phone
>> and you can connect fs_cli (or fs_logger.pl) to it and see what happens
>> when it tries to connect to your server.
>>
>> ~mitch
>>
>>
>> On Wed, Aug 14, 2013 at 4:06 PM, Peter <eidevm5 at gmail.com> wrote:
>>
>>> My mistake, I meant loglevel rather than logging.
>>>
>>> So when you say "no progress" are you seeing nothing at all on the FS
>>> console?
>>>
>>> If so, then that indicates a network problem as FS isn't even seeing the
>>> registration requests.
>>>
>>> Can you ping the windows box from the FS server?
>>>
>>>
>>> On Wed, Aug 14, 2013 at 7:04 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> I went for "sofia loglevel 5" instead of "sofia logging 5" but still no
>>>> progress in connection of softphone...:-(
>>>> Thanks
>>>> Ashish Mishra
>>>> On Aug 14, 2013 2:14 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>>>>
>>>>> Also the command that you had mentioned "sofia logging all 5" gives me
>>>>> an error message :
>>>>> Unknown command [logging]
>>>>> On Aug 14, 2013 7:24 AM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>>
>>>>>> Connect to the FS console with fs_cli and do
>>>>>>
>>>>>> sofia logging all 5
>>>>>>
>>>>>> then try to register your softphone.    If you don't even see a
>>>>>> registration attempt on the FS console, it means you have a network problem.
>>>>>>
>>>>>> If the network isn't a problem, the debug output should give you a
>>>>>> clue as to what the problem is.
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra <itsme.kunnu at gmail.com
>>>>>> > wrote:
>>>>>>
>>>>>>> Thank you Peter...you mean to say that i should first turn on the
>>>>>>> fs_cli and then retry to connect the softphone...??? I would also like to
>>>>>>> remind you that fs_cli and freeswitch are running on the same machine and i
>>>>>>> am trying to connect thru a network cable my windows 8 pc that has the
>>>>>>> softphone on it...
>>>>>>> Regards
>>>>>>> Ashish Mishra
>>>>>>> On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>>>>
>>>>>>>> Have you turned on debugging from the FS cli and seen if any
>>>>>>>> registration requests come in?
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <
>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> I installed freeswitch on my ubuntu 12.04 machine...but when i am
>>>>>>>>> trying to connect the softphone installed on my windows 8 pc with ubuntu
>>>>>>>>> machine the softphone gives me an error that account not enabled...i have
>>>>>>>>> used a network cable to connect the two machines...also the firewall in
>>>>>>>>> both the machines is disabled...
>>>>>>>>> Kindly help..
>>>>>>>>>
>>>>>>>>> Ashish Mishra
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jpyle at fidelityvoice.com  Fri Aug 16 05:10:54 2013
From: jpyle at fidelityvoice.com (Jeff Pyle)
Date: Thu, 15 Aug 2013 21:10:54 -0400
Subject: [Freeswitch-users] squelch repeated log entries for Ping failed
Message-ID: <CACYJG3K5oWUHquON_dnFAEd+zgVsrWb-8d=xOW2=HciLr+OT6A@mail.gmail.com>

Hello,

When a gateway is not reachable I see messages like this in the log:
  [WARNING] sofia.c:4880 Ping failed gw1 with code 408 - count 1/0/1, state
DOWN

This message continues to repeat until the gateway becomes reachable.

I do like to know a gateway is down, but I don't need to know if a gateway
is *still* down.  It fills up the syslog on a relatively limited embedded
device in this case.  Is there a way to limit logging to only when a
gateway changes state?

I tried the "$RepeatedMsgReduction on" option in rsyslog but it seems
that's no longer available.  Hmm.

Version 1.2.11-n20130813T230126Z-1~wheezy+1+git~20130813T212504Z~2131cd6cf2
(-n20130813T230126Z-1~wheezy+1git 2131cd6 2013-08-13 21:25:04Z)


Regards,
Jeff
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From sirimmfs at gmail.com  Fri Aug 16 05:17:15 2013
From: sirimmfs at gmail.com (Siri MM)
Date: Fri, 16 Aug 2013 11:17:15 +1000
Subject: [Freeswitch-users] Auto-detecting Local IP when no gateway set
In-Reply-To: <CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
	<CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>
	<CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>
Message-ID: <CAHccO0X_Dr+fP8eHmkeTVwergLDaP+kjAbQ0n_p24x-NdXY1tQ@mail.gmail.com>

Thanks for all the suggestions, As suggested by Steven, I have raised a
Jira wishlist - http://jira.freeswitch.org/browse/FS-5707

Anthony, could you please elaborate (or redirect me to the wiki page) on
your config directive suggestion? Where do I set this in Freeswitch, and
how do I make it run on config reload?

Thanks!


On Fri, Aug 16, 2013 at 5:15 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> No such functionality, meanwhile you could use the #exec config directive
> similar to SSI in apache to manually run a shell command every time the
> config is loaded to call a command that prints the ip.
> On Aug 15, 2013 2:08 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>
>> Personally I think allowing something like <param name="sip-ip"
>> value="interface:eth0"/> to pick up the primary IP of that interface might
>> work best.
>>
>> Anthm, can you clarify if FS already has any such functionality? Or time
>> for a wishlist Jira?
>>
>> Doing so could work alongside auto-restart=true and therefore pick up IP
>> changes such as when using DHCP.
>>
>>
>>
>> On 15 August 2013 20:00, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> I was going to suggest this too, but if it's on a LAN I'm guessing there
>>> won't be a server and serving it from httpd on the localhost would only see
>>> 127.0.0.1
>>>
>>> ...although you could lookup the current IP of the LAN interface from a
>>> script on the local webserver.
>>>
>>> -Steve
>>>
>>>
>>> On 15 August 2013 13:14, Raymond Chandler <intralanman at freeswitch.org>wrote:
>>>
>>>> Alternatively, you could use xml_curl to serve your configs so that you
>>>> don't actually need local_ip_v4 as you can figure it out based on the IP
>>>> hitting the web server. Doing that will allow you to set the IP in all of
>>>> the configs that need it using functionality that already exists in
>>>> FreeSWITCH today.
>>>>
>>>> -Ray
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at davri.com  Fri Aug 16 05:46:02 2013
From: bdfoster at davri.com (Brian Foster)
Date: Thu, 15 Aug 2013 21:46:02 -0400
Subject: [Freeswitch-users] Auto-detecting Local IP when no gateway set
In-Reply-To: <CAHccO0X_Dr+fP8eHmkeTVwergLDaP+kjAbQ0n_p24x-NdXY1tQ@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
	<CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>
	<CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>
	<CAHccO0X_Dr+fP8eHmkeTVwergLDaP+kjAbQ0n_p24x-NdXY1tQ@mail.gmail.com>
Message-ID: <CAHgk4BVujRwVT55BfLLo8kcLU76Jr09hdQua01dvYYuL7tuVeg@mail.gmail.com>

#!/bin/bash

# interface2ip.sh

/sbin/ifconfig $1 | grep "inet addr" | awk -F: '{print $2}' | awk '{print
$1}'

----

./interface2ip.sh eth0
Returns ip assigned to eth0

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
Thanks for all the suggestions, As suggested by Steven, I have raised a
Jira wishlist - http://jira.freeswitch.org/browse/FS-5707

Anthony, could you please elaborate (or redirect me to the wiki page) on
your config directive suggestion? Where do I set this in Freeswitch, and
how do I make it run on config reload?

Thanks!


On Fri, Aug 16, 2013 at 5:15 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> No such functionality, meanwhile you could use the #exec config directive
> similar to SSI in apache to manually run a shell command every time the
> config is loaded to call a command that prints the ip.
>  On Aug 15, 2013 2:08 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>
>> Personally I think allowing something like <param name="sip-ip"
>> value="interface:eth0"/> to pick up the primary IP of that interface might
>> work best.
>>
>> Anthm, can you clarify if FS already has any such functionality? Or time
>> for a wishlist Jira?
>>
>> Doing so could work alongside auto-restart=true and therefore pick up IP
>> changes such as when using DHCP.
>>
>>
>>
>> On 15 August 2013 20:00, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> I was going to suggest this too, but if it's on a LAN I'm guessing there
>>> won't be a server and serving it from httpd on the localhost would only see
>>> 127.0.0.1
>>>
>>> ...although you could lookup the current IP of the LAN interface from a
>>> script on the local webserver.
>>>
>>> -Steve
>>>
>>>
>>> On 15 August 2013 13:14, Raymond Chandler <intralanman at freeswitch.org>wrote:
>>>
>>>> Alternatively, you could use xml_curl to serve your configs so that you
>>>> don't actually need local_ip_v4 as you can figure it out based on the IP
>>>> hitting the web server. Doing that will allow you to set the IP in all of
>>>> the configs that need it using functionality that already exists in
>>>> FreeSWITCH today.
>>>>
>>>> -Ray
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 bdfoster at davri.com  Fri Aug 16 05:49:57 2013
From: bdfoster at davri.com (Brian Foster)
Date: Thu, 15 Aug 2013 21:49:57 -0400
Subject: [Freeswitch-users] Auto-detecting Local IP when no gateway set
In-Reply-To: <CAHgk4BVujRwVT55BfLLo8kcLU76Jr09hdQua01dvYYuL7tuVeg@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
	<CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>
	<CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>
	<CAHccO0X_Dr+fP8eHmkeTVwergLDaP+kjAbQ0n_p24x-NdXY1tQ@mail.gmail.com>
	<CAHgk4BVujRwVT55BfLLo8kcLU76Jr09hdQua01dvYYuL7tuVeg@mail.gmail.com>
Message-ID: <CAHgk4BUd9cB9wzWATSg06MwzdQs8+uS+LpdDQN5SkYhTA6pg_A@mail.gmail.com>

Also I think you can do this in your freeswitch config:

<param name="sip-ip" value="${system interface2ip.sh eth0}"/>

...in your sip profile or similarly in vars.xml.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 15, 2013 9:46 PM, "Brian Foster" <bdfoster at davri.com> wrote:

> #!/bin/bash
>
> # interface2ip.sh
>
> /sbin/ifconfig $1 | grep "inet addr" | awk -F: '{print $2}' | awk '{print
> $1}'
>
> ----
>
> ./interface2ip.sh eth0
> Returns ip assigned to eth0
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana
>
> Sent from a mobile device.
> Thanks for all the suggestions, As suggested by Steven, I have raised a
> Jira wishlist - http://jira.freeswitch.org/browse/FS-5707
>
> Anthony, could you please elaborate (or redirect me to the wiki page) on
> your config directive suggestion? Where do I set this in Freeswitch, and
> how do I make it run on config reload?
>
> Thanks!
>
>
> On Fri, Aug 16, 2013 at 5:15 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> No such functionality, meanwhile you could use the #exec config directive
>> similar to SSI in apache to manually run a shell command every time the
>> config is loaded to call a command that prints the ip.
>>  On Aug 15, 2013 2:08 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>>
>>> Personally I think allowing something like <param name="sip-ip"
>>> value="interface:eth0"/> to pick up the primary IP of that interface might
>>> work best.
>>>
>>> Anthm, can you clarify if FS already has any such functionality? Or time
>>> for a wishlist Jira?
>>>
>>> Doing so could work alongside auto-restart=true and therefore pick up IP
>>> changes such as when using DHCP.
>>>
>>>
>>>
>>> On 15 August 2013 20:00, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>>> I was going to suggest this too, but if it's on a LAN I'm guessing
>>>> there won't be a server and serving it from httpd on the localhost would
>>>> only see 127.0.0.1
>>>>
>>>> ...although you could lookup the current IP of the LAN interface from a
>>>> script on the local webserver.
>>>>
>>>> -Steve
>>>>
>>>>
>>>> On 15 August 2013 13:14, Raymond Chandler <intralanman at freeswitch.org>wrote:
>>>>
>>>>> Alternatively, you could use xml_curl to serve your configs so that
>>>>> you don't actually need local_ip_v4 as you can figure it out based on the
>>>>> IP hitting the web server. Doing that will allow you to set the IP in all
>>>>> of the configs that need it using functionality that already exists in
>>>>> FreeSWITCH today.
>>>>>
>>>>> -Ray
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 karl at xtronics.com  Fri Aug 16 07:36:35 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Thu, 15 Aug 2013 22:36:35 -0500
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <CAHgk4BUd9cB9wzWATSg06MwzdQs8+uS+LpdDQN5SkYhTA6pg_A@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
	<CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>
	<CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>
	<CAHccO0X_Dr+fP8eHmkeTVwergLDaP+kjAbQ0n_p24x-NdXY1tQ@mail.gmail.com>
	<CAHgk4BVujRwVT55BfLLo8kcLU76Jr09hdQua01dvYYuL7tuVeg@mail.gmail.com>
	<CAHgk4BUd9cB9wzWATSg06MwzdQs8+uS+LpdDQN5SkYhTA6pg_A@mail.gmail.com>
Message-ID: <520D9E43.9000704@xtronics.com>

I am thinking of getting a couple of SIP phones for our system - I see some used on ebay.

Are there ones to avoid? Good ones to seek?

I have a couple of places where there needs to be a backup phone in case the workstation is 
rebooting. I can set up POE if that sounds like a good idea.

Depending on what happens, I could then start retiring some of the analog phones replacing with SIP 
phones.

The bits I'm finding on the web is almost all by someone selling and mostly spammy information. A 
few clues could go a long way.




--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

Never ruin an apology with an excuse.
--------------------------------------------------------------------------------


From bdfoster at davri.com  Fri Aug 16 08:32:34 2013
From: bdfoster at davri.com (Brian Foster)
Date: Fri, 16 Aug 2013 00:32:34 -0400
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <520D9E43.9000704@xtronics.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
	<CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>
	<CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>
	<CAHccO0X_Dr+fP8eHmkeTVwergLDaP+kjAbQ0n_p24x-NdXY1tQ@mail.gmail.com>
	<CAHgk4BVujRwVT55BfLLo8kcLU76Jr09hdQua01dvYYuL7tuVeg@mail.gmail.com>
	<CAHgk4BUd9cB9wzWATSg06MwzdQs8+uS+LpdDQN5SkYhTA6pg_A@mail.gmail.com>
	<520D9E43.9000704@xtronics.com>
Message-ID: <CAHgk4BWDh6-Rjx-joxAPdAwOUAypu2zUv+QtLWsK9UidjAKEKA@mail.gmail.com>

I like Polycoms. For your use case, an IP 33X would work well. We have IP
335's but any in that category would do.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 15, 2013 11:42 PM, "Karl Schmidt" <karl at xtronics.com> wrote:

> I am thinking of getting a couple of SIP phones for our system - I see
> some used on ebay.
>
> Are there ones to avoid? Good ones to seek?
>
> I have a couple of places where there needs to be a backup phone in case
> the workstation is
> rebooting. I can set up POE if that sounds like a good idea.
>
> Depending on what happens, I could then start retiring some of the analog
> phones replacing with SIP
> phones.
>
> The bits I'm finding on the web is almost all by someone selling and
> mostly spammy information. A
> few clues could go a long way.
>
>
>
>
>
> --------------------------------------------------------------------------------
> Karl Schmidt                                  EMail Karl at xtronics.com
> Transtronics, Inc.                              WEB
> http://secure.transtronics.com
> 3209 West 9th Street                             Ph (785) 841-3089
> Lawrence, KS 66049                              FAX (785) 841-0434
>
> Never ruin an apology with an excuse.
>
> --------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Fri Aug 16 09:54:15 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Fri, 16 Aug 2013 11:24:15 +0530
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CABkS=xorE-TTwdKb=m0pO-C4stk4JC5KBJGcpbrhpFhdeMYKbQ@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
	<CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
	<CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
	<CALn233ovuLR8rgD==bqiOi+Fa=yDwS8azsrD3cWkBz9sz7Mw7w@mail.gmail.com>
	<CAMvhjJut4W_hwnAG45o2wFWCOKknwPzE8vFGrkNv6JoZZA8o1Q@mail.gmail.com>
	<CABkS=xorE-TTwdKb=m0pO-C4stk4JC5KBJGcpbrhpFhdeMYKbQ@mail.gmail.com>
Message-ID: <CAMvhjJs_YqJB6q11ADhKiPY-OdAiGZ9euzjja45VoXtwcycBXw@mail.gmail.com>

Any recommendations for a softphone that works on ubuntu linux. As xlite
will not work on linux.
Thanks
Ashish Mishra
On Aug 16, 2013 4:55 AM, "Peter" <eidevm5 at gmail.com> wrote:

> Of course you can use a softphone on the freeswitch server.   It would
> eliminate any network issues.
>
>
> On Thu, Aug 15, 2013 at 7:12 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> I also think that there may be some network problem as my windows pc is
>> not showing that some linux machine is connected but the linux machine
>> shows the windows system connected to it. Also can i install the softphone
>> on the same linux machine on which my freeswitch runs and can check whether
>> it gets connected or not...???
>>  On Aug 15, 2013 5:10 AM, "Mitch Capper" <mitch.capper at gmail.com> wrote:
>>
>>> You may want to try FSClient its a freeswitch based windows soft phone
>>> and you can connect fs_cli (or fs_logger.pl) to it and see what happens
>>> when it tries to connect to your server.
>>>
>>> ~mitch
>>>
>>>
>>> On Wed, Aug 14, 2013 at 4:06 PM, Peter <eidevm5 at gmail.com> wrote:
>>>
>>>> My mistake, I meant loglevel rather than logging.
>>>>
>>>> So when you say "no progress" are you seeing nothing at all on the FS
>>>> console?
>>>>
>>>> If so, then that indicates a network problem as FS isn't even seeing
>>>> the registration requests.
>>>>
>>>> Can you ping the windows box from the FS server?
>>>>
>>>>
>>>> On Wed, Aug 14, 2013 at 7:04 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>
>>>>> I went for "sofia loglevel 5" instead of "sofia logging 5" but still
>>>>> no progress in connection of softphone...:-(
>>>>> Thanks
>>>>> Ashish Mishra
>>>>> On Aug 14, 2013 2:14 PM, "Ashish Mishra" <itsme.kunnu at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Also the command that you had mentioned "sofia logging all 5" gives
>>>>>> me an error message :
>>>>>> Unknown command [logging]
>>>>>> On Aug 14, 2013 7:24 AM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>>>
>>>>>>> Connect to the FS console with fs_cli and do
>>>>>>>
>>>>>>> sofia logging all 5
>>>>>>>
>>>>>>> then try to register your softphone.    If you don't even see a
>>>>>>> registration attempt on the FS console, it means you have a network problem.
>>>>>>>
>>>>>>> If the network isn't a problem, the debug output should give you a
>>>>>>> clue as to what the problem is.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra <
>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>
>>>>>>>> Thank you Peter...you mean to say that i should first turn on the
>>>>>>>> fs_cli and then retry to connect the softphone...??? I would also like to
>>>>>>>> remind you that fs_cli and freeswitch are running on the same machine and i
>>>>>>>> am trying to connect thru a network cable my windows 8 pc that has the
>>>>>>>> softphone on it...
>>>>>>>> Regards
>>>>>>>> Ashish Mishra
>>>>>>>> On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Have you turned on debugging from the FS cli and seen if any
>>>>>>>>> registration requests come in?
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <
>>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> I installed freeswitch on my ubuntu 12.04 machine...but when i am
>>>>>>>>>> trying to connect the softphone installed on my windows 8 pc with ubuntu
>>>>>>>>>> machine the softphone gives me an error that account not enabled...i have
>>>>>>>>>> used a network cable to connect the two machines...also the firewall in
>>>>>>>>>> both the machines is disabled...
>>>>>>>>>> Kindly help..
>>>>>>>>>>
>>>>>>>>>> Ashish Mishra
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nandy1925 at gmail.com  Fri Aug 16 10:01:23 2013
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 16 Aug 2013 14:01:23 +0800
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CAMvhjJut4W_hwnAG45o2wFWCOKknwPzE8vFGrkNv6JoZZA8o1Q@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
	<CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
	<CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
	<CALn233ovuLR8rgD==bqiOi+Fa=yDwS8azsrD3cWkBz9sz7Mw7w@mail.gmail.com>
	<CAMvhjJut4W_hwnAG45o2wFWCOKknwPzE8vFGrkNv6JoZZA8o1Q@mail.gmail.com>
Message-ID: <CAHKqMb5Mb458YaWg01QLs3VxUP+rt-1ZeSy+0v5Sx3QCa_Cn9g@mail.gmail.com>

How about checking firewall or ACL rules that might be blocking your
softphone PC?

/Nandy

On Thu, Aug 15, 2013 at 5:12 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> I also think that there may be some network problem as my windows pc is
> not showing that some linux machine is connected but the linux machine
> shows the windows system connected to it. Also can i install the softphone
> on the same linux machine on which my freeswitch runs and can check whether
> it gets connected or not...???
>  On Aug 15, 2013 5:10 AM, "Mitch Capper" <mitch.capper at gmail.com> wrote:
>
>> You may want to try FSClient its a freeswitch based windows soft phone
>> and you can connect fs_cli (or fs_logger.pl) to it and see what happens
>> when it tries to connect to your server.
>>
>> ~mitch
>>
>>
>> On Wed, Aug 14, 2013 at 4:06 PM, Peter <eidevm5 at gmail.com> wrote:
>>
>>> My mistake, I meant loglevel rather than logging.
>>>
>>> So when you say "no progress" are you seeing nothing at all on the FS
>>> console?
>>>
>>> If so, then that indicates a network problem as FS isn't even seeing the
>>> registration requests.
>>>
>>> Can you ping the windows box from the FS server?
>>>
>>>
>>> On Wed, Aug 14, 2013 at 7:04 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> I went for "sofia loglevel 5" instead of "sofia logging 5" but still no
>>>> progress in connection of softphone...:-(
>>>> Thanks
>>>> Ashish Mishra
>>>> On Aug 14, 2013 2:14 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>>>>
>>>>> Also the command that you had mentioned "sofia logging all 5" gives me
>>>>> an error message :
>>>>> Unknown command [logging]
>>>>> On Aug 14, 2013 7:24 AM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>>
>>>>>> Connect to the FS console with fs_cli and do
>>>>>>
>>>>>> sofia logging all 5
>>>>>>
>>>>>> then try to register your softphone.    If you don't even see a
>>>>>> registration attempt on the FS console, it means you have a network problem.
>>>>>>
>>>>>> If the network isn't a problem, the debug output should give you a
>>>>>> clue as to what the problem is.
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra <itsme.kunnu at gmail.com
>>>>>> > wrote:
>>>>>>
>>>>>>> Thank you Peter...you mean to say that i should first turn on the
>>>>>>> fs_cli and then retry to connect the softphone...??? I would also like to
>>>>>>> remind you that fs_cli and freeswitch are running on the same machine and i
>>>>>>> am trying to connect thru a network cable my windows 8 pc that has the
>>>>>>> softphone on it...
>>>>>>> Regards
>>>>>>> Ashish Mishra
>>>>>>> On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>>>>
>>>>>>>> Have you turned on debugging from the FS cli and seen if any
>>>>>>>> registration requests come in?
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <
>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> I installed freeswitch on my ubuntu 12.04 machine...but when i am
>>>>>>>>> trying to connect the softphone installed on my windows 8 pc with ubuntu
>>>>>>>>> machine the softphone gives me an error that account not enabled...i have
>>>>>>>>> used a network cable to connect the two machines...also the firewall in
>>>>>>>>> both the machines is disabled...
>>>>>>>>> Kindly help..
>>>>>>>>>
>>>>>>>>> Ashish Mishra
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Fri Aug 16 10:09:59 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Fri, 16 Aug 2013 11:39:59 +0530
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CAHKqMb5Mb458YaWg01QLs3VxUP+rt-1ZeSy+0v5Sx3QCa_Cn9g@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
	<CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
	<CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
	<CALn233ovuLR8rgD==bqiOi+Fa=yDwS8azsrD3cWkBz9sz7Mw7w@mail.gmail.com>
	<CAMvhjJut4W_hwnAG45o2wFWCOKknwPzE8vFGrkNv6JoZZA8o1Q@mail.gmail.com>
	<CAHKqMb5Mb458YaWg01QLs3VxUP+rt-1ZeSy+0v5Sx3QCa_Cn9g@mail.gmail.com>
Message-ID: <CAMvhjJtXq2Rxjs2p99VACbF_2ORuGAsnjYdO6jWA5toRctGjEg@mail.gmail.com>

Thank you. I have already flushed the firewall rules of ubuntu linux and
disabled the firewall of windows pc as well. But have not checked the acl
rules ? Can you tell me how to do so ?
On Aug 16, 2013 11:35 AM, "Nandy Dagondon" <nandy1925 at gmail.com> wrote:

> How about checking firewall or ACL rules that might be blocking your
> softphone PC?
>
> /Nandy
>
> On Thu, Aug 15, 2013 at 5:12 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> I also think that there may be some network problem as my windows pc is
>> not showing that some linux machine is connected but the linux machine
>> shows the windows system connected to it. Also can i install the softphone
>> on the same linux machine on which my freeswitch runs and can check whether
>> it gets connected or not...???
>>  On Aug 15, 2013 5:10 AM, "Mitch Capper" <mitch.capper at gmail.com> wrote:
>>
>>> You may want to try FSClient its a freeswitch based windows soft phone
>>> and you can connect fs_cli (or fs_logger.pl) to it and see what happens
>>> when it tries to connect to your server.
>>>
>>> ~mitch
>>>
>>>
>>> On Wed, Aug 14, 2013 at 4:06 PM, Peter <eidevm5 at gmail.com> wrote:
>>>
>>>> My mistake, I meant loglevel rather than logging.
>>>>
>>>> So when you say "no progress" are you seeing nothing at all on the FS
>>>> console?
>>>>
>>>> If so, then that indicates a network problem as FS isn't even seeing
>>>> the registration requests.
>>>>
>>>> Can you ping the windows box from the FS server?
>>>>
>>>>
>>>> On Wed, Aug 14, 2013 at 7:04 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>
>>>>> I went for "sofia loglevel 5" instead of "sofia logging 5" but still
>>>>> no progress in connection of softphone...:-(
>>>>> Thanks
>>>>> Ashish Mishra
>>>>> On Aug 14, 2013 2:14 PM, "Ashish Mishra" <itsme.kunnu at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Also the command that you had mentioned "sofia logging all 5" gives
>>>>>> me an error message :
>>>>>> Unknown command [logging]
>>>>>> On Aug 14, 2013 7:24 AM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>>>
>>>>>>> Connect to the FS console with fs_cli and do
>>>>>>>
>>>>>>> sofia logging all 5
>>>>>>>
>>>>>>> then try to register your softphone.    If you don't even see a
>>>>>>> registration attempt on the FS console, it means you have a network problem.
>>>>>>>
>>>>>>> If the network isn't a problem, the debug output should give you a
>>>>>>> clue as to what the problem is.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra <
>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>
>>>>>>>> Thank you Peter...you mean to say that i should first turn on the
>>>>>>>> fs_cli and then retry to connect the softphone...??? I would also like to
>>>>>>>> remind you that fs_cli and freeswitch are running on the same machine and i
>>>>>>>> am trying to connect thru a network cable my windows 8 pc that has the
>>>>>>>> softphone on it...
>>>>>>>> Regards
>>>>>>>> Ashish Mishra
>>>>>>>> On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Have you turned on debugging from the FS cli and seen if any
>>>>>>>>> registration requests come in?
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <
>>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> I installed freeswitch on my ubuntu 12.04 machine...but when i am
>>>>>>>>>> trying to connect the softphone installed on my windows 8 pc with ubuntu
>>>>>>>>>> machine the softphone gives me an error that account not enabled...i have
>>>>>>>>>> used a network cable to connect the two machines...also the firewall in
>>>>>>>>>> both the machines is disabled...
>>>>>>>>>> Kindly help..
>>>>>>>>>>
>>>>>>>>>> Ashish Mishra
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nandy1925 at gmail.com  Fri Aug 16 10:27:32 2013
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 16 Aug 2013 14:27:32 +0800
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CAMvhjJtXq2Rxjs2p99VACbF_2ORuGAsnjYdO6jWA5toRctGjEg@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
	<CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
	<CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
	<CALn233ovuLR8rgD==bqiOi+Fa=yDwS8azsrD3cWkBz9sz7Mw7w@mail.gmail.com>
	<CAMvhjJut4W_hwnAG45o2wFWCOKknwPzE8vFGrkNv6JoZZA8o1Q@mail.gmail.com>
	<CAHKqMb5Mb458YaWg01QLs3VxUP+rt-1ZeSy+0v5Sx3QCa_Cn9g@mail.gmail.com>
	<CAMvhjJtXq2Rxjs2p99VACbF_2ORuGAsnjYdO6jWA5toRctGjEg@mail.gmail.com>
Message-ID: <CAHKqMb4xM0dHP=zUP5qhjK-XJgFiu0P5Ou7QaXimHBDjwCS+tw@mail.gmail.com>

I'm not well-versed on ACL but here's the link
http://wiki.freeswitch.org/wiki/ACL

/Nandy

On Fri, Aug 16, 2013 at 2:09 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Thank you. I have already flushed the firewall rules of ubuntu linux and
> disabled the firewall of windows pc as well. But have not checked the acl
> rules ? Can you tell me how to do so ?
> On Aug 16, 2013 11:35 AM, "Nandy Dagondon" <nandy1925 at gmail.com> wrote:
>
>> How about checking firewall or ACL rules that might be blocking your
>> softphone PC?
>>
>> /Nandy
>>
>> On Thu, Aug 15, 2013 at 5:12 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> I also think that there may be some network problem as my windows pc is
>>> not showing that some linux machine is connected but the linux machine
>>> shows the windows system connected to it. Also can i install the softphone
>>> on the same linux machine on which my freeswitch runs and can check whether
>>> it gets connected or not...???
>>>  On Aug 15, 2013 5:10 AM, "Mitch Capper" <mitch.capper at gmail.com> wrote:
>>>
>>>> You may want to try FSClient its a freeswitch based windows soft phone
>>>> and you can connect fs_cli (or fs_logger.pl) to it and see what
>>>> happens when it tries to connect to your server.
>>>>
>>>> ~mitch
>>>>
>>>>
>>>> On Wed, Aug 14, 2013 at 4:06 PM, Peter <eidevm5 at gmail.com> wrote:
>>>>
>>>>> My mistake, I meant loglevel rather than logging.
>>>>>
>>>>> So when you say "no progress" are you seeing nothing at all on the FS
>>>>> console?
>>>>>
>>>>> If so, then that indicates a network problem as FS isn't even seeing
>>>>> the registration requests.
>>>>>
>>>>> Can you ping the windows box from the FS server?
>>>>>
>>>>>
>>>>> On Wed, Aug 14, 2013 at 7:04 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>>
>>>>>> I went for "sofia loglevel 5" instead of "sofia logging 5" but still
>>>>>> no progress in connection of softphone...:-(
>>>>>> Thanks
>>>>>> Ashish Mishra
>>>>>> On Aug 14, 2013 2:14 PM, "Ashish Mishra" <itsme.kunnu at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Also the command that you had mentioned "sofia logging all 5" gives
>>>>>>> me an error message :
>>>>>>> Unknown command [logging]
>>>>>>> On Aug 14, 2013 7:24 AM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>>>>
>>>>>>>> Connect to the FS console with fs_cli and do
>>>>>>>>
>>>>>>>> sofia logging all 5
>>>>>>>>
>>>>>>>> then try to register your softphone.    If you don't even see a
>>>>>>>> registration attempt on the FS console, it means you have a network problem.
>>>>>>>>
>>>>>>>> If the network isn't a problem, the debug output should give you a
>>>>>>>> clue as to what the problem is.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra <
>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Thank you Peter...you mean to say that i should first turn on the
>>>>>>>>> fs_cli and then retry to connect the softphone...??? I would also like to
>>>>>>>>> remind you that fs_cli and freeswitch are running on the same machine and i
>>>>>>>>> am trying to connect thru a network cable my windows 8 pc that has the
>>>>>>>>> softphone on it...
>>>>>>>>> Regards
>>>>>>>>> Ashish Mishra
>>>>>>>>> On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Have you turned on debugging from the FS cli and seen if any
>>>>>>>>>> registration requests come in?
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <
>>>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> I installed freeswitch on my ubuntu 12.04 machine...but when i
>>>>>>>>>>> am trying to connect the softphone installed on my windows 8 pc with ubuntu
>>>>>>>>>>> machine the softphone gives me an error that account not enabled...i have
>>>>>>>>>>> used a network cable to connect the two machines...also the firewall in
>>>>>>>>>>> both the machines is disabled...
>>>>>>>>>>> Kindly help..
>>>>>>>>>>>
>>>>>>>>>>> Ashish Mishra
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>>
>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ssinyagin at yahoo.com  Fri Aug 16 11:22:52 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Fri, 16 Aug 2013 00:22:52 -0700 (PDT)
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <520D9E43.9000704@xtronics.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>	<CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>	<CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>	<CAHccO0X_Dr+fP8eHmkeTVwergLDaP+kjAbQ0n_p24x-NdXY1tQ@mail.gmail.com>	<CAHgk4BVujRwVT55BfLLo8kcLU76Jr09hdQua01dvYYuL7tuVeg@mail.gmail.com>	<CAHgk4BUd9cB9wzWATSg06MwzdQs8+uS+LpdDQN5SkYhTA6pg_A@mail.gmail.com>
	<520D9E43.9000704@xtronics.com>
Message-ID: <1376637772.65677.YahooMailNeo@web126202.mail.ne1.yahoo.com>

Gigaset 610IP is a great cordless ip-phone. Works perfectly with FreeSWITCH. You can also have several handsets served from one base.






>________________________________
> From: Karl Schmidt <karl at xtronics.com>
>To: freeswitch-users at lists.freeswitch.org 
>Sent: Friday, August 16, 2013 5:36 AM
>Subject: [Freeswitch-users] SIP phones -
> 
>
>I am thinking of getting a couple of SIP phones for our system - I see some used on ebay.
>
>Are there ones to avoid? Good ones to seek?
>
>I have a couple of places where there needs to be a backup phone in case the workstation is 
>rebooting. I can set up POE if that sounds like a good idea.
>
>Depending on what happens, I could then start retiring some of the analog phones replacing with SIP 
>phones.
>
>The bits I'm finding on the web is almost all by someone selling and mostly spammy information. A 
>few clues could go a long way.
>
>
>
>
>--------------------------------------------------------------------------------
>Karl Schmidt? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? EMail Karl at xtronics.com
>Transtronics, Inc.? ? ? ? ? ? ? ? ? ? ? ? ? ? ? WEB http://secure.transtronics.com
>3209 West 9th Street? ? ? ? ? ? ? ? ? ? ? ? ? ?  Ph (785) 841-3089
>Lawrence, KS 66049? ? ? ? ? ? ? ? ? ? ? ? ? ? ? FAX (785) 841-0434
>
>Never ruin an apology with an excuse.
>--------------------------------------------------------------------------------
>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 at cassidywebservices.co.uk  Fri Aug 16 13:15:34 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Fri, 16 Aug 2013 10:15:34 +0100
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <1376637772.65677.YahooMailNeo@web126202.mail.ne1.yahoo.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
	<CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>
	<CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>
	<CAHccO0X_Dr+fP8eHmkeTVwergLDaP+kjAbQ0n_p24x-NdXY1tQ@mail.gmail.com>
	<CAHgk4BVujRwVT55BfLLo8kcLU76Jr09hdQua01dvYYuL7tuVeg@mail.gmail.com>
	<CAHgk4BUd9cB9wzWATSg06MwzdQs8+uS+LpdDQN5SkYhTA6pg_A@mail.gmail.com>
	<520D9E43.9000704@xtronics.com>
	<1376637772.65677.YahooMailNeo@web126202.mail.ne1.yahoo.com>
Message-ID: <CAMbW9oLwPzPAraod5AmsGP2b6h4zHR4YxTwZYUjszixXtXWRFw@mail.gmail.com>

Polycom is the name I see the most on this list, I've some some experience
with Snom and Grandstream. The do the job but my favourites are Cisco
SPA50x series.


On 16 August 2013 08:22, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:

> Gigaset 610IP is a great cordless ip-phone. Works perfectly with
> FreeSWITCH. You can also have several handsets served from one base.
>
>
>
>   ------------------------------
>  *From:* Karl Schmidt <karl at xtronics.com>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Friday, August 16, 2013 5:36 AM
> *Subject:* [Freeswitch-users] SIP phones -
>
> I am thinking of getting a couple of SIP phones for our system - I see
> some used on ebay.
>
> Are there ones to avoid? Good ones to seek?
>
> I have a couple of places where there needs to be a backup phone in case
> the workstation is
> rebooting. I can set up POE if that sounds like a good idea.
>
> Depending on what happens, I could then start retiring some of the analog
> phones replacing with SIP
> phones.
>
> The bits I'm finding on the web is almost all by someone selling and
> mostly spammy information. A
> few clues could go a long way.
>
>
>
>
>
> --------------------------------------------------------------------------------
> Karl Schmidt                                  EMail Karl at xtronics.com
> Transtronics, Inc.                              WEB
> http://secure.transtronics.com
> 3209 West 9th Street                            Ph (785) 841-3089
> Lawrence, KS 66049                              FAX (785) 841-0434
>
> Never ruin an apology with an excuse.
>
> --------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From lconroy at insensate.co.uk  Fri Aug 16 14:35:00 2013
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Fri, 16 Aug 2013 11:35:00 +0100
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <CAMbW9oLwPzPAraod5AmsGP2b6h4zHR4YxTwZYUjszixXtXWRFw@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
	<CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>
	<CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>
	<CAHccO0X_Dr+fP8eHmkeTVwergLDaP+kjAbQ0n_p24x-NdXY1tQ@mail.gmail.com>
	<CAHgk4BVujRwVT55BfLLo8kcLU76Jr09hdQua01dvYYuL7tuVeg@mail.gmail.com>
	<CAHgk4BUd9cB9wzWATSg06MwzdQs8+uS+LpdDQN5SkYhTA6pg_A@mail.gmail.com>
	<520D9E43.9000704@xtronics.com>
	<1376637772.65677.YahooMailNeo@web126202.mail.ne1.yahoo.com>
	<CAMbW9oLwPzPAraod5AmsGP2b6h4zHR4YxTwZYUjszixXtXWRFw@mail.gmail.com>
Message-ID: <DFC40D59-FD0E-46B1-942B-EA741412D6A3@insensate.co.uk>

Hi Folks,
 +3 for the Gigasets. I have a number of these setups -- with different handsets.
These are working fine with fS in all of my servers.
TL;DR: I paid my own money for two of these for my own use. My wife doesn't know
 she's using SIP for all calls -- it's just a 'phone.

Voice Quality seems to be good, range is excellent, battery life is fine (though
they spend most of their time in the desk stands, charging).
I have not used bluetooth headsets with them, so can't talk on that. The 2.5 mm
audio connector works fine (some folk prefer clipping the phone to their waist
and wearing a headset -- YMMV).
The N300A (consumer DECT SIP base station) voicemail is OK, but IMHO over-compresses.
The in-built web server by which the base is configured is OK if slow.
Configuring the base via a connected handset is possible, but like every manufacturer's
version of that I've seen, it's not recommended for mere mortals without a lot of patience.
[...and apart from the annoyance of the battery compartment door falling open if you
drop them, and the handset NOT being waterproof/toiletproof, they're remarkably robust]

As for Polycom, can't speak for the "straight" SIP versions, but I have experience
 of their Lync DECT multicell system & handsets; IMHO these don't even reach the
 'blows goats' level. Work has 50 of them; user feedback is that voice quality,
 coverage, battery life is "not as good as we might hope".
Opinion seems to be that these are a step back from the siemens cellular DECT
'phones we had on the old "steam" PBX (being polite).
Sigh.  For all portable systems, Try before you buy.

all the best,
 Lawrence


On 16 Aug 2013, at 10:15, Andrew Cassidy wrote:
> Polycom is the name I see the most on this list, I've some some experience
> with Snom and Grandstream. The do the job but my favourites are Cisco
> SPA50x series.
> 
> 
> On 16 August 2013 08:22, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
> 
>> Gigaset 610IP is a great cordless ip-phone. Works perfectly with
>> FreeSWITCH. You can also have several handsets served from one base.
>> 
>> 
>> 
>>  ------------------------------
>> *From:* Karl Schmidt <karl at xtronics.com>
>> *To:* freeswitch-users at lists.freeswitch.org
>> *Sent:* Friday, August 16, 2013 5:36 AM
>> *Subject:* [Freeswitch-users] SIP phones -
>> 
>> I am thinking of getting a couple of SIP phones for our system - I see
>> some used on ebay.
>> 
>> Are there ones to avoid? Good ones to seek?
>> 
>> I have a couple of places where there needs to be a backup phone in case
>> the workstation is
>> rebooting. I can set up POE if that sounds like a good idea.
>> 
>> Depending on what happens, I could then start retiring some of the analog
>> phones replacing with SIP
>> phones.
>> 
>> The bits I'm finding on the web is almost all by someone selling and
>> mostly spammy information. A
>> few clues could go a long way.
>> 
>> 
>> 
>> 
>> 
>> --------------------------------------------------------------------------------
>> Karl Schmidt                                  EMail Karl at xtronics.com
>> Transtronics, Inc.                              WEB
>> http://secure.transtronics.com
>> 3209 West 9th Street                            Ph (785) 841-3089
>> Lawrence, KS 66049                              FAX (785) 841-0434
>> 
>> Never ruin an apology with an excuse.
>> 
>> --------------------------------------------------------------------------------
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>> 
>> 
> 
> 
> -- 
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
> 
> 
> *T <info at cassidywebservices.co.uk> *03300 100 960
> *F<info at cassidywebservices.co.uk>
> *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 GB at cm.nl  Fri Aug 16 15:28:01 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Fri, 16 Aug 2013 13:28:01 +0200
Subject: [Freeswitch-users] Public and default context question
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD1957F@clubvirtual40.ClubMessage.local>

Hello,

I just started to play around with FreeSwitch, and I have a couple of questions.

Is the default context only used for devices which are registered with FreeSwitch?
If I have a Carrier which communicates with FS through a public IP and a Kamailio SIP Proxy which communicates with FS through a private IP address, both will use the public context right?

My goal is to configure FS as a SBC, and I see there is a sbc directory in the ../conf folder, but when I change the pre-process value to point to sbc/freeswitch.xml, I'm not able to start freeswitch anymore.
<X-PRE-PROCESS cmd="include" data="sbc/freeswitch.xml"/>

There are no devices(IP Phones for example) which will register with FreeSwitch, so basically I won't be needing the default context, correct?

Regards,

Grant
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From peter at onemetric.com  Fri Aug 16 16:25:46 2013
From: peter at onemetric.com (Peter Blackford)
Date: Fri, 16 Aug 2013 22:25:46 +1000
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <DFC40D59-FD0E-46B1-942B-EA741412D6A3@insensate.co.uk>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
	<CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>
	<CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>
	<CAHccO0X_Dr+fP8eHmkeTVwergLDaP+kjAbQ0n_p24x-NdXY1tQ@mail.gmail.com>
	<CAHgk4BVujRwVT55BfLLo8kcLU76Jr09hdQua01dvYYuL7tuVeg@mail.gmail.com>
	<CAHgk4BUd9cB9wzWATSg06MwzdQs8+uS+LpdDQN5SkYhTA6pg_A@mail.gmail.com>
	<520D9E43.9000704@xtronics.com>
	<1376637772.65677.YahooMailNeo@web126202.mail.ne1.yahoo.com>
	<CAMbW9oLwPzPAraod5AmsGP2b6h4zHR4YxTwZYUjszixXtXWRFw@mail.gmail.com>
	<DFC40D59-FD0E-46B1-942B-EA741412D6A3@insensate.co.uk>
Message-ID: <CAG1aRi89yoXcHipVJYBFzg6YzZfLKiUOVbGJGsfA1Qdd6ALi0w@mail.gmail.com>

I HATE Polycoms they are hard to configure and hard to debug.

I really like the Cisco SPA series as they are reliable however also hard
to debug. Snoms and Yealinks are good and a lot easier to debug (Pulling
packet captures directly from the phone makes life a lot easier)

Just my 2c.


On 16 August 2013 20:35, Lawrence Conroy <lconroy at insensate.co.uk> wrote:

> Hi Folks,
>  +3 for the Gigasets. I have a number of these setups -- with different
> handsets.
> These are working fine with fS in all of my servers.
> TL;DR: I paid my own money for two of these for my own use. My wife
> doesn't know
>  she's using SIP for all calls -- it's just a 'phone.
>
> Voice Quality seems to be good, range is excellent, battery life is fine
> (though
> they spend most of their time in the desk stands, charging).
> I have not used bluetooth headsets with them, so can't talk on that. The
> 2.5 mm
> audio connector works fine (some folk prefer clipping the phone to their
> waist
> and wearing a headset -- YMMV).
> The N300A (consumer DECT SIP base station) voicemail is OK, but IMHO
> over-compresses.
> The in-built web server by which the base is configured is OK if slow.
> Configuring the base via a connected handset is possible, but like every
> manufacturer's
> version of that I've seen, it's not recommended for mere mortals without a
> lot of patience.
> [...and apart from the annoyance of the battery compartment door falling
> open if you
> drop them, and the handset NOT being waterproof/toiletproof, they're
> remarkably robust]
>
> As for Polycom, can't speak for the "straight" SIP versions, but I have
> experience
>  of their Lync DECT multicell system & handsets; IMHO these don't even
> reach the
>  'blows goats' level. Work has 50 of them; user feedback is that voice
> quality,
>  coverage, battery life is "not as good as we might hope".
> Opinion seems to be that these are a step back from the siemens cellular
> DECT
> 'phones we had on the old "steam" PBX (being polite).
> Sigh.  For all portable systems, Try before you buy.
>
> all the best,
>  Lawrence
>
>
> On 16 Aug 2013, at 10:15, Andrew Cassidy wrote:
> > Polycom is the name I see the most on this list, I've some some
> experience
> > with Snom and Grandstream. The do the job but my favourites are Cisco
> > SPA50x series.
> >
> >
> > On 16 August 2013 08:22, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
> >
> >> Gigaset 610IP is a great cordless ip-phone. Works perfectly with
> >> FreeSWITCH. You can also have several handsets served from one base.
> >>
> >>
> >>
> >>  ------------------------------
> >> *From:* Karl Schmidt <karl at xtronics.com>
> >> *To:* freeswitch-users at lists.freeswitch.org
> >> *Sent:* Friday, August 16, 2013 5:36 AM
> >> *Subject:* [Freeswitch-users] SIP phones -
> >>
> >> I am thinking of getting a couple of SIP phones for our system - I see
> >> some used on ebay.
> >>
> >> Are there ones to avoid? Good ones to seek?
> >>
> >> I have a couple of places where there needs to be a backup phone in case
> >> the workstation is
> >> rebooting. I can set up POE if that sounds like a good idea.
> >>
> >> Depending on what happens, I could then start retiring some of the
> analog
> >> phones replacing with SIP
> >> phones.
> >>
> >> The bits I'm finding on the web is almost all by someone selling and
> >> mostly spammy information. A
> >> few clues could go a long way.
> >>
> >>
> >>
> >>
> >>
> >>
> --------------------------------------------------------------------------------
> >> Karl Schmidt                                  EMail Karl at xtronics.com
> >> Transtronics, Inc.                              WEB
> >> http://secure.transtronics.com
> >> 3209 West 9th Street                            Ph (785) 841-3089
> >> Lawrence, KS 66049                              FAX (785) 841-0434
> >>
> >> Never ruin an apology with an excuse.
> >>
> >>
> --------------------------------------------------------------------------------
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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
> >>
> >>
> >
> >
> > --
> > *Andrew Cassidy BSc (Hons) MBCS SSCA*
> > Managing Director
> >
> >
> > *T <info at cassidywebservices.co.uk> *03300 100 960
> > *F<info at cassidywebservices.co.uk>
> > *03300 100 961
> > *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> > *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Fri Aug 16 16:34:54 2013
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 16 Aug 2013 07:34:54 -0500
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <CAG1aRi89yoXcHipVJYBFzg6YzZfLKiUOVbGJGsfA1Qdd6ALi0w@mail.gmail.com>
Message-ID: <CE33869E.78FAE%krice@freeswitch.org>

I?ve never found polycoms hard to debug or configure, you just need to set
up a proper boot server for them... Once you do that, the configs are
straight forward, and it?ll actually logs back to that server...


On 8/16/13 7:25 AM, "Peter Blackford" <peter at onemetric.com> wrote:

> I HATE Polycoms they are hard to configure and hard to debug.
> 
> I really like the Cisco SPA series as they are reliable however also hard to
> debug. Snoms and Yealinks are good and a lot easier to debug (Pulling packet
> captures directly from the phone makes life a lot easier)
> 
> Just my 2c.
> 
> 
> On 16 August 2013 20:35, Lawrence Conroy <lconroy at insensate.co.uk> wrote:
>> Hi Folks,
>> ?+3 for the Gigasets. I have a number of these setups -- with different
>> handsets.
>> These are working fine with fS in all of my servers.
>> TL;DR: I paid my own money for two of these for my own use. My wife doesn't
>> know
>> ?she's using SIP for all calls -- it's just a 'phone.
>> 
>> Voice Quality seems to be good, range is excellent, battery life is fine
>> (though
>> they spend most of their time in the desk stands, charging).
>> I have not used bluetooth headsets with them, so can't talk on that. The 2.5
>> mm
>> audio connector works fine (some folk prefer clipping the phone to their
>> waist
>> and wearing a headset -- YMMV).
>> The N300A (consumer DECT SIP base station) voicemail is OK, but IMHO
>> over-compresses.
>> The in-built web server by which the base is configured is OK if slow.
>> Configuring the base via a connected handset is possible, but like every
>> manufacturer's
>> version of that I've seen, it's not recommended for mere mortals without a
>> lot of patience.
>> [...and apart from the annoyance of the battery compartment door falling open
>> if you
>> drop them, and the handset NOT being waterproof/toiletproof, they're
>> remarkably robust]
>> 
>> As for Polycom, can't speak for the "straight" SIP versions, but I have
>> experience
>> ?of their Lync DECT multicell system & handsets; IMHO these don't even reach
>> the
>> ?'blows goats' level. Work has 50 of them; user feedback is that voice
>> quality,
>> ?coverage, battery life is "not as good as we might hope".
>> Opinion seems to be that these are a step back from the siemens cellular DECT
>> 'phones we had on the old "steam" PBX (being polite).
>> Sigh. ?For all portable systems, Try before you buy.
>> 
>> all the best,
>> ?Lawrence
>> 
>> 
>> On 16 Aug 2013, at 10:15, Andrew Cassidy wrote:
>>> > Polycom is the name I see the most on this list, I've some some experience
>>> > with Snom and Grandstream. The do the job but my favourites are Cisco
>>> > SPA50x series.
>>> >
>>> >
>>> > On 16 August 2013 08:22, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>>> >
>>>> >> Gigaset 610IP is a great cordless ip-phone. Works perfectly with
>>>> >> FreeSWITCH. You can also have several handsets served from one base.
>>>> >>
>>>> >>
>>>> >>
>>>> >> ?------------------------------
>>>> >> *From:* Karl Schmidt <karl at xtronics.com>
>>>> >> *To:* freeswitch-users at lists.freeswitch.org
>>>> >> *Sent:* Friday, August 16, 2013 5:36 AM
>>>> >> *Subject:* [Freeswitch-users] SIP phones -
>>>> >>
>>>> >> I am thinking of getting a couple of SIP phones for our system - I see
>>>> >> some used on ebay.
>>>> >>
>>>> >> Are there ones to avoid? Good ones to seek?
>>>> >>
>>>> >> I have a couple of places where there needs to be a backup phone in case
>>>> >> the workstation is
>>>> >> rebooting. I can set up POE if that sounds like a good idea.
>>>> >>
>>>> >> Depending on what happens, I could then start retiring some of the
>>>> analog
>>>> >> phones replacing with SIP
>>>> >> phones.
>>>> >>
>>>> >> The bits I'm finding on the web is almost all by someone selling and
>>>> >> mostly spammy information. A
>>>> >> few clues could go a long way.
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >> 
>>>> ---------------------------------------------------------------------------
>>>> -----
>>>> >> Karl Schmidt ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?EMail Karl at xtronics.com
>>>> >> Transtronics, Inc. ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?WEB
>>>> >> http://secure.transtronics.com
>>>> >> 3209 West 9th Street ? ? ? ? ? ? ? ? ? ? ? ? ? ?Ph (785) 841-3089
>>>> <tel:%28785%29%20841-3089>
>>>> >> Lawrence, KS 66049 ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?FAX (785) 841-0434
>>>> <tel:%28785%29%20841-0434>
>>>> >>
>>>> >> Never ruin an apology with an excuse.
>>>> >>
>>>> >> 
>>>> ---------------------------------------------------------------------------
>>>> -----
>>>> >>
>>>> >> 
>>>> _________________________________________________________________________
>>>> >> Professional FreeSWITCH Consulting Services:
>>>> >> consulting at freeswitch.org
>>>> >> http://www.freeswitchsolutions.com
>>>> >>
>>>> >> 
>>>> >> 
>>>> >>
>>>> >> Official FreeSWITCH Sites
>>>> >> http://www.freeswitch.org
>>>> >> http://wiki.freeswitch.org
>>>> >> http://www.cluecon.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
>>>> >>
>>>> >>
>>>> >>
>>>> >> 
>>>> _________________________________________________________________________
>>>> >> Professional FreeSWITCH Consulting Services:
>>>> >> consulting at freeswitch.org
>>>> >> http://www.freeswitchsolutions.com
>>>> >>
>>>> >> 
>>>> >> 
>>>> >>
>>>> >> Official FreeSWITCH Sites
>>>> >> http://www.freeswitch.org
>>>> >> http://wiki.freeswitch.org
>>>> >> http://www.cluecon.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
>>>> >>
>>>> >>
>>> >
>>> >
>>> > --
>>> > *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>> > Managing Director
>>> >
>>> >
>>> > *T <info at cassidywebservices.co.uk> *03300 100 960
>>> > *F<info at cassidywebservices.co.uk>
>>> > *03300 100 961
>>> > *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>>> > *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>> <http://www.cassidywebservices.co.uk>
>>> > _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.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
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From max at nysolutions.com  Fri Aug 16 16:44:12 2013
From: max at nysolutions.com (Moishe Grunstein)
Date: Fri, 16 Aug 2013 12:44:12 +0000
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <CAHgk4BWDh6-Rjx-joxAPdAwOUAypu2zUv+QtLWsK9UidjAKEKA@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
	<CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>
	<CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>
	<CAHccO0X_Dr+fP8eHmkeTVwergLDaP+kjAbQ0n_p24x-NdXY1tQ@mail.gmail.com>
	<CAHgk4BVujRwVT55BfLLo8kcLU76Jr09hdQua01dvYYuL7tuVeg@mail.gmail.com>
	<CAHgk4BUd9cB9wzWATSg06MwzdQs8+uS+LpdDQN5SkYhTA6pg_A@mail.gmail.com>
	<520D9E43.9000704@xtronics.com>
	<CAHgk4BWDh6-Rjx-joxAPdAwOUAypu2zUv+QtLWsK9UidjAKEKA@mail.gmail.com>
Message-ID: <BA6A05D48E282C4C9989496521266BA10298775DB5@ex2.connex.local>

VVX 300/310 are low cost new models.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster
Sent: Friday, August 16, 2013 12:33 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] SIP phones -


I like Polycoms. For your use case, an IP 33X would work well. We have IP 335's but any in that category would do.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com<mailto:bdfoster at davri.com>
Indianapolis, Indiana

Sent from a mobile device.
On Aug 15, 2013 11:42 PM, "Karl Schmidt" <karl at xtronics.com<mailto:karl at xtronics.com>> wrote:
I am thinking of getting a couple of SIP phones for our system - I see some used on ebay.

Are there ones to avoid? Good ones to seek?

I have a couple of places where there needs to be a backup phone in case the workstation is
rebooting. I can set up POE if that sounds like a good idea.

Depending on what happens, I could then start retiring some of the analog phones replacing with SIP
phones.

The bits I'm finding on the web is almost all by someone selling and mostly spammy information. A
few clues could go a long way.




--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com<mailto:Karl at xtronics.com>
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089<tel:%28785%29%20841-3089>
Lawrence, KS 66049                              FAX (785) 841-0434<tel:%28785%29%20841-0434>

Never ruin an apology with an excuse.
--------------------------------------------------------------------------------

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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
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From sravi123 at yahoo.com  Fri Aug 16 16:51:00 2013
From: sravi123 at yahoo.com (Ravi S)
Date: Fri, 16 Aug 2013 18:21:00 +0530
Subject: [Freeswitch-users] Recording calls with Freeswitch - log says
 recorded, but no file found
In-Reply-To: <1376554347.89086.YahooMailNeo@web126204.mail.ne1.yahoo.com>
References: <1376487277.65226.YahooMailNeo@web160503.mail.bf1.yahoo.com>	<CAKzWOxWk9bUD4skaAO1cpYJC2kAiY6jK7+77DCQpBardNPsKcg@mail.gmail.com>
	<1376552695.54693.YahooMailNeo@web160505.mail.bf1.yahoo.com>
	<1376554347.89086.YahooMailNeo@web126204.mail.ne1.yahoo.com>
Message-ID: <520E2034.90608@yahoo.com>

Thanks Stanislav !
The $ sign was the culprit !! The issue is resolved and I am able to 
record my calls.

Thanks again !!
Ravi


On 15-08-2013 13:42, Stanislav Sinyagin wrote:
> is the dollar sign before {recordings_ dir} and {record_file_name} 
> eaten by your mail client, or it's really so in your XML?
>
>
>
>     ------------------------------------------------------------------------
>     *From:* Ravi <sravi123 at yahoo.com>
>     *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>     *Sent:* Thursday, August 15, 2013 9:44 AM
>     *Subject:* Re: [Freeswitch-users] Recording calls with Freeswitch
>     - log says recorded, but no file found
>
>     Thanks Michael.
>
>     I got the log with the "debug" level output. The pastebin link for
>     the log is :
>     http://pastebin.freeswitch.org/21301
>
>     The call is made from extension 1020 to 1030. The dialplan code
>     snippet is given below as well. If any other information is
>     required, please let me know.
>
>     <extension name="003">
>         <condition field="destination_number" expression="^003$">
>             <action application="set"
>     data="record_file_name={recordings_
>     dir}/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.wav" inli$
>             <action application="record_session"
>     data="{record_file_name}"/>
>             <action application="bridge" data="user/1003"/>
>         </condition>
>     </extension>
>
>
>     Please review and let me know where I am stuck, and what can be
>     done. Freeswitch crashed a couple of times, while I was testing. I
>     will mail that in a separate post.
>
>     Thanks again !
>     Ravi
>
>
>     ------------------------------------------------------------------------
>     *From:* Michael Collins <msc at freeswitch.org>
>     *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>     *Sent:* Wednesday, August 14, 2013 11:32 PM
>     *Subject:* Re: [Freeswitch-users] Recording calls with Freeswitch
>     - log says recorded, but no file found
>
>     You'll need debug level output. That looks like it's info level
>     output. The freeswitch.log file should have debug level output.
>     Also, if you're on the freeswitch console (not the fs_cli) then
>     info is the default level. Try "console loglevel debug" and then
>     retest and re-capture. Also, best place to put fs log data is:
>     pastebin.freeswitch.org <http://pastebin.freeswitch.org/>. Use
>     "FreeSWITCH Log" as the syntax highlighting.
>
>     -MC
>
>
>     On Wed, Aug 14, 2013 at 6:34 AM, Ravi <sravi123 at yahoo.com
>     <mailto:sravi123 at yahoo.com>> wrote:
>
>         Hello Everyone !
>
>         I am trying to record calls received in an extension (1003).
>         The dialplan is given below
>
>         <extension name="003">
>             <condition field="destination_number" expression="^003$">
>                 <action application="set"
>         data="record_file_name={recordings_
>         dir}/${strftime(%Y-%m-%d-%H-%M-%S)}_${uuid}.wav" inli$
>                 <action application="record_session"
>         data="{record_file_name}"/>
>                 <action application="bridge" data="user/1003"/>
>             </condition>
>         </extension>
>
>         When I call this extension from an extension 1006, this is
>         what I get in the log. But when I go to the folder
>         /usr/local/freeswitch/recordings/ there are no files at all. I
>         am just wondering if this has got to do with any file
>         permissions, or if I am missing something. Please help.
>
>         Thanks.
>         Ravi
>
>
>         freeswitch at bfree-server> 2013-08-14 06:53:51.185709 [NOTICE]
>         switch_channel.c:1030 New Channel
>         sofia/internal/1006 at 10.0.0.16 <mailto:1006 at 10.0.0.16>
>         [2d3314f4-0480-11e3-9a2f-4d1cb7d8b5b4]
>         2013-08-14 06:53:51.245710 [INFO] mod_dialplan_xml.c:558
>         Processing Ravi <1006>->1003 in context default
>         2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628
>         Bound B-Leg: *1 execute_extension::dx XML features
>         2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628
>         Bound B-Leg: *2
>         record_session::/usr/local/freeswitch/recordings/1006.2013-08-14-06-53-51.wav
>         2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628
>         Bound B-Leg: *3 execute_extension::cf XML features
>         2013-08-14 06:53:51.245710 [INFO] switch_ivr_async.c:3628
>         Bound B-Leg: *4 execute_extension::att_xfer XML features
>         2013-08-14 06:53:51.245710 [NOTICE] switch_channel.c:1030 New
>         Channel sofia/internal/sip:1003 at 10.0.0.10:5060
>         <http://sip:1003 at 10.0.0.10:5060/>
>         [2d3d1a26-0480-11e3-9a4d-4d1cb7d8b5b4]
>         2013-08-14 06:53:51.445707 [NOTICE] sofia.c:5884 Ring-Ready
>         sofia/internal/sip:1003 at 10.0.0.10:5060
>         <http://sip:1003 at 10.0.0.10:5060/>!
>         2013-08-14 06:53:51.445707 [INFO] switch_ivr_originate.c:1190
>         Sending early media
>         2013-08-14 06:53:51.445707 [NOTICE] sofia_media.c:92
>         Pre-Answer sofia/internal/1006 at 10.0.0.16 <mailto:1006 at 10.0.0.16>!
>         2013-08-14 06:53:52.705700 [NOTICE] sofia.c:6547 Channel
>         [sofia/internal/sip:1003 at 10.0.0.10:5060
>         <http://sip:1003 at 10.0.0.10:5060/>] has been answered
>         2013-08-14 06:53:52.725708 [NOTICE]
>         switch_ivr_originate.c:3437 Channel
>         [sofia/internal/1006 at 10.0.0.16 <mailto:1006 at 10.0.0.16>] has
>         been answered
>         2013-08-14 06:54:05.005696 [NOTICE] sofia.c:715 Hangup
>         sofia/internal/sip:1003 at 10.0.0.10:5060
>         <http://sip:1003 at 10.0.0.10:5060/> [CS_EXCHANGE_MEDIA]
>         [NORMAL_CLEARING]
>         2013-08-14 06:54:05.025711 [NOTICE] switch_ivr_bridge.c:1575
>         Hangup sofia/internal/1006 at 10.0.0.16 <mailto:1006 at 10.0.0.16>
>         [CS_EXECUTE] [NORMAL_CLEARING]
>         2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1567
>         Session 139 (sofia/internal/sip:1003 at 10.0.0.10:5060
>         <http://sip:1003 at 10.0.0.10:5060/>) Ended
>         2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1571
>         Close Channel sofia/internal/sip:1003 at 10.0.0.10:5060
>         <http://sip:1003 at 10.0.0.10:5060/> [CS_DESTROY]
>         2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1567
>         Session 138 (sofia/internal/1006 at 10.0.0.16
>         <mailto:1006 at 10.0.0.16>) Ended
>         2013-08-14 06:54:05.025711 [NOTICE] switch_core_session.c:1571
>         Close Channel sofia/internal/1006 at 10.0.0.16
>         <mailto:1006 at 10.0.0.16> [CS_DESTROY]
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>         <http://www.freeswitchsolutions.com/>
>
>         
>          </>
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org <http://www.freeswitch.org/>
>         http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>         http://www.cluecon.com <http://www.cluecon.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 <http://www.freeswitch.org/>
>
>
>
>
>     -- 
>     Michael S Collins
>     Twitter: @mercutioviz
>     http://www.FreeSWITCH.org <http://www.freeswitch.org/>
>     http://www.ClueCon.com <http://www.cluecon.com/>
>     http://www.OSTAG.org <http://www.ostag.org/>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com/>
>
>     
>      </>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org <http://www.freeswitch.org/>
>     http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>     http://www.cluecon.com <http://www.cluecon.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 <http://www.freeswitch.org/>
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>     <http://www.freeswitchsolutions.com/>
>
>     
>      </>
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org <http://www.freeswitch.org/>
>     http://wiki.freeswitch.org <http://wiki.freeswitch.org/>
>     http://www.cluecon.com <http://www.cluecon.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 <http://www.freeswitch.org/>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ssinyagin at yahoo.com  Fri Aug 16 16:56:11 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Fri, 16 Aug 2013 05:56:11 -0700 (PDT)
Subject: [Freeswitch-users] Public and default context question
In-Reply-To: <FB7D97A214987F458242ACBDF87614073B1BD1957F@clubvirtual40.ClubMessage.local>
References: <FB7D97A214987F458242ACBDF87614073B1BD1957F@clubvirtual40.ClubMessage.local>
Message-ID: <1376657771.33928.YahooMailNeo@web126203.mail.ne1.yahoo.com>

here I started building an SBC configuration, but the project was put on hold because of changing priorities:
https://github.com/xlab1/sipfe_freeswitch_common

basically, you don't have to stick to "default" and "public" contexts, and you can easily create your own. In my example, I'm using "inside" and "outside" contexts, so that it's more logical for an SBC.

Also an advice, start with an empty configuration and add and modify files from the vanilla config as you feel needed. The standard configuration is fine as a demo and for quick tests, but it contains too many unneeded things for a production system.







>________________________________
> From: Grant Bagdasarian <GB at cm.nl>
>To: "FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)" <freeswitch-users at lists.freeswitch.org> 
>Sent: Friday, August 16, 2013 1:28 PM
>Subject: [Freeswitch-users] Public and default context question
> 
>
>
>Hello,
>?
>I just started to play around with FreeSwitch, and I have a couple of questions.
>?
>Is the default context only used for devices which are registered with FreeSwitch?
>If I have a Carrier which communicates with FS through a public IP and a Kamailio SIP Proxy which communicates with FS through a private IP address, both will use the public context right?
>?
>My goal is to configure FS as a SBC, and I see there is a sbc directory in the ../conf folder, but when I change the pre-process value to point to sbc/freeswitch.xml, I?m not able to start freeswitch anymore.
><X-PRE-PROCESS cmd="include" data="sbc/freeswitch.xml"/>
>?
>There are no devices(IP Phones for example) which will register with FreeSwitch, so basically I won?t be needing the default context, correct?
>?
>Regards,
>?
>Grant
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 intralanman at freeswitch.org  Fri Aug 16 16:56:32 2013
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Fri, 16 Aug 2013 08:56:32 -0400
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <520D9E43.9000704@xtronics.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
	<CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>
	<CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>
	<CAHccO0X_Dr+fP8eHmkeTVwergLDaP+kjAbQ0n_p24x-NdXY1tQ@mail.gmail.com>
	<CAHgk4BVujRwVT55BfLLo8kcLU76Jr09hdQua01dvYYuL7tuVeg@mail.gmail.com>
	<CAHgk4BUd9cB9wzWATSg06MwzdQs8+uS+LpdDQN5SkYhTA6pg_A@mail.gmail.com>
	<520D9E43.9000704@xtronics.com>
Message-ID: <CAB+Kzh7op6pPz9xFYtkwqC-_2R93PhY8waxiULuHGAwno6KwXQ@mail.gmail.com>

My personal experiences leave me with the following list by most to least
preferred:

Polycom
Cisco SPA 5xx
Yealink
Aastra
Grandstream

All of them have issues. Some are easier than others to work around.  Some
are better for NATed environments.  It really just depends what you need in
your particular situation.
-Ray
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From freeswitch-list at puzzled.xs4all.nl  Fri Aug 16 17:14:51 2013
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 16 Aug 2013 15:14:51 +0200
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <CE33869E.78FAE%krice@freeswitch.org>
References: <CE33869E.78FAE%krice@freeswitch.org>
Message-ID: <520E25CB.7030404@puzzled.xs4all.nl>

On 08/16/2013 02:34 PM, Ken Rice wrote:
> I?ve never found polycoms hard to debug or configure, you just need to
> set up a proper boot server for them... Once you do that, the configs
> are straight forward, and it?ll actually logs back to that server...

Compared to other brands Polycom sure could make it easier to configure 
their phones. The zillion (sub-sub-sub-sub-)xml options, (afaik) lack of 
a good xml editor that can handle the Polycom xml files on Linux and a 
500 page admin guide don't really scream easy. The bootserver is easy, 
IMHO an advanced configuration with a lot of features not so much.

Perhaps it's an idea to do a few "How to become a Polycom config master" 
sessions during the weekly conf call with video showing the steps to 
configure a Polycom phone with basic, advanced and very advanced configs?

Regards,
Patrick



From krice at freeswitch.org  Fri Aug 16 17:21:58 2013
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 16 Aug 2013 08:21:58 -0500
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <520E25CB.7030404@puzzled.xs4all.nl>
Message-ID: <CE3391A6.78FB7%krice@freeswitch.org>

The thing is with the polycom configs, theres really only a few things you
actually have to touch from the default configs... And their 500 page admin
guide is that size because they exhaustively document things... They even
document the settings you should never touch...

And that's a good idea for the weekly conference call! I like it... We'll
get that one on the schedule soon!

K

On 8/16/13 8:14 AM, "Patrick Lists" <freeswitch-list at puzzled.xs4all.nl>
wrote:

> On 08/16/2013 02:34 PM, Ken Rice wrote:
>> I?ve never found polycoms hard to debug or configure, you just need to
>> set up a proper boot server for them... Once you do that, the configs
>> are straight forward, and it?ll actually logs back to that server...
> 
> Compared to other brands Polycom sure could make it easier to configure
> their phones. The zillion (sub-sub-sub-sub-)xml options, (afaik) lack of
> a good xml editor that can handle the Polycom xml files on Linux and a
> 500 page admin guide don't really scream easy. The bootserver is easy,
> IMHO an advanced configuration with a lot of features not so much.
> 
> Perhaps it's an idea to do a few "How to become a Polycom config master"
> sessions during the weekly conf call with video showing the steps to
> configure a Polycom phone with basic, advanced and very advanced configs?
> 
> Regards,
> Patrick
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From avi at avimarcus.net  Fri Aug 16 17:22:04 2013
From: avi at avimarcus.net (Avi Marcus)
Date: Fri, 16 Aug 2013 13:22:04 +0000
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <520E25CB.7030404@puzzled.xs4all.nl>
References: <CE33869E.78FAE%krice@freeswitch.org>
	<520E25CB.7030404@puzzled.xs4all.nl>
Message-ID: <00000140874a3df1-e0a86002-7ad9-485d-b86b-200a83300469-000000@email.amazonses.com>

Just btw -- this is hardly the first time this question has been asked, and
nothing has been summarized on the wiki for future reference (that I know
of).
So, someone that has the time....

-Avi


On Fri, Aug 16, 2013 at 4:14 PM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 08/16/2013 02:34 PM, Ken Rice wrote:
> > I?ve never found polycoms hard to debug or configure, you just need to
> > set up a proper boot server for them... Once you do that, the configs
> > are straight forward, and it?ll actually logs back to that server...
>
> Compared to other brands Polycom sure could make it easier to configure
> their phones. The zillion (sub-sub-sub-sub-)xml options, (afaik) lack of
> a good xml editor that can handle the Polycom xml files on Linux and a
> 500 page admin guide don't really scream easy. The bootserver is easy,
> IMHO an advanced configuration with a lot of features not so much.
>
> Perhaps it's an idea to do a few "How to become a Polycom config master"
> sessions during the weekly conf call with video showing the steps to
> configure a Polycom phone with basic, advanced and very advanced configs?
>
> Regards,
> Patrick
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lconroy at insensate.co.uk  Fri Aug 16 17:26:20 2013
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Fri, 16 Aug 2013 14:26:20 +0100
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <CE33869E.78FAE%krice@freeswitch.org>
References: <CE33869E.78FAE%krice@freeswitch.org>
Message-ID: <57B0D003-A269-448E-A323-3E8F3B6DD8C0@insensate.co.uk>

Hi there again folks,
 for clarification, I don't know about the "pure SIP" versions or Poly deskphones
-- I whinged only about the Lync variants of their DECT phones.
In my case, once you've had one of your users blame you for the setup and describe the
Polycom kirk butterfly as looking like a sanitary towel (only less useful), it's hard
to retain your enthusiasm.

Do their pure SIP phones really need a separate boot server to be set up before they'll
work properly? For central config sure, but for general operation of individual phones?

Re. SIP Deskphones, the Aastras seem to be popular on site and work quite well.
We have a couple of hundred of them, and with no failures (at all) yet, so they must be
pretty robust.

Older Cisco desk phones have left me with mental scars -- don't know about the new ones.
[In the early days, Cisco's interpretation of SIP was kinda different, causing many debug
sessions to find out what they needed]
Grandstream 2K series worked, but suffered from unacceptable mains hum and odd little
incompatibilities with fS.
The Siemens/Gigaset SIP desk phones were pretty boring, IIRC, but had pretty good sound
 quality.
For the others, can't say.

all the best,
  Lawrence


On 16 Aug 2013, at 13:34, Ken Rice wrote:
> I?ve never found polycoms hard to debug or configure, you just need to set
> up a proper boot server for them... Once you do that, the configs are
> straight forward, and it?ll actually logs back to that server...
> 
> 
> On 8/16/13 7:25 AM, "Peter Blackford" <peter at onemetric.com> wrote:
> 
>> I HATE Polycoms they are hard to configure and hard to debug.
>> 
>> I really like the Cisco SPA series as they are reliable however also hard to
>> debug. Snoms and Yealinks are good and a lot easier to debug (Pulling packet
>> captures directly from the phone makes life a lot easier)
>> 
>> Just my 2c.
>> 
>> 
>> On 16 August 2013 20:35, Lawrence Conroy <lconroy at insensate.co.uk> wrote:
>>> Hi Folks,
>>>  +3 for the Gigasets. I have a number of these setups -- with different
>>> handsets.
>>> These are working fine with fS in all of my servers.
>>> TL;DR: I paid my own money for two of these for my own use. My wife doesn't
>>> know
>>>  she's using SIP for all calls -- it's just a 'phone.
>>> 
>>> Voice Quality seems to be good, range is excellent, battery life is fine
>>> (though
>>> they spend most of their time in the desk stands, charging).
>>> I have not used bluetooth headsets with them, so can't talk on that. The 2.5
>>> mm
>>> audio connector works fine (some folk prefer clipping the phone to their
>>> waist
>>> and wearing a headset -- YMMV).
>>> The N300A (consumer DECT SIP base station) voicemail is OK, but IMHO
>>> over-compresses.
>>> The in-built web server by which the base is configured is OK if slow.
>>> Configuring the base via a connected handset is possible, but like every
>>> manufacturer's
>>> version of that I've seen, it's not recommended for mere mortals without a
>>> lot of patience.
>>> [...and apart from the annoyance of the battery compartment door falling open
>>> if you
>>> drop them, and the handset NOT being waterproof/toiletproof, they're
>>> remarkably robust]
>>> 
>>> As for Polycom, can't speak for the "straight" SIP versions, but I have
>>> experience
>>>  of their Lync DECT multicell system & handsets; IMHO these don't even reach
>>> the
>>>  'blows goats' level. Work has 50 of them; user feedback is that voice
>>> quality,
>>>  coverage, battery life is "not as good as we might hope".
>>> Opinion seems to be that these are a step back from the siemens cellular DECT
>>> 'phones we had on the old "steam" PBX (being polite).
>>> Sigh.  For all portable systems, Try before you buy.
>>> 
>>> all the best,
>>>  Lawrence
>>> 
>>> 
>>> On 16 Aug 2013, at 10:15, Andrew Cassidy wrote:
>>>>> Polycom is the name I see the most on this list, I've some some experience
>>>>> with Snom and Grandstream. The do the job but my favourites are Cisco
>>>>> SPA50x series.
>>>>> 
>>>>> 
>>>>> On 16 August 2013 08:22, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>>>>> 
>>>>>>> Gigaset 610IP is a great cordless ip-phone. Works perfectly with
>>>>>>> FreeSWITCH. You can also have several handsets served from one base.
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>>  ------------------------------
>>>>>>> *From:* Karl Schmidt <karl at xtronics.com>
>>>>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>>>> *Sent:* Friday, August 16, 2013 5:36 AM
>>>>>>> *Subject:* [Freeswitch-users] SIP phones -
>>>>>>> 
>>>>>>> I am thinking of getting a couple of SIP phones for our system - I see
>>>>>>> some used on ebay.
>>>>>>> 
>>>>>>> Are there ones to avoid? Good ones to seek?
>>>>>>> 
>>>>>>> I have a couple of places where there needs to be a backup phone in case
>>>>>>> the workstation is
>>>>>>> rebooting. I can set up POE if that sounds like a good idea.
>>>>>>> 
>>>>>>> Depending on what happens, I could then start retiring some of the
>>>>> analog
>>>>>>> phones replacing with SIP
>>>>>>> phones.
>>>>>>> 
>>>>>>> The bits I'm finding on the web is almost all by someone selling and
>>>>>>> mostly spammy information. A
>>>>>>> few clues could go a long way.
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>> ---------------------------------------------------------------------------
>>>>> -----
>>>>>>> Karl Schmidt                                  EMail Karl at xtronics.com
>>>>>>> Transtronics, Inc.                              WEB
>>>>>>> http://secure.transtronics.com
>>>>>>> 3209 West 9th Street                            Ph (785) 841-3089
>>>>> <tel:%28785%29%20841-3089>
>>>>>>> Lawrence, KS 66049                              FAX (785) 841-0434
>>>>> <tel:%28785%29%20841-0434>
>>>>>>> 
>>>>>>> Never ruin an apology with an excuse.
>>>>>>> 
>>>>>>> 
>>>>> ---------------------------------------------------------------------------
>>>>> -----
>>>>>>> 
>>>>>>> 
>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>> 
>>>>>>> 
>>>>> 
>>>>> 
>>>>> --
>>>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>>>> Managing Director
>>>>> 
>>>>> 
>>>>> *T <info at cassidywebservices.co.uk> *03300 100 960
>>>>> *F<info at cassidywebservices.co.uk>
>>>>> *03300 100 961
>>>>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>>>>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>>> <http://www.cassidywebservices.co.uk>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> 
>> 
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> -- 
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 moises.silva at gmail.com  Fri Aug 16 17:27:19 2013
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 16 Aug 2013 09:27:19 -0400
Subject: [Freeswitch-users] freetdm rxgain txgain
In-Reply-To: <520D44BB.5050805@xtronics.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
	<CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
	<CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
	<520C4938.4010202@xtronics.com>
	<CAHKqMb7uoZKy0O4NAdSnFxdCzxv5sTJ5S4rJYDncv6dgyjPNEA@mail.gmail.com>
	<520D44BB.5050805@xtronics.com>
Message-ID: <CAA4nhyDp8AyFN4zV03P4chac33cv+cP9Xw35z53-RYdcwdkqSw@mail.gmail.com>

On Thu, Aug 15, 2013 at 5:14 PM, Karl Schmidt <karl at xtronics.com> wrote:

> On 08/15/2013 04:59 AM, Nandy Dagondon wrote:
> > Hi Karl,
> >
> > Did you test the milliwatt tone using the FXS port? I asked because you
> said it's not sending the
> > ringing signal. Do you mean the phones did not ring. The Digium cards
> require a separate power
> > supply feed. Without it - it can't generate the 100V ringing voltage.
> This is very obvious but let's
> > not discounting this possibility. :-)
> >
> > /Nandy
>
> I used the tone on both fxo and fxs ports - the extra power connector is
> there - for some reason the
> gain settings in /etc/freeswitch/freetdm.conf don't seem to get applied.
>
> The key test is the output of
>   $ dahdi_diag
>
> I just ran it again while is active to see if it makes a difference.
>
> I looks like it wants to see settings in zt.conf
>
> Which config file should this be using - zt.conf or freetdm.conf ? I can
> find documentation that
> points to both?
>
>
freetdm.conf is mandatory, zt.conf is optional. The gain parameters
specified in freetdm.conf are common to all board manufacturers, the gains
are applied *after* reading from the driver and *before* writing to the
driver, so if you changed the gains in freetdm.conf, is no surprise DAHDI
debug logs did not indicate any gain change. The DAHDI drivers have their
own kernel-level gain adjustment. You will have problems if you're tweaking
gains at both the kernel level and the freetdm level (user space).

Post your freetdm.conf, freetdm.conf.xml, zt.conf, system.conf files using
http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_Pastebin

Note you can tweak the gain live during the call using the fs_cli command
'ftdm gains'

-
Moy

*Moises Silva
**Manager, Software Engineering***

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



**<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>

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From moises.silva at gmail.com  Fri Aug 16 17:32:14 2013
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 16 Aug 2013 09:32:14 -0400
Subject: [Freeswitch-users] Wanpipe - Wanrouter help - freetdm and
	freeswitch
In-Reply-To: <1376051767.14626.YahooMailNeo@web160501.mail.bf1.yahoo.com>
References: <1376051767.14626.YahooMailNeo@web160501.mail.bf1.yahoo.com>
Message-ID: <CAA4nhyD0p7UHEJAw_JD-WThwBYtnNGOyrNA1J-vGr7WU02uufg@mail.gmail.com>

On Fri, Aug 9, 2013 at 8:36 AM, Ravi <sravi123 at yahoo.com> wrote:

> Hello Everyone !
>
> I am not sure if this error is relevant for this mailing list. My
> apologies if this is not the forum. I am trying to configure the Sangoma
> card, with the freetdm installation for Freeswitch. When I complete the
> wanpipe installation, and try wanrouter status, and wanrouter start I get
> the following:
>
> [root at bfree-server log]# wanrouter start
>
> Starting up device: wanpipe1
>
>
> wanconfig: WAN device wanpipe1 driver load failed !!
> : ioctl(wanpipe1,ROUTER_SETUP) failed:
> : 22 - Invalid argument
>
>
> Wanpipe driver did not load properly
> Please check /var/log/wanrouter and
> /var/log/messages for errors
>
> Configuring interfaces: w1g1 w1g1: unknown interface: No such device
>
> done.
> ___________________________________________________________________
>
> Can anyone help me understand why this is happening ? And how to resolve
> it.
>
>
Per the error message, it failed to configure the card due to an invalid
argument. As indicated in the error message as well, more information is
contained in /var/log/messages and /var/log/wanrouter

When asking for help like this, you have to include your wanpipe
configuration. If you provide all configuration to tech support (they have
a nice script that collects your hardware info and what not) they should be
able to point to the error right away (may be you can as well if you check
the logs).

Cheers,

*Moises Silva
**Manager, Software Engineering***

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



**<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>

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From eagle.antonio at gmail.com  Fri Aug 16 17:43:36 2013
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Fri, 16 Aug 2013 14:43:36 +0100
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <57B0D003-A269-448E-A323-3E8F3B6DD8C0@insensate.co.uk>
References: <CE33869E.78FAE%krice@freeswitch.org>
	<57B0D003-A269-448E-A323-3E8F3B6DD8C0@insensate.co.uk>
Message-ID: <520E2C88.4070405@gmail.com>

Hi ,

I have around 100 , Polycom 331,  onsite for a callcenter operation 
connected trough headsets.
All of them connected to FS and it works like a charm.

You don't need the boot server it simple helps in large setups , 
firmware roll outs , etc BTW i'm using a plain FTP server ( Win Freeftpd).

Also if the boot server is not present the phone will resume its normal 
boot , it complains , but proceeds.
It only gets tricky if your have their "roaming profile" option enabled 
, and i do and it also works like a charm but it requires a boot server 
to be there.

Sometimes you have to fight with the xml config to get the thing rolling.
The only complain i have is with the low volume when using a headset , 
have to find the AMP option on the manual.

A/t




On 8/16/13 2:26 PM, Lawrence Conroy wrote:
> Hi there again folks,
>   for clarification, I don't know about the "pure SIP" versions or Poly deskphones
> -- I whinged only about the Lync variants of their DECT phones.
> In my case, once you've had one of your users blame you for the setup and describe the
> Polycom kirk butterfly as looking like a sanitary towel (only less useful), it's hard
> to retain your enthusiasm.
>
> Do their pure SIP phones really need a separate boot server to be set up before they'll
> work properly? For central config sure, but for general operation of individual phones?
>
> Re. SIP Deskphones, the Aastras seem to be popular on site and work quite well.
> We have a couple of hundred of them, and with no failures (at all) yet, so they must be
> pretty robust.
>
> Older Cisco desk phones have left me with mental scars -- don't know about the new ones.
> [In the early days, Cisco's interpretation of SIP was kinda different, causing many debug
> sessions to find out what they needed]
> Grandstream 2K series worked, but suffered from unacceptable mains hum and odd little
> incompatibilities with fS.
> The Siemens/Gigaset SIP desk phones were pretty boring, IIRC, but had pretty good sound
>   quality.
> For the others, can't say.
>
> all the best,
>    Lawrence
>
>
> On 16 Aug 2013, at 13:34, Ken Rice wrote:
>> I?ve never found polycoms hard to debug or configure, you just need to set
>> up a proper boot server for them... Once you do that, the configs are
>> straight forward, and it?ll actually logs back to that server...
>>
>>
>> On 8/16/13 7:25 AM, "Peter Blackford" <peter at onemetric.com> wrote:
>>
>>> I HATE Polycoms they are hard to configure and hard to debug.
>>>
>>> I really like the Cisco SPA series as they are reliable however also hard to
>>> debug. Snoms and Yealinks are good and a lot easier to debug (Pulling packet
>>> captures directly from the phone makes life a lot easier)
>>>
>>> Just my 2c.
>>>
>>>
>>> On 16 August 2013 20:35, Lawrence Conroy <lconroy at insensate.co.uk> wrote:
>>>> Hi Folks,
>>>>   +3 for the Gigasets. I have a number of these setups -- with different
>>>> handsets.
>>>> These are working fine with fS in all of my servers.
>>>> TL;DR: I paid my own money for two of these for my own use. My wife doesn't
>>>> know
>>>>   she's using SIP for all calls -- it's just a 'phone.
>>>>
>>>> Voice Quality seems to be good, range is excellent, battery life is fine
>>>> (though
>>>> they spend most of their time in the desk stands, charging).
>>>> I have not used bluetooth headsets with them, so can't talk on that. The 2.5
>>>> mm
>>>> audio connector works fine (some folk prefer clipping the phone to their
>>>> waist
>>>> and wearing a headset -- YMMV).
>>>> The N300A (consumer DECT SIP base station) voicemail is OK, but IMHO
>>>> over-compresses.
>>>> The in-built web server by which the base is configured is OK if slow.
>>>> Configuring the base via a connected handset is possible, but like every
>>>> manufacturer's
>>>> version of that I've seen, it's not recommended for mere mortals without a
>>>> lot of patience.
>>>> [...and apart from the annoyance of the battery compartment door falling open
>>>> if you
>>>> drop them, and the handset NOT being waterproof/toiletproof, they're
>>>> remarkably robust]
>>>>
>>>> As for Polycom, can't speak for the "straight" SIP versions, but I have
>>>> experience
>>>>   of their Lync DECT multicell system & handsets; IMHO these don't even reach
>>>> the
>>>>   'blows goats' level. Work has 50 of them; user feedback is that voice
>>>> quality,
>>>>   coverage, battery life is "not as good as we might hope".
>>>> Opinion seems to be that these are a step back from the siemens cellular DECT
>>>> 'phones we had on the old "steam" PBX (being polite).
>>>> Sigh.  For all portable systems, Try before you buy.
>>>>
>>>> all the best,
>>>>   Lawrence
>>>>
>>>>
>>>> On 16 Aug 2013, at 10:15, Andrew Cassidy wrote:
>>>>>> Polycom is the name I see the most on this list, I've some some experience
>>>>>> with Snom and Grandstream. The do the job but my favourites are Cisco
>>>>>> SPA50x series.
>>>>>>
>>>>>>
>>>>>> On 16 August 2013 08:22, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>>>>>>
>>>>>>>> Gigaset 610IP is a great cordless ip-phone. Works perfectly with
>>>>>>>> FreeSWITCH. You can also have several handsets served from one base.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>   ------------------------------
>>>>>>>> *From:* Karl Schmidt <karl at xtronics.com>
>>>>>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>>>>> *Sent:* Friday, August 16, 2013 5:36 AM
>>>>>>>> *Subject:* [Freeswitch-users] SIP phones -
>>>>>>>>
>>>>>>>> I am thinking of getting a couple of SIP phones for our system - I see
>>>>>>>> some used on ebay.
>>>>>>>>
>>>>>>>> Are there ones to avoid? Good ones to seek?
>>>>>>>>
>>>>>>>> I have a couple of places where there needs to be a backup phone in case
>>>>>>>> the workstation is
>>>>>>>> rebooting. I can set up POE if that sounds like a good idea.
>>>>>>>>
>>>>>>>> Depending on what happens, I could then start retiring some of the
>>>>>> analog
>>>>>>>> phones replacing with SIP
>>>>>>>> phones.
>>>>>>>>
>>>>>>>> The bits I'm finding on the web is almost all by someone selling and
>>>>>>>> mostly spammy information. A
>>>>>>>> few clues could go a long way.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>> ---------------------------------------------------------------------------
>>>>>> -----
>>>>>>>> Karl Schmidt                                  EMail Karl at xtronics.com
>>>>>>>> Transtronics, Inc.                              WEB
>>>>>>>> http://secure.transtronics.com
>>>>>>>> 3209 West 9th Street                            Ph (785) 841-3089
>>>>>> <tel:%28785%29%20841-3089>
>>>>>>>> Lawrence, KS 66049                              FAX (785) 841-0434
>>>>>> <tel:%28785%29%20841-0434>
>>>>>>>> Never ruin an apology with an excuse.
>>>>>>>>
>>>>>>>>
>>>>>> ---------------------------------------------------------------------------
>>>>>> -----
>>>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>
>>>>>> --
>>>>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>>>>> Managing Director
>>>>>>
>>>>>>
>>>>>> *T <info at cassidywebservices.co.uk> *03300 100 960
>>>>>> *F<info at cassidywebservices.co.uk>
>>>>>> *03300 100 961
>>>>>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>>>>>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>>>> <http://www.cassidywebservices.co.uk>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> -- 
>> Ken
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> irc.freenode.net #freeswitch
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eagle.antonio at gmail.com  Fri Aug 16 17:45:12 2013
From: eagle.antonio at gmail.com (Antonio Teixeira)
Date: Fri, 16 Aug 2013 14:45:12 +0100
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <57B0D003-A269-448E-A323-3E8F3B6DD8C0@insensate.co.uk>
References: <CE33869E.78FAE%krice@freeswitch.org>
	<57B0D003-A269-448E-A323-3E8F3B6DD8C0@insensate.co.uk>
Message-ID: <520E2CE8.8000307@gmail.com>

Oops forgot about the major advantage , the webhooks you can force to 
phone to call home when it receives a phone call , etc Create some 
simple interactive apps ,etc.



On 8/16/13 2:26 PM, Lawrence Conroy wrote:
> Hi there again folks,
>   for clarification, I don't know about the "pure SIP" versions or Poly deskphones
> -- I whinged only about the Lync variants of their DECT phones.
> In my case, once you've had one of your users blame you for the setup and describe the
> Polycom kirk butterfly as looking like a sanitary towel (only less useful), it's hard
> to retain your enthusiasm.
>
> Do their pure SIP phones really need a separate boot server to be set up before they'll
> work properly? For central config sure, but for general operation of individual phones?
>
> Re. SIP Deskphones, the Aastras seem to be popular on site and work quite well.
> We have a couple of hundred of them, and with no failures (at all) yet, so they must be
> pretty robust.
>
> Older Cisco desk phones have left me with mental scars -- don't know about the new ones.
> [In the early days, Cisco's interpretation of SIP was kinda different, causing many debug
> sessions to find out what they needed]
> Grandstream 2K series worked, but suffered from unacceptable mains hum and odd little
> incompatibilities with fS.
> The Siemens/Gigaset SIP desk phones were pretty boring, IIRC, but had pretty good sound
>   quality.
> For the others, can't say.
>
> all the best,
>    Lawrence
>
>
> On 16 Aug 2013, at 13:34, Ken Rice wrote:
>> I?ve never found polycoms hard to debug or configure, you just need to set
>> up a proper boot server for them... Once you do that, the configs are
>> straight forward, and it?ll actually logs back to that server...
>>
>>
>> On 8/16/13 7:25 AM, "Peter Blackford" <peter at onemetric.com> wrote:
>>
>>> I HATE Polycoms they are hard to configure and hard to debug.
>>>
>>> I really like the Cisco SPA series as they are reliable however also hard to
>>> debug. Snoms and Yealinks are good and a lot easier to debug (Pulling packet
>>> captures directly from the phone makes life a lot easier)
>>>
>>> Just my 2c.
>>>
>>>
>>> On 16 August 2013 20:35, Lawrence Conroy <lconroy at insensate.co.uk> wrote:
>>>> Hi Folks,
>>>>   +3 for the Gigasets. I have a number of these setups -- with different
>>>> handsets.
>>>> These are working fine with fS in all of my servers.
>>>> TL;DR: I paid my own money for two of these for my own use. My wife doesn't
>>>> know
>>>>   she's using SIP for all calls -- it's just a 'phone.
>>>>
>>>> Voice Quality seems to be good, range is excellent, battery life is fine
>>>> (though
>>>> they spend most of their time in the desk stands, charging).
>>>> I have not used bluetooth headsets with them, so can't talk on that. The 2.5
>>>> mm
>>>> audio connector works fine (some folk prefer clipping the phone to their
>>>> waist
>>>> and wearing a headset -- YMMV).
>>>> The N300A (consumer DECT SIP base station) voicemail is OK, but IMHO
>>>> over-compresses.
>>>> The in-built web server by which the base is configured is OK if slow.
>>>> Configuring the base via a connected handset is possible, but like every
>>>> manufacturer's
>>>> version of that I've seen, it's not recommended for mere mortals without a
>>>> lot of patience.
>>>> [...and apart from the annoyance of the battery compartment door falling open
>>>> if you
>>>> drop them, and the handset NOT being waterproof/toiletproof, they're
>>>> remarkably robust]
>>>>
>>>> As for Polycom, can't speak for the "straight" SIP versions, but I have
>>>> experience
>>>>   of their Lync DECT multicell system & handsets; IMHO these don't even reach
>>>> the
>>>>   'blows goats' level. Work has 50 of them; user feedback is that voice
>>>> quality,
>>>>   coverage, battery life is "not as good as we might hope".
>>>> Opinion seems to be that these are a step back from the siemens cellular DECT
>>>> 'phones we had on the old "steam" PBX (being polite).
>>>> Sigh.  For all portable systems, Try before you buy.
>>>>
>>>> all the best,
>>>>   Lawrence
>>>>
>>>>
>>>> On 16 Aug 2013, at 10:15, Andrew Cassidy wrote:
>>>>>> Polycom is the name I see the most on this list, I've some some experience
>>>>>> with Snom and Grandstream. The do the job but my favourites are Cisco
>>>>>> SPA50x series.
>>>>>>
>>>>>>
>>>>>> On 16 August 2013 08:22, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>>>>>>
>>>>>>>> Gigaset 610IP is a great cordless ip-phone. Works perfectly with
>>>>>>>> FreeSWITCH. You can also have several handsets served from one base.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>   ------------------------------
>>>>>>>> *From:* Karl Schmidt <karl at xtronics.com>
>>>>>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>>>>> *Sent:* Friday, August 16, 2013 5:36 AM
>>>>>>>> *Subject:* [Freeswitch-users] SIP phones -
>>>>>>>>
>>>>>>>> I am thinking of getting a couple of SIP phones for our system - I see
>>>>>>>> some used on ebay.
>>>>>>>>
>>>>>>>> Are there ones to avoid? Good ones to seek?
>>>>>>>>
>>>>>>>> I have a couple of places where there needs to be a backup phone in case
>>>>>>>> the workstation is
>>>>>>>> rebooting. I can set up POE if that sounds like a good idea.
>>>>>>>>
>>>>>>>> Depending on what happens, I could then start retiring some of the
>>>>>> analog
>>>>>>>> phones replacing with SIP
>>>>>>>> phones.
>>>>>>>>
>>>>>>>> The bits I'm finding on the web is almost all by someone selling and
>>>>>>>> mostly spammy information. A
>>>>>>>> few clues could go a long way.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>> ---------------------------------------------------------------------------
>>>>>> -----
>>>>>>>> Karl Schmidt                                  EMail Karl at xtronics.com
>>>>>>>> Transtronics, Inc.                              WEB
>>>>>>>> http://secure.transtronics.com
>>>>>>>> 3209 West 9th Street                            Ph (785) 841-3089
>>>>>> <tel:%28785%29%20841-3089>
>>>>>>>> Lawrence, KS 66049                              FAX (785) 841-0434
>>>>>> <tel:%28785%29%20841-0434>
>>>>>>>> Never ruin an apology with an excuse.
>>>>>>>>
>>>>>>>>
>>>>>> ---------------------------------------------------------------------------
>>>>>> -----
>>>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>
>>>>>> --
>>>>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>>>>> Managing Director
>>>>>>
>>>>>>
>>>>>> *T <info at cassidywebservices.co.uk> *03300 100 960
>>>>>> *F<info at cassidywebservices.co.uk>
>>>>>> *03300 100 961
>>>>>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>>>>>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>>>> <http://www.cassidywebservices.co.uk>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> -- 
>> Ken
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> irc.freenode.net #freeswitch
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 fdelawarde at wirelessmundi.com  Fri Aug 16 17:56:06 2013
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?=)
Date: Fri, 16 Aug 2013 15:56:06 +0200
Subject: [Freeswitch-users] Issue when receiving INVITE with both audio and
	image (T38)
Message-ID: <1376661366.3068.47.camel@luna.madrid.commsmundi.com>

Hello list,

I've seen several gateways (Patton, Audiocodes, Nortel CS2K), sending
both audio+image (T38) in the initial INVITE. In a simple bridge
scenario, FS will just ignore the T38 part and setup an audio connection
only with b-leg.

My issue is later on if/when we receive a re-INVITE (ex: SIP timers
refresh) with the same SDP (audio+T38), FS now considers it as FAX and
sends a re-INVITE to b-leg with T38 only! In case the destination is
voice only, the call hangs up.

---

Asterisk apparently had the same issue, they solved it by setting up
both streams on a-leg (RTP + UDPTL). If they receive packets on UDPTL
they just re-INVITE b-leg with T38.

See: https://reviewboard.asterisk.org/r/208/
Also: https://issues.asterisk.org/jira/browse/ASTERISK-11843

Is this the right way to go? (Also is it feasible in FreeSwitch)? I can
add a JIRA if necessary.

Thanks,
Fran?ois.



From krice at freeswitch.org  Fri Aug 16 17:59:33 2013
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 16 Aug 2013 08:59:33 -0500
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <57B0D003-A269-448E-A323-3E8F3B6DD8C0@insensate.co.uk>
Message-ID: <CE339A75.78FC6%krice@freeswitch.org>

Their pure sip phones do not require a boot server, but if you are deploying
more then 1 it sure makes like a ton easier... In this case the boot server
is little more then a webserver with the correct files loaded on it...

I have found the web interfaces on the polycoms to be less than stellar...
And its a little hard to expose all the buttons and knobs they have in a web
interface... But simple configs are fairly straight forward...


On 8/16/13 8:26 AM, "Lawrence Conroy" <lconroy at insensate.co.uk> wrote:

> Hi there again folks,
>  for clarification, I don't know about the "pure SIP" versions or Poly
> deskphones
> -- I whinged only about the Lync variants of their DECT phones.
> In my case, once you've had one of your users blame you for the setup and
> describe the
> Polycom kirk butterfly as looking like a sanitary towel (only less useful),
> it's hard
> to retain your enthusiasm.
> 
> Do their pure SIP phones really need a separate boot server to be set up
> before they'll
> work properly? For central config sure, but for general operation of
> individual phones?
> 
> Re. SIP Deskphones, the Aastras seem to be popular on site and work quite
> well.
> We have a couple of hundred of them, and with no failures (at all) yet, so
> they must be
> pretty robust.
> 
> Older Cisco desk phones have left me with mental scars -- don't know about the
> new ones.
> [In the early days, Cisco's interpretation of SIP was kinda different, causing
> many debug
> sessions to find out what they needed]
> Grandstream 2K series worked, but suffered from unacceptable mains hum and odd
> little
> incompatibilities with fS.
> The Siemens/Gigaset SIP desk phones were pretty boring, IIRC, but had pretty
> good sound
>  quality.
> For the others, can't say.
> 
> all the best,
>   Lawrence
> 
> 
> On 16 Aug 2013, at 13:34, Ken Rice wrote:
>> I?ve never found polycoms hard to debug or configure, you just need to set
>> up a proper boot server for them... Once you do that, the configs are
>> straight forward, and it?ll actually logs back to that server...
>> 
>> 
>> On 8/16/13 7:25 AM, "Peter Blackford" <peter at onemetric.com> wrote:
>> 
>>> I HATE Polycoms they are hard to configure and hard to debug.
>>> 
>>> I really like the Cisco SPA series as they are reliable however also hard to
>>> debug. Snoms and Yealinks are good and a lot easier to debug (Pulling packet
>>> captures directly from the phone makes life a lot easier)
>>> 
>>> Just my 2c.
>>> 
>>> 
>>> On 16 August 2013 20:35, Lawrence Conroy <lconroy at insensate.co.uk> wrote:
>>>> Hi Folks,
>>>>  +3 for the Gigasets. I have a number of these setups -- with different
>>>> handsets.
>>>> These are working fine with fS in all of my servers.
>>>> TL;DR: I paid my own money for two of these for my own use. My wife doesn't
>>>> know
>>>>  she's using SIP for all calls -- it's just a 'phone.
>>>> 
>>>> Voice Quality seems to be good, range is excellent, battery life is fine
>>>> (though
>>>> they spend most of their time in the desk stands, charging).
>>>> I have not used bluetooth headsets with them, so can't talk on that. The
>>>> 2.5
>>>> mm
>>>> audio connector works fine (some folk prefer clipping the phone to their
>>>> waist
>>>> and wearing a headset -- YMMV).
>>>> The N300A (consumer DECT SIP base station) voicemail is OK, but IMHO
>>>> over-compresses.
>>>> The in-built web server by which the base is configured is OK if slow.
>>>> Configuring the base via a connected handset is possible, but like every
>>>> manufacturer's
>>>> version of that I've seen, it's not recommended for mere mortals without a
>>>> lot of patience.
>>>> [...and apart from the annoyance of the battery compartment door falling
>>>> open
>>>> if you
>>>> drop them, and the handset NOT being waterproof/toiletproof, they're
>>>> remarkably robust]
>>>> 
>>>> As for Polycom, can't speak for the "straight" SIP versions, but I have
>>>> experience
>>>>  of their Lync DECT multicell system & handsets; IMHO these don't even
>>>> reach
>>>> the
>>>>  'blows goats' level. Work has 50 of them; user feedback is that voice
>>>> quality,
>>>>  coverage, battery life is "not as good as we might hope".
>>>> Opinion seems to be that these are a step back from the siemens cellular
>>>> DECT
>>>> 'phones we had on the old "steam" PBX (being polite).
>>>> Sigh.  For all portable systems, Try before you buy.
>>>> 
>>>> all the best,
>>>>  Lawrence
>>>> 
>>>> 
>>>> On 16 Aug 2013, at 10:15, Andrew Cassidy wrote:
>>>>>> Polycom is the name I see the most on this list, I've some some
>>>>>> experience
>>>>>> with Snom and Grandstream. The do the job but my favourites are Cisco
>>>>>> SPA50x series.
>>>>>> 
>>>>>> 
>>>>>> On 16 August 2013 08:22, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>>>>>> 
>>>>>>>> Gigaset 610IP is a great cordless ip-phone. Works perfectly with
>>>>>>>> FreeSWITCH. You can also have several handsets served from one base.
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>>  ------------------------------
>>>>>>>> *From:* Karl Schmidt <karl at xtronics.com>
>>>>>>>> *To:* freeswitch-users at lists.freeswitch.org
>>>>>>>> *Sent:* Friday, August 16, 2013 5:36 AM
>>>>>>>> *Subject:* [Freeswitch-users] SIP phones -
>>>>>>>> 
>>>>>>>> I am thinking of getting a couple of SIP phones for our system - I see
>>>>>>>> some used on ebay.
>>>>>>>> 
>>>>>>>> Are there ones to avoid? Good ones to seek?
>>>>>>>> 
>>>>>>>> I have a couple of places where there needs to be a backup phone in
>>>>>>>> case
>>>>>>>> the workstation is
>>>>>>>> rebooting. I can set up POE if that sounds like a good idea.
>>>>>>>> 
>>>>>>>> Depending on what happens, I could then start retiring some of the
>>>>>> analog
>>>>>>>> phones replacing with SIP
>>>>>>>> phones.
>>>>>>>> 
>>>>>>>> The bits I'm finding on the web is almost all by someone selling and
>>>>>>>> mostly spammy information. A
>>>>>>>> few clues could go a long way.
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>> -------------------------------------------------------------------------
>>>>>> --
>>>>>> -----
>>>>>>>> Karl Schmidt                                  EMail Karl at xtronics.com
>>>>>>>> Transtronics, Inc.                              WEB
>>>>>>>> http://secure.transtronics.com
>>>>>>>> 3209 West 9th Street                            Ph (785) 841-3089
>>>>>> <tel:%28785%29%20841-3089>
>>>>>>>> Lawrence, KS 66049                              FAX (785) 841-0434
>>>>>> <tel:%28785%29%20841-0434>
>>>>>>>> 
>>>>>>>> Never ruin an apology with an excuse.
>>>>>>>> 
>>>>>>>> 
>>>>>> -------------------------------------------------------------------------
>>>>>> --
>>>>>> -----
>>>>>>>> 
>>>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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-user>>>>>>>>
s
>>>>>>>> http://www.freeswitch.org
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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-user>>>>>>>>
s
>>>>>>>> http://www.freeswitch.org
>>>>>>>> 
>>>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> --
>>>>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>>>>> Managing Director
>>>>>> 
>>>>>> 
>>>>>> *T <info at cassidywebservices.co.uk> *03300 100 960
>>>>>> *F<info at cassidywebservices.co.uk>
>>>>>> *03300 100 961
>>>>>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>>>>>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>>>> <http://www.cassidywebservices.co.uk>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>> 
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>> 
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>> 
>> -- 
>> Ken
>> http://www.FreeSWITCH.org
>> http://www.ClueCon.com
>> http://www.OSTAG.org
>> irc.freenode.net #freeswitch
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From anthony.minessale at gmail.com  Fri Aug 16 19:00:41 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 16 Aug 2013 10:00:41 -0500
Subject: [Freeswitch-users] Issue when receiving INVITE with both audio
 and image (T38)
In-Reply-To: <1376661366.3068.47.camel@luna.madrid.commsmundi.com>
References: <1376661366.3068.47.camel@luna.madrid.commsmundi.com>
Message-ID: <CAKbxfG8gO9pg_8umr591Ejjgt80txYgy=fCfpQxfkJG9g4O0MA@mail.gmail.com>

We can investigate that,  Meanwhile you could use regex in your dialplan
for this situation and set "sip_ignore_reinvites=true" to mitigate.



On Fri, Aug 16, 2013 at 8:56 AM, Fran?ois <fdelawarde at wirelessmundi.com>wrote:

> Hello list,
>
> I've seen several gateways (Patton, Audiocodes, Nortel CS2K), sending
> both audio+image (T38) in the initial INVITE. In a simple bridge
> scenario, FS will just ignore the T38 part and setup an audio connection
> only with b-leg.
>
> My issue is later on if/when we receive a re-INVITE (ex: SIP timers
> refresh) with the same SDP (audio+T38), FS now considers it as FAX and
> sends a re-INVITE to b-leg with T38 only! In case the destination is
> voice only, the call hangs up.
>
> ---
>
> Asterisk apparently had the same issue, they solved it by setting up
> both streams on a-leg (RTP + UDPTL). If they receive packets on UDPTL
> they just re-INVITE b-leg with T38.
>
> See: https://reviewboard.asterisk.org/r/208/
> Also: https://issues.asterisk.org/jira/browse/ASTERISK-11843
>
> Is this the right way to go? (Also is it feasible in FreeSwitch)? I can
> add a JIRA if necessary.
>
> Thanks,
> Fran?ois.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From royce3 at gmail.com  Fri Aug 16 19:07:31 2013
From: royce3 at gmail.com (Royce Mitchell III)
Date: Fri, 16 Aug 2013 10:07:31 -0500
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
Message-ID: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>

Hi,

My client has been running FreeSWITCH with low call volume and has been
stable for several months now. This Tuesday morning we switched all call
volume over to FreeSWITCH and everything was running fine until this
morning.

Out of the blue, *all* calls started rejecting
with INCOMPATIBLE_DESTINATION.

Restarting FreeSWITCH fixed the problem. The version of FreeSWITCH is a
trunk build from approximately 7/11/2013. Is this an issue that an update
could fix? Could this be port starvation? Please assist, thanks.

Here's a snippet from the freeswitch log where the first occurrence of that
error:

2013-08-16 09:08:20.048687 [NOTICE] switch_channel.c:1030 New Channel
sofia/external/******9333 at 192.168.1.212:5080[bfa3419b-0f79-4220-8807-773e3f680751]
2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send signal
sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send signal
sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:416
(sofia/external/******9333 at 192.168.1.212:5080) Running State Change CS_NEW
2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:434
(sofia/external/******9333 at 192.168.1.212:5080) State NEW
2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5793 Channel sofia/external/
******9333 at 192.168.1.212:5080 entering state [received][100]
2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5802 Remote SDP:
v=0
o=- 1376662098 1 IN IP4 192.168.1.193
s=-
c=IN IP4 192.168.1.193
t=0 0
m=audio 0 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -

2013-08-16 09:08:20.068687 [NOTICE] sofia.c:6093 Hangup sofia/external/
******9333 at 192.168.1.212:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION]



Royce Mitchell, IT Consultant
ITAS Solutions
royce3 at itas-solutions.com
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From kworm at sofnet.com  Fri Aug 16 02:20:26 2013
From: kworm at sofnet.com (Kevin Wormington)
Date: Thu, 15 Aug 2013 17:20:26 -0500
Subject: [Freeswitch-users] BLF and Registration State
Message-ID: <520D542A.6010105@sofnet.com>

Hi,

I'm doing some testing of BLF with Freeswitch and Yealink T28 phones. It 
works fine as far as flashing the indicator when a monitored extension 
is ringing, solid red when in use, etc.  However, the indicator is green 
even when an extension is not registered.  I haven't found much 
documentation on this, but it seems the indicator should be unlit if the 
extension is not registered.  If I manually clear the registration with 
sip tracing on I see that FS sends the below content in the NOTIFY and 
gets a 200 OK back from the T28:

    Subscription-State: active;expires=1175
    Content-Type: application/dialog-info+xml
    Content-Length: 155

    <?xml version="1.0"?>
    <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="13" 
state="partial" entity="sip:1102 at x.x.x.x">
    </dialog-info>


I assume this is not working because the phone is expecting something 
else in order to turn off the light.  Is this something that is 
configurable or would it need code changes?

Also, is it possible to use an OPTIONS ping or otherwise to determine if 
a registered endpoint is available rather than wait for the expiration? 
  I know this can be done on un-registered gateways.

Thanks,

Kevin


From babak.freeswitch at gmail.com  Fri Aug 16 18:14:12 2013
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Fri, 16 Aug 2013 18:44:12 +0430
Subject: [Freeswitch-users] error after fax receiving completes
Message-ID: <CAEVfcaZaRynN37zsiRo2kc4C3HuUTqhezXxN5Wx9NfdE3_wi-A@mail.gmail.com>

Hi
I tested fax receiving using Freeswitch stable versions 1.2.10 11 and 12
and I get same results. at the end of fax receiving some assertion fails in
heap validation. I build Freeswitch on windows 7 64 bit with visual studio
2010 in 64 bit debug mode. I attached the error screen to this email. if
any more information is needed I can provide it.
thanks
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From kworm at sofnet.com  Fri Aug 16 19:09:22 2013
From: kworm at sofnet.com (Kevin Wormington)
Date: Fri, 16 Aug 2013 10:09:22 -0500
Subject: [Freeswitch-users] BLF
Message-ID: <520E40A2.2080502@sofnet.com>

Hi,

I'm doing some testing of BLF with Freeswitch and Yealink T28 phones. It 
works fine as far as flashing the indicator when a monitored extension 
is ringing, solid red when in use, etc.  However, the indicator is green 
even when an extension is not registered.  I haven't found much 
documentation on this, but it seems the indicator should be unlit if the 
extension is not registered.  If I manually clear the registration with 
sip tracing on I see that FS sends the below content in the NOTIFY and 
gets a 200 OK back from the T28:

    Subscription-State: active;expires=1175
    Content-Type: application/dialog-info+xml
    Content-Length: 155

    <?xml version="1.0"?>
    <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="13" 
state="partial" entity="sip:1102 at x.x.x.x">
    </dialog-info>


I assume this is not working because the phone is expecting something 
else in order to turn off the light.  Is this something that is 
configurable or would it need code changes?

Also, is it possible to use an OPTIONS ping or otherwise to determine if 
a registered endpoint is available rather than wait for the expiration? 
  I know this can be done on un-registered gateways.

Thanks,

Kevin


From ira at connectmevoice.com  Fri Aug 16 19:25:32 2013
From: ira at connectmevoice.com (Ira Tessler)
Date: Fri, 16 Aug 2013 11:25:32 -0400
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
Message-ID: <5d366cbc33ec3d6426038b20168a3a00@mail.gmail.com>

In the past I have seen INCOMPATIBLE_DESTINATION as a result of mis-matched
Codecs. Try and take a look at that.



Ira Tessler
Lead Software Engineer
ConnectMe
(732) 490-9007 x2
ira at connectmevoice.com



*From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Royce
Mitchell III
*Sent:* Friday, August 16, 2013 11:08 AM
*To:* FreeSWITCH Users Help
*Subject:* [Freeswitch-users] INCOMPATIBLE_DESTINATION



Hi,



My client has been running FreeSWITCH with low call volume and has been
stable for several months now. This Tuesday morning we switched all call
volume over to FreeSWITCH and everything was running fine until this
morning.



Out of the blue, *all* calls started rejecting
with INCOMPATIBLE_DESTINATION.



Restarting FreeSWITCH fixed the problem. The version of FreeSWITCH is a
trunk build from approximately 7/11/2013. Is this an issue that an update
could fix? Could this be port starvation? Please assist, thanks.



Here's a snippet from the freeswitch log where the first occurrence of that
error:



2013-08-16 09:08:20.048687 [NOTICE] switch_channel.c:1030 New Channel
sofia/external/******9333 at 192.168.1.212:5080[bfa3419b-0f79-4220-8807-773e3f680751]

2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send signal
sofia/external/******9333 at 192.168.1.212:5080 [BREAK]

2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send signal
sofia/external/******9333 at 192.168.1.212:5080 [BREAK]

2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:416
(sofia/external/******9333 at 192.168.1.212:5080) Running State Change CS_NEW

2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:434
(sofia/external/******9333 at 192.168.1.212:5080) State NEW

2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5793 Channel
sofia/external/******9333 at 192.168.1.212:5080 entering state [received][100]

2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5802 Remote SDP:

v=0

o=- 1376662098 1 IN IP4 192.168.1.193

s=-

c=IN IP4 192.168.1.193

t=0 0

m=audio 0 RTP/AVP 0 18 101

a=rtpmap:0 PCMU/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-15

a=silenceSupp:off - - - -



2013-08-16 09:08:20.068687 [NOTICE] sofia.c:6093 Hangup
sofia/external/******9333 at 192.168.1.212:5080 [CS_NEW]
[INCOMPATIBLE_DESTINATION]







Royce Mitchell, IT Consultant

ITAS Solutions

royce3 at itas-solutions.com
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From freeswitch-list at puzzled.xs4all.nl  Fri Aug 16 19:42:01 2013
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Fri, 16 Aug 2013 17:42:01 +0200
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
Message-ID: <520E4849.9060006@puzzled.xs4all.nl>

On 08/16/2013 05:07 PM, Royce Mitchell III wrote:
[snip]
> Restarting FreeSWITCH fixed the problem. The version of FreeSWITCH is a
> trunk build from approximately 7/11/2013.

Why do you use a development tree ("trunk") in production? Did it have 
some fix not available in the 1.2 branch that you needed? You do know 
that "trunk" is the development tree, may not be stable, have bugs and 
is subject to changes that may make it fall apart? Why not use the 1.2 
branch which is aimed at stability?

Regards,
Patrick


From kworm at sofnet.com  Fri Aug 16 19:48:05 2013
From: kworm at sofnet.com (Kevin Wormington)
Date: Fri, 16 Aug 2013 10:48:05 -0500
Subject: [Freeswitch-users] BLF
In-Reply-To: <520E40A2.2080502@sofnet.com>
References: <520E40A2.2080502@sofnet.com>
Message-ID: <520E49B5.2070306@sofnet.com>

Sorry, for the double post...just joined the list yesterday and though 
the original post did not no through.

On 08/16/2013 10:09 AM, Kevin Wormington wrote:
> Hi,
>
> I'm doing some testing of BLF with Freeswitch and Yealink T28 phones. It
> works fine as far as flashing the indicator when a monitored extension
> is ringing, solid red when in use, etc.  However, the indicator is green
> even when an extension is not registered.  I haven't found much
> documentation on this, but it seems the indicator should be unlit if the
> extension is not registered.  If I manually clear the registration with
> sip tracing on I see that FS sends the below content in the NOTIFY and
> gets a 200 OK back from the T28:
>
[snip]


From krice at freeswitch.org  Fri Aug 16 20:02:42 2013
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 16 Aug 2013 11:02:42 -0500
Subject: [Freeswitch-users] error after fax receiving completes
In-Reply-To: <CAEVfcaZaRynN37zsiRo2kc4C3HuUTqhezXxN5Wx9NfdE3_wi-A@mail.gmail.com>
Message-ID: <CE33B752.78FE0%krice@freeswitch.org>

In the future please be sure to just paste the text in your email and do not
attach screen shots...


On 8/16/13 9:14 AM, "Babak Yakhchali" <babak.freeswitch at gmail.com> wrote:

> Hi
> I tested fax receiving using Freeswitch stable versions 1.2.10 11 and 12 and I
> get same results. at the end of fax receiving some assertion fails in heap
> validation. I build Freeswitch on windows 7 64 bit with visual studio 2010 in
> 64 bit debug mode. I attached the error screen to this email. if any more
> information is needed I can provide it.
> thanks
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From steveayre at gmail.com  Fri Aug 16 20:04:25 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 16 Aug 2013 17:04:25 +0100
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
Message-ID: <CAFiqYumBFafVazhjXUf1CdXgVaYqvbU4QzrPEc2svCLJu0BaFw@mail.gmail.com>

INCOMPATIBLE_DESTINATION means a codec problem.

The remote SDP sends they're offerring PCMU and G729.

What codecs are you allowing, what codecs are you bridging with, and since
G729 is on the list are you perhaps trying to transcode without using
mod_com_g729+licenses?


On 16 August 2013 16:07, Royce Mitchell III <royce3 at gmail.com> wrote:

> Hi,
>
> My client has been running FreeSWITCH with low call volume and has been
> stable for several months now. This Tuesday morning we switched all call
> volume over to FreeSWITCH and everything was running fine until this
> morning.
>
> Out of the blue, *all* calls started rejecting
> with INCOMPATIBLE_DESTINATION.
>
> Restarting FreeSWITCH fixed the problem. The version of FreeSWITCH is a
> trunk build from approximately 7/11/2013. Is this an issue that an update
> could fix? Could this be port starvation? Please assist, thanks.
>
> Here's a snippet from the freeswitch log where the first occurrence of
> that error:
>
> 2013-08-16 09:08:20.048687 [NOTICE] switch_channel.c:1030 New Channel
> sofia/external/******9333 at 192.168.1.212:5080[bfa3419b-0f79-4220-8807-773e3f680751]
> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send signal
> sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send signal
> sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:416
> (sofia/external/******9333 at 192.168.1.212:5080) Running State Change CS_NEW
> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:434
> (sofia/external/******9333 at 192.168.1.212:5080) State NEW
> 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5793 Channel sofia/external/
> ******9333 at 192.168.1.212:5080 entering state [received][100]
> 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5802 Remote SDP:
> v=0
> o=- 1376662098 1 IN IP4 192.168.1.193
> s=-
> c=IN IP4 192.168.1.193
> t=0 0
> m=audio 0 RTP/AVP 0 18 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=fmtp:18 annexb=no
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=silenceSupp:off - - - -
>
> 2013-08-16 09:08:20.068687 [NOTICE] sofia.c:6093 Hangup sofia/external/
> ******9333 at 192.168.1.212:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>
>
>
> Royce Mitchell, IT Consultant
> ITAS Solutions
> royce3 at itas-solutions.com
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Fri Aug 16 20:09:13 2013
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?=)
Date: Fri, 16 Aug 2013 18:09:13 +0200
Subject: [Freeswitch-users] Issue when receiving INVITE with both audio
 and image (T38)
In-Reply-To: <CAKbxfG8gO9pg_8umr591Ejjgt80txYgy=fCfpQxfkJG9g4O0MA@mail.gmail.com>
References: <1376661366.3068.47.camel@luna.madrid.commsmundi.com>
	<CAKbxfG8gO9pg_8umr591Ejjgt80txYgy=fCfpQxfkJG9g4O0MA@mail.gmail.com>
Message-ID: <1376669353.3068.67.camel@luna.madrid.commsmundi.com>

Thanks for the tip. Should I add a JIRA ticket?

Fran?ois.

On Fri, 2013-08-16 at 10:00 -0500, Anthony Minessale wrote: 
> We can investigate that,  Meanwhile you could use regex in your
> dialplan for this situation and set "sip_ignore_reinvites=true" to
> mitigate.
> 
> 
> 
> 
> On Fri, Aug 16, 2013 at 8:56 AM, Fran?ois
> <fdelawarde at wirelessmundi.com> wrote:
>         Hello list,
>         
>         I've seen several gateways (Patton, Audiocodes, Nortel CS2K),
>         sending
>         both audio+image (T38) in the initial INVITE. In a simple
>         bridge
>         scenario, FS will just ignore the T38 part and setup an audio
>         connection
>         only with b-leg.
>         
>         My issue is later on if/when we receive a re-INVITE (ex: SIP
>         timers
>         refresh) with the same SDP (audio+T38), FS now considers it as
>         FAX and
>         sends a re-INVITE to b-leg with T38 only! In case the
>         destination is
>         voice only, the call hangs up.
>         
>         ---
>         
>         Asterisk apparently had the same issue, they solved it by
>         setting up
>         both streams on a-leg (RTP + UDPTL). If they receive packets
>         on UDPTL
>         they just re-INVITE b-leg with T38.
>         
>         See: https://reviewboard.asterisk.org/r/208/
>         Also: https://issues.asterisk.org/jira/browse/ASTERISK-11843
>         
>         Is this the right way to go? (Also is it feasible in
>         FreeSwitch)? I can
>         add a JIRA if necessary.
>         
>         Thanks,
>         Fran?ois.
>         
>         
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org
>         http://www.freeswitchsolutions.com
>         
>         
>         
>         
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 babak.freeswitch at gmail.com  Fri Aug 16 20:12:00 2013
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Fri, 16 Aug 2013 20:42:00 +0430
Subject: [Freeswitch-users] error after fax receiving completes
In-Reply-To: <CE33B752.78FE0%krice@freeswitch.org>
References: <CAEVfcaZaRynN37zsiRo2kc4C3HuUTqhezXxN5Wx9NfdE3_wi-A@mail.gmail.com>
	<CE33B752.78FE0%krice@freeswitch.org>
Message-ID: <CAEVfcabtVOQ74dFoptmQ+Ey=zsSrytBjQmur=Q738JkrUQLkuQ@mail.gmail.com>

sure. I just thought it may help.


On Fri, Aug 16, 2013 at 8:32 PM, Ken Rice <krice at freeswitch.org> wrote:

>  In the future please be sure to just paste the text in your email and do
> not attach screen shots...
>
>
> On 8/16/13 9:14 AM, "Babak Yakhchali" <babak.freeswitch at gmail.com> wrote:
>
> Hi
> I tested fax receiving using Freeswitch stable versions 1.2.10 11 and 12
> and I get same results. at the end of fax receiving some assertion fails in
> heap validation. I build Freeswitch on windows 7 64 bit with visual studio
> 2010 in 64 bit debug mode. I attached the error screen to this email. if
> any more information is needed I can provide it.
> thanks
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sravi123 at yahoo.com  Fri Aug 16 20:36:21 2013
From: sravi123 at yahoo.com (Ravi S)
Date: Fri, 16 Aug 2013 22:06:21 +0530
Subject: [Freeswitch-users] Wanpipe - Wanrouter help - freetdm and
	freeswitch
In-Reply-To: <CAA4nhyD0p7UHEJAw_JD-WThwBYtnNGOyrNA1J-vGr7WU02uufg@mail.gmail.com>
References: <1376051767.14626.YahooMailNeo@web160501.mail.bf1.yahoo.com>
	<CAA4nhyD0p7UHEJAw_JD-WThwBYtnNGOyrNA1J-vGr7WU02uufg@mail.gmail.com>
Message-ID: <520E5505.4050301@yahoo.com>

Thanks Moises, for your response.

Just that I am new to the world of telephony and networking. Was not 
aware initially about what configuration files to look for and what logs 
to check. I am picking up, though. Next time, surely, I will have those 
information handy. My freeswitch is getting configured for a PRI 
connection and if all goes well, there is a market for such a solution 
in India, and I would be very happy to promote Sangoma, given their 
pretty good customer support !

Thanks.
Ravi


On 16-08-2013 19:02, Moises Silva wrote:
> On Fri, Aug 9, 2013 at 8:36 AM, Ravi <sravi123 at yahoo.com 
> <mailto:sravi123 at yahoo.com>> wrote:
>
>     Hello Everyone !
>
>     I am not sure if this error is relevant for this mailing list. My
>     apologies if this is not the forum. I am trying to configure the
>     Sangoma card, with the freetdm installation for Freeswitch. When I
>     complete the wanpipe installation, and try wanrouter status, and
>     wanrouter start I get the following:
>
>     [root at bfree-server log]# wanrouter start
>
>     Starting up device: wanpipe1
>
>
>     wanconfig: WAN device wanpipe1 driver load failed !!
>     : ioctl(wanpipe1,ROUTER_SETUP) failed:
>     : 22 - Invalid argument
>
>
>     Wanpipe driver did not load properly
>     Please check /var/log/wanrouter and
>     /var/log/messages for errors
>
>     Configuring interfaces: w1g1 w1g1: unknown interface: No such device
>
>     done.
>     ___________________________________________________________________
>
>     Can anyone help me understand why this is happening ? And how to
>     resolve it.
>
>
> Per the error message, it failed to configure the card due to an 
> invalid argument. As indicated in the error message as well, more 
> information is contained in /var/log/messages and /var/log/wanrouter
>
> When asking for help like this, you have to include your wanpipe 
> configuration. If you provide all configuration to tech support (they 
> have a nice script that collects your hardware info and what not) they 
> should be able to point to the error right away (may be you can as 
> well if you check the logs).
>
> Cheers,
>
> *Moises Silva
> **/Manager, Software Engineering/***
>
> msilva at sangoma.com <mailto:msilva at sangoma.com>
>
> Sangoma Technologies
>
> 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada
>
> 	
>
>
> t.   +1 800 388 2475 (N. America)
>
> t.   +1 905 474 1990 x128
>
> f.   +1 905 474 9223
>
> 	
>
> ** 
> <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>
>
> Products 
> <http://sangoma.com/products?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | 
> Solutions 
> <http://sangoma.com/solutions?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | 
> Events 
> <http://sangoma.com/about_us/events?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | 
> Contact 
> <http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | 
> Wiki 
> <http://wiki.sangoma.com/?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | 
> Facebook 
> <http://www.facebook.com/pages/Sangoma-VoIP-Cards/43578453335?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures> | 
> Twitter 
> <http://www.twitter.com/sangoma?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>`| 
> | YouTube 
> <http://www.youtube.com/sangomatechnologies?utm_source=signature&utm_medium=email&utm_campaign=email%2Bsignatures>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at davri.com  Fri Aug 16 20:52:15 2013
From: bdfoster at davri.com (Brian Foster)
Date: Fri, 16 Aug 2013 12:52:15 -0400
Subject: [Freeswitch-users] Ademco signaling with mod_spandsp
Message-ID: <CAHgk4BXwkQdwmCX7y=K8Vv03m2U6xpnO0gTCbKgKyRPdFmr0eg@mail.gmail.com>

I've been tasked with securing a storage facility with an alarm system.
Basically we'd use POTS or GSM on the site and a POTS line on the
monitoring station. The alarm system is capable of using the ademco
standard. I see there's some code in git in regards to the ademco standard.
Is this usable inside Freeswitch? If so, how can I access it? I'd like to
pay the wiki tax to doxument it if anyone can get me started.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
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From anthony.minessale at gmail.com  Fri Aug 16 22:01:02 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 16 Aug 2013 13:01:02 -0500
Subject: [Freeswitch-users] Issue when receiving INVITE with both audio
 and image (T38)
In-Reply-To: <1376669353.3068.67.camel@luna.madrid.commsmundi.com>
References: <1376661366.3068.47.camel@luna.madrid.commsmundi.com>
	<CAKbxfG8gO9pg_8umr591Ejjgt80txYgy=fCfpQxfkJG9g4O0MA@mail.gmail.com>
	<1376669353.3068.67.camel@luna.madrid.commsmundi.com>
Message-ID: <CAKbxfG_qKgrng5Fc29EBJzvNkkrhh_az6c5i2AjuSxna77vwJA@mail.gmail.com>

Sure
On Aug 16, 2013 11:14 AM, "Fran?ois" <fdelawarde at wirelessmundi.com> wrote:

> Thanks for the tip. Should I add a JIRA ticket?
>
> Fran?ois.
>
> On Fri, 2013-08-16 at 10:00 -0500, Anthony Minessale wrote:
> > We can investigate that,  Meanwhile you could use regex in your
> > dialplan for this situation and set "sip_ignore_reinvites=true" to
> > mitigate.
> >
> >
> >
> >
> > On Fri, Aug 16, 2013 at 8:56 AM, Fran?ois
> > <fdelawarde at wirelessmundi.com> wrote:
> >         Hello list,
> >
> >         I've seen several gateways (Patton, Audiocodes, Nortel CS2K),
> >         sending
> >         both audio+image (T38) in the initial INVITE. In a simple
> >         bridge
> >         scenario, FS will just ignore the T38 part and setup an audio
> >         connection
> >         only with b-leg.
> >
> >         My issue is later on if/when we receive a re-INVITE (ex: SIP
> >         timers
> >         refresh) with the same SDP (audio+T38), FS now considers it as
> >         FAX and
> >         sends a re-INVITE to b-leg with T38 only! In case the
> >         destination is
> >         voice only, the call hangs up.
> >
> >         ---
> >
> >         Asterisk apparently had the same issue, they solved it by
> >         setting up
> >         both streams on a-leg (RTP + UDPTL). If they receive packets
> >         on UDPTL
> >         they just re-INVITE b-leg with T38.
> >
> >         See: https://reviewboard.asterisk.org/r/208/
> >         Also: https://issues.asterisk.org/jira/browse/ASTERISK-11843
> >
> >         Is this the right way to go? (Also is it feasible in
> >         FreeSwitch)? I can
> >         add a JIRA if necessary.
> >
> >         Thanks,
> >         Fran?ois.
> >
> >
> >
> _________________________________________________________________________
> >         Professional FreeSWITCH Consulting Services:
> >         consulting at freeswitch.org
> >         http://www.freeswitchsolutions.com
> >
> >         
> >         
> >
> >         Official FreeSWITCH Sites
> >         http://www.freeswitch.org
> >         http://wiki.freeswitch.org
> >         http://www.cluecon.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
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jorgefren12 at gmail.com  Fri Aug 16 22:22:15 2013
From: jorgefren12 at gmail.com (=?ISO-8859-1?Q?Jorge_N=FA=F1ez?=)
Date: Fri, 16 Aug 2013 13:22:15 -0500
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <CADk4Sr1g2c3toiU7+ikM_WByfUzvxwXKQwLrFXF8KnrtZoXWnQ@mail.gmail.com>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>
	<CE311BFC.78EA0%krice@freeswitch.org>
	<045801ce990e$205b4f90$6111eeb0$@207me.com>
	<CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>
	<049101ce9915$e1cbb5f0$a56321d0$@207me.com>
	<CAOP4xcy6NZG0JKPMYhJp_wxaLqnAjsEkGMgTD5itvBPNdSCPzQ@mail.gmail.com>
	<050301ce993b$4dc37b60$e94a7220$@207me.com>
	<CAKbxfG80vr3b+BULjn5ZhKW6ZjxY6Khy4Ymhk8RWGVFLpuSKoA@mail.gmail.com>
	<0E0D5ED7-E97F-4F17-8290-2D893649A8F8@jerris.com>
	<CAOP4xcwBNPxaxsrFBr48-02QriO_baDv-=o+QhoDsiMKUnq4ug@mail.gmail.com>
	<CAKbxfG8zu0+_AM4zzmAq01BFHxpLygb8-2JdXNujOw6X6UMesw@mail.gmail.com>
	<CADk4Sr1g2c3toiU7+ikM_WByfUzvxwXKQwLrFXF8KnrtZoXWnQ@mail.gmail.com>
Message-ID: <CAOP4xcy9p1PvL6ggmWK=betiGb=N=2DENvMA=ieiSNo0mZF9Rw@mail.gmail.com>

So Can anybody tell any other solution instead of webrtc and icecast to
broadcast a conference in realtime without delay or maybe just a little bit
of delay 1s or 2s?

Thank you very much for your help

Jorge


2013/8/15 Jeff Leung <jleung at v10networks.ca>

> Icecast is known to have a relatively high latency for audio.
>
>
> On Thu, Aug 15, 2013 at 10:34 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> enable_file_write_buffering=false
>> stream_prebuffer=0
>>
>> On the leg doing the recording.  It does't stop the iceast server or
>> libshout from buffering it more.
>>
>>
>>
>>
>>
>> On Thu, Aug 15, 2013 at 12:07 PM, Jorge N??ez <jorgefren12 at gmail.com>wrote:
>>
>>> Anthony and just for test how can I reduce the latency? and thanks
>>> Michael for use webrtc Do I need to use Sipml5 or JSSIP o are there any
>>> othe way to use because I was trying to find some information about
>>> mod_html5 but I didn?t find
>>>
>>> thanks
>>>
>>> Jorge
>>>
>>>
>>> 2013/8/15 Michael Jerris <mike at jerris.com>
>>>
>>>> The alternative if you want in sync audio would be to use webrtc
>>>> endpoints to listen.
>>>>
>>>> On Aug 14, 2013, at 7:19 PM, Anthony Minessale <
>>>> anthony.minessale at gmail.com> wrote:
>>>>
>>>> Icecast is not designed for low-latency and realtime audio.  Its
>>>> designed for higher quality reliable audio.  The servers and the client
>>>> libs both use latency and buffering to guarantee this.  There are some
>>>> techniques to reduce the buffering on at least the FS side but then you
>>>> will start getting skips and resets if you miss any packets.
>>>>
>>>> This comes up all the time when people first try doing this.  Either
>>>> you need to just accept the delay since most people will not even know its
>>>> there or use some other method.
>>>>
>>>>
>>>>
>>>>
>>>> On Wed, Aug 14, 2013 at 5:12 PM, Stephen Dame <sdame at 207me.com> wrote:
>>>>
>>>>> Did you restart the freeswitch server after changes to the config
>>>>> files,   I?m not sure 100% if reloadxml resets the conference settings.
>>>>> ****
>>>>>
>>>>> ** **
>>>>>
>>>>> Also I use some flash based players, I think there where client
>>>>> settiings for buffering.   Can you set buffer in HTML tag?****
>>>>>
>>>>> ** **
>>>>>
>>>>> Regards,****
>>>>>
>>>>> Stephen****
>>>>>
>>>>> ** **
>>>>>
>>>>> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>>>>>
>>>>> ** **
>>>>>
>>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jorge
>>>>> N??ez
>>>>> *Sent:* Wednesday, August 14, 2013 4:18 PM
>>>>>
>>>>> *To:* FreeSWITCH Users Help
>>>>> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big
>>>>> delay****
>>>>>
>>>>> ** **
>>>>>
>>>>> Hi thanks, I set your configuration but nothing changed, I reduced the
>>>>> burst size and it takes me just 11s and also I tried with 8k, 16k and 32k
>>>>> and nothing change****
>>>>>
>>>>> ** **
>>>>>
>>>>>     <limits>****
>>>>>
>>>>>         <clients>100</clients>****
>>>>>
>>>>>         <sources>10</sources>****
>>>>>
>>>>>         <threadpool>5</threadpool>****
>>>>>
>>>>>         <queue-size>524288</queue-size>****
>>>>>
>>>>>         <!-- <queue-size>524288</queue-size> -->****
>>>>>
>>>>>         <client-timeout>30</client-timeout>****
>>>>>
>>>>>         <header-timeout>15</header-timeout>****
>>>>>
>>>>>         <source-timeout>10</source-timeout>****
>>>>>
>>>>>         <!-- If enabled, this will provide a burst of data when a
>>>>> client ****
>>>>>
>>>>>              first connects, thereby significantly reducing the
>>>>> startup ****
>>>>>
>>>>>              time for listeners that do substantial buffering. However,
>>>>> ****
>>>>>
>>>>>              it also significantly increases latency between the source
>>>>> ****
>>>>>
>>>>>              client and listening client.  For low-latency setups, you
>>>>> ****
>>>>>
>>>>>              might want to disable this. -->****
>>>>>
>>>>>         <burst-on-connect>0</burst-on-connect>****
>>>>>
>>>>>         <!-- same as burst-on-connect, but this allows for being more*
>>>>> ***
>>>>>
>>>>>              specific on how much to burst. Most people won't need to*
>>>>> ***
>>>>>
>>>>>              change from the default 64k. Applies to all mountpoints *
>>>>> ***
>>>>>
>>>>> ** **
>>>>>
>>>>>  -->****
>>>>>
>>>>>         <burst-size>4096</burst-size>****
>>>>>
>>>>>     </limits>****
>>>>>
>>>>> ** **
>>>>>
>>>>> Regards****
>>>>>
>>>>> ** **
>>>>>
>>>>> Jorge****
>>>>>
>>>>> ** **
>>>>>
>>>>> ** **
>>>>>
>>>>> 2013/8/14 Stephen Dame <sdame at 207me.com>****
>>>>>
>>>>> Jorge, ****
>>>>>
>>>>>  ****
>>>>>
>>>>> Play around with the burst size and queue size?  here is my xml config
>>>>> around 3-4 second delay from 16k freeswitch conference. ****
>>>>>
>>>>> To small a buffer and the players disconnect? Im also running icecast
>>>>> on same server.****
>>>>>
>>>>>  ****
>>>>>
>>>>> <icecast>****
>>>>>
>>>>>     <limits>****
>>>>>
>>>>>         <clients>100</clients>****
>>>>>
>>>>>         <sources>10</sources>****
>>>>>
>>>>>         <threadpool>5</threadpool>****
>>>>>
>>>>>         <queue-size>524288</queue-size>****
>>>>>
>>>>>         <client-timeout>30</client-timeout>****
>>>>>
>>>>>         <header-timeout>15</header-timeout>****
>>>>>
>>>>>         <source-timeout>10</source-timeout>****
>>>>>
>>>>>         <burst-on-connect>1</burst-on-connect>****
>>>>>
>>>>>         <!-- same as burst-on-connect, but this allows for being more*
>>>>> ***
>>>>>
>>>>>              specific on how much to burst. Most people won't need to*
>>>>> ***
>>>>>
>>>>>              change from the default 64k. Applies to all mountpoints
>>>>> -->****
>>>>>
>>>>>         <burst-size>65535</burst-size>****
>>>>>
>>>>>     </limits>****
>>>>>
>>>>>  ****
>>>>>
>>>>> Regards,****
>>>>>
>>>>> Stephen****
>>>>>
>>>>>  ****
>>>>>
>>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice
>>>>> *Sent:* Wednesday, August 14, 2013 12:35 PM
>>>>> *To:* FreeSWITCH Users Help
>>>>> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big
>>>>> delay****
>>>>>
>>>>>  ****
>>>>>
>>>>> You have to keep in mind that icecast itself has a fair bit of
>>>>> buffering involved with it and theres not a lot you can do about that
>>>>>
>>>>>
>>>>> On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com> wrote:****
>>>>>
>>>>> Hi I am using mod_shout to send a conference call to icecast and hear
>>>>> it from a tag audio in html5 in realtime, but it has a big delay like 12
>>>>> seconds. How can I reduce the latency of the audio sent from freeswitch or
>>>>> what can I do to improve this.
>>>>>
>>>>> Regards
>>>>>
>>>>> Jorge****
>>>>> ------------------------------
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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****
>>>>>
>>>>>
>>>>> --
>>>>> Ken
>>>>> *http://www.FreeSWITCH.org <http://www.freeswitch.org/>
>>>>> http://www.ClueCon.com <http://www.cluecon.com/>
>>>>> http://www.OSTAG.org <http://www.ostag.org/>
>>>>> *irc.freenode.net #freeswitch****
>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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****
>>>>>
>>>>> ** **
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 royce3 at gmail.com  Sat Aug 17 00:03:22 2013
From: royce3 at gmail.com (Royce Mitchell III)
Date: Fri, 16 Aug 2013 15:03:22 -0500
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <520E4849.9060006@puzzled.xs4all.nl>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
	<520E4849.9060006@puzzled.xs4all.nl>
Message-ID: <CAHwzD9=rgWduCQHFQUaU0a+13uiLjiZRJoQKdGKQdODVjK7ihg@mail.gmail.com>

yes I paid them to implement a needed feature.



Royce Mitchell, IT Consultant
ITAS Solutions
royce3 at itas-solutions.com


On Fri, Aug 16, 2013 at 10:42 AM, Patrick Lists <
freeswitch-list at puzzled.xs4all.nl> wrote:

> On 08/16/2013 05:07 PM, Royce Mitchell III wrote:
> [snip]
> > Restarting FreeSWITCH fixed the problem. The version of FreeSWITCH is a
> > trunk build from approximately 7/11/2013.
>
> Why do you use a development tree ("trunk") in production? Did it have
> some fix not available in the 1.2 branch that you needed? You do know
> that "trunk" is the development tree, may not be stable, have bugs and
> is subject to changes that may make it fall apart? Why not use the 1.2
> branch which is aimed at stability?
>
> Regards,
> Patrick
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 royce3 at gmail.com  Sat Aug 17 00:06:37 2013
From: royce3 at gmail.com (Royce Mitchell III)
Date: Fri, 16 Aug 2013 15:06:37 -0500
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <CAFiqYumBFafVazhjXUf1CdXgVaYqvbU4QzrPEc2svCLJu0BaFw@mail.gmail.com>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
	<CAFiqYumBFafVazhjXUf1CdXgVaYqvbU4QzrPEc2svCLJu0BaFw@mail.gmail.com>
Message-ID: <CAHwzD9n+CfCbCYNvwPcSiwYQoQe9bved3uK6q1=okh+2ravZoA@mail.gmail.com>

My FreeSWITCH is configured to prefer PCMU, and the devices it is talking
to are Adtran 908e's. The Adtrans are configured for the default codec
group which is supposed to be PCMU, but I can reconfigure them to
explicitly allow only PCMU. I will try that and see if it makes a
difference.



Royce Mitchell, IT Consultant
ITAS Solutions
royce3 at itas-solutions.com


On Fri, Aug 16, 2013 at 11:04 AM, Steven Ayre <steveayre at gmail.com> wrote:

> INCOMPATIBLE_DESTINATION means a codec problem.
>
> The remote SDP sends they're offerring PCMU and G729.
>
> What codecs are you allowing, what codecs are you bridging with, and since
> G729 is on the list are you perhaps trying to transcode without using
> mod_com_g729+licenses?
>
>
> On 16 August 2013 16:07, Royce Mitchell III <royce3 at gmail.com> wrote:
>
>> Hi,
>>
>> My client has been running FreeSWITCH with low call volume and has been
>> stable for several months now. This Tuesday morning we switched all call
>> volume over to FreeSWITCH and everything was running fine until this
>> morning.
>>
>> Out of the blue, *all* calls started rejecting
>> with INCOMPATIBLE_DESTINATION.
>>
>> Restarting FreeSWITCH fixed the problem. The version of FreeSWITCH is a
>> trunk build from approximately 7/11/2013. Is this an issue that an update
>> could fix? Could this be port starvation? Please assist, thanks.
>>
>> Here's a snippet from the freeswitch log where the first occurrence of
>> that error:
>>
>> 2013-08-16 09:08:20.048687 [NOTICE] switch_channel.c:1030 New Channel
>> sofia/external/******9333 at 192.168.1.212:5080[bfa3419b-0f79-4220-8807-773e3f680751]
>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send signal
>> sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send signal
>> sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:416
>> (sofia/external/******9333 at 192.168.1.212:5080) Running State Change
>> CS_NEW
>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:434
>> (sofia/external/******9333 at 192.168.1.212:5080) State NEW
>> 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5793 Channel sofia/external/
>> ******9333 at 192.168.1.212:5080 entering state [received][100]
>> 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5802 Remote SDP:
>> v=0
>> o=- 1376662098 1 IN IP4 192.168.1.193
>> s=-
>> c=IN IP4 192.168.1.193
>> t=0 0
>> m=audio 0 RTP/AVP 0 18 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:18 G729/8000
>> a=fmtp:18 annexb=no
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=silenceSupp:off - - - -
>>
>> 2013-08-16 09:08:20.068687 [NOTICE] sofia.c:6093 Hangup sofia/external/
>> ******9333 at 192.168.1.212:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>
>>
>>
>> Royce Mitchell, IT Consultant
>> ITAS Solutions
>> royce3 at itas-solutions.com
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveayre at gmail.com  Sat Aug 17 00:07:08 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 16 Aug 2013 21:07:08 +0100
Subject: [Freeswitch-users] Auto-detecting Local IP when no gateway set
In-Reply-To: <CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
	<CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>
	<CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>
Message-ID: <CAFiqYu=zPzuKy7eEDBESETjWTaPGLwgdnm8Eu8ZO6PvQOufDgQ@mail.gmail.com>

Wishlist Jira ticket:
http://jira.freeswitch.org/browse/FS-5709

I have a patch I will attach to the ticket shortly that implements this
feature (at least on most Linux platforms).


On 15 August 2013 20:15, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> No such functionality, meanwhile you could use the #exec config directive
> similar to SSI in apache to manually run a shell command every time the
> config is loaded to call a command that prints the ip.
>  On Aug 15, 2013 2:08 PM, "Steven Ayre" <steveayre at gmail.com> wrote:
>
>> Personally I think allowing something like <param name="sip-ip"
>> value="interface:eth0"/> to pick up the primary IP of that interface might
>> work best.
>>
>> Anthm, can you clarify if FS already has any such functionality? Or time
>> for a wishlist Jira?
>>
>> Doing so could work alongside auto-restart=true and therefore pick up IP
>> changes such as when using DHCP.
>>
>>
>>
>> On 15 August 2013 20:00, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> I was going to suggest this too, but if it's on a LAN I'm guessing there
>>> won't be a server and serving it from httpd on the localhost would only see
>>> 127.0.0.1
>>>
>>> ...although you could lookup the current IP of the LAN interface from a
>>> script on the local webserver.
>>>
>>> -Steve
>>>
>>>
>>> On 15 August 2013 13:14, Raymond Chandler <intralanman at freeswitch.org>wrote:
>>>
>>>> Alternatively, you could use xml_curl to serve your configs so that you
>>>> don't actually need local_ip_v4 as you can figure it out based on the IP
>>>> hitting the web server. Doing that will allow you to set the IP in all of
>>>> the configs that need it using functionality that already exists in
>>>> FreeSWITCH today.
>>>>
>>>> -Ray
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lconroy at insensate.co.uk  Sat Aug 17 00:35:53 2013
From: lconroy at insensate.co.uk (Lawrence Conroy)
Date: Fri, 16 Aug 2013 21:35:53 +0100
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <CAHwzD9n+CfCbCYNvwPcSiwYQoQe9bved3uK6q1=okh+2ravZoA@mail.gmail.com>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
	<CAFiqYumBFafVazhjXUf1CdXgVaYqvbU4QzrPEc2svCLJu0BaFw@mail.gmail.com>
	<CAHwzD9n+CfCbCYNvwPcSiwYQoQe9bved3uK6q1=okh+2ravZoA@mail.gmail.com>
Message-ID: <A46F2C5A-8082-4155-A4E0-96CE65AF3431@insensate.co.uk>

Hi there,
 Are you sure about fS allowing PCMU?
According to the remote SDP, your incall is proposing PCMU as its top choice; g729 is 2nd best (quite right too :).
Forcing the adtran to offer only PCMU should not make ay difference to that -- it'll still propose PCMU so no change.
Either your setup is somehow blocking PCMU on the b-leg (but I'd expect to see that on the log), OR is trying to transcode because the b-leg requires some (non-PCMU) codec and can't (again, I'd expect that to be logged), OR fS is not accepting PCMU.
Assuming that PCMU is in the fS vars codec lists, does your dialplan do anything funky with the codec list for an incall?
all the best,
  Lawrence

On 16 Aug 2013, at 21:06, Royce Mitchell III wrote:
> My FreeSWITCH is configured to prefer PCMU, and the devices it is talking
> to are Adtran 908e's. The Adtrans are configured for the default codec
> group which is supposed to be PCMU, but I can reconfigure them to
> explicitly allow only PCMU. I will try that and see if it makes a
> difference.
> 
> 
> 
> Royce Mitchell, IT Consultant
> ITAS Solutions
> royce3 at itas-solutions.com
> 
> 
> On Fri, Aug 16, 2013 at 11:04 AM, Steven Ayre <steveayre at gmail.com> wrote:
> 
>> INCOMPATIBLE_DESTINATION means a codec problem.
>> 
>> The remote SDP sends they're offerring PCMU and G729.
>> 
>> What codecs are you allowing, what codecs are you bridging with, and since
>> G729 is on the list are you perhaps trying to transcode without using
>> mod_com_g729+licenses?
>> 
>> 
>> On 16 August 2013 16:07, Royce Mitchell III <royce3 at gmail.com> wrote:
>> 
>>> Hi,
>>> 
>>> My client has been running FreeSWITCH with low call volume and has been
>>> stable for several months now. This Tuesday morning we switched all call
>>> volume over to FreeSWITCH and everything was running fine until this
>>> morning.
>>> 
>>> Out of the blue, *all* calls started rejecting
>>> with INCOMPATIBLE_DESTINATION.
>>> 
>>> Restarting FreeSWITCH fixed the problem. The version of FreeSWITCH is a
>>> trunk build from approximately 7/11/2013. Is this an issue that an update
>>> could fix? Could this be port starvation? Please assist, thanks.
>>> 
>>> Here's a snippet from the freeswitch log where the first occurrence of
>>> that error:
>>> 
>>> 2013-08-16 09:08:20.048687 [NOTICE] switch_channel.c:1030 New Channel
>>> sofia/external/******9333 at 192.168.1.212:5080[bfa3419b-0f79-4220-8807-773e3f680751]
>>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send signal
>>> sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
>>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send signal
>>> sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
>>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:416
>>> (sofia/external/******9333 at 192.168.1.212:5080) Running State Change
>>> CS_NEW
>>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:434
>>> (sofia/external/******9333 at 192.168.1.212:5080) State NEW
>>> 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5793 Channel sofia/external/
>>> ******9333 at 192.168.1.212:5080 entering state [received][100]
>>> 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5802 Remote SDP:
>>> v=0
>>> o=- 1376662098 1 IN IP4 192.168.1.193
>>> s=-
>>> c=IN IP4 192.168.1.193
>>> t=0 0
>>> m=audio 0 RTP/AVP 0 18 101
>>> a=rtpmap:0 PCMU/8000
>>> a=rtpmap:18 G729/8000
>>> a=fmtp:18 annexb=no
>>> a=rtpmap:101 telephone-event/8000
>>> a=fmtp:101 0-15
>>> a=silenceSupp:off - - - -
>>> 
>>> 2013-08-16 09:08:20.068687 [NOTICE] sofia.c:6093 Hangup sofia/external/
>>> ******9333 at 192.168.1.212:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>> 
>>> 
>>> 
>>> Royce Mitchell, IT Consultant
>>> ITAS Solutions
>>> royce3 at itas-solutions.com
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>> 
>>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>> 
>> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 khorsmann at gmail.com  Sat Aug 17 00:37:16 2013
From: khorsmann at gmail.com (Karsten Horsmann - privat)
Date: Fri, 16 Aug 2013 22:37:16 +0200
Subject: [Freeswitch-users] SIP TLS Issues
In-Reply-To: <F1EBD7E7528EA446BB9898D22914AA3B3D947FEAE2@CGNMSG01.oc.qsc.de>
References: <F1EBD7E7528EA446BB9898D22914AA3B3D947FEAE2@CGNMSG01.oc.qsc.de>
Message-ID: <d5a99b8b-68db-435a-9013-41cbf274ec1f@email.android.com>

Hi Adam, 

try to change the tls mode in the vars.xml to ssl (see the comments in that file for the correct value). 

Some phones are to stupid for tls. This setting helps. And its documented on the wiki. 

Cheers karsten 
-- 
Diese Nachricht wurde von meinem Android-Mobiltelefon mit K-9 Mail gesendet.



"Lappe, Adam" <Adam.Lappe at qsc.de> schrieb:

Hi all,

 

Some more things I tried so far:

 

openssl x509 -noout -modulus -in agent.pem | openssl md5

(stdin)= ebdfb317206ba89d07217c06e1f0d6eb

openssl rsa -noout -modulus -in agent.pem | openssl md5

(stdin)= ebdfb317206ba89d07217c06e1f0d6eb

 

At least the certificate and private key in the agent.pem are correct.

 

There is no output on the cli when I try to register a phone.

 

My guess is that the content of agent.pem and/or cafile.pem is wrong.

 

Can someone please confirm this?

 

Best regards,

Adam

 

 

Wed, 14 Aug, 2013 at 16:07 PM, Adam <ala at qsc.de>:

            

Hi all,

            

i am trying to configure FreeSWITCH to speak TLS with all Clients.

I followed the tutorial on http://wiki.freeswitch.com/wiki/SIP_TLS but I am still not sure what key / cert belongs in which file.

            

I have a SSL123 Thawte Wildcard Certificate.

Am I supposed to cat this cert + priv. key into agent.pem and the primary and secondary intermediate into the cafile.pem?

            

I did this and set the right permissions. The internal sofia profile on port 5061 (TLS) is RUNNING.

            

But no client (for example Polycom VVX1500) can register now.

If I set it TCP and Port 5060 (which is RUNNING as well) everything works fine.

            

Wireshark shows me the following

            

Client    ->         FS                    Client Hello

FS        ->         Client                Alert (Level Fatal, Description: Handshake Failure)

            

I also tested openssl s_client ?connect (IP):5061 ?showcerts but it only says:

CONNECTED(00000003)

139847050823328:error:14077410:SSL routines:SSL23_GET_SERVER_HELLO:sslv3 alert handshake failure:s23_clnt.c:724:

---

no peer certificate available

---

No client certificate CA names sent

---

SSL handshake has read 7 bytes and written 225 bytes

---

New, (NONE), Cipher is (NONE)

Secure Renegotiation IS NOT supported

Compression: NONE

Expansion: NONE

---

            

I guess the problem is the agent.pem and/or cafile.pem

            

agent.pem looks like this

-----BEGIN CERTIFICATE-----

(Thawte SSL123 Wildcard Web Certificate)

-----END CERTIFICATE-----

-----BEGIN RSA PRIVATE KEY-----

(Unencrypted Private Key)

-----END RSA PRIVATE KEY-----

            

cafile.pem like that:

-----BEGIN CERTIFICATE-----

(Thawte Primary Intermediate)

-----END CERTIFICATE-----

-----BEGIN CERTIFICATE-----

(Thawte Secondary Intermediate

-----END CERTIFICATE-----

            

Any suggestions?

            

Thanks in advance,

Adam

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From royce3 at gmail.com  Sat Aug 17 00:57:08 2013
From: royce3 at gmail.com (Royce Mitchell III)
Date: Fri, 16 Aug 2013 15:57:08 -0500
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <A46F2C5A-8082-4155-A4E0-96CE65AF3431@insensate.co.uk>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
	<CAFiqYumBFafVazhjXUf1CdXgVaYqvbU4QzrPEc2svCLJu0BaFw@mail.gmail.com>
	<CAHwzD9n+CfCbCYNvwPcSiwYQoQe9bved3uK6q1=okh+2ravZoA@mail.gmail.com>
	<A46F2C5A-8082-4155-A4E0-96CE65AF3431@insensate.co.uk>
Message-ID: <CAHwzD9ncYxxJxaMYnu2qx78Q_SSLp3onz+5=kN=DSz+vRVQGKA@mail.gmail.com>

The only thing I'm doing in FS regarding codecs is I force PCMU in certain
conditions ( ran into a transcoding bug in FreeSwitch between G722 HD and
G711 with the Polycom phones )



Royce Mitchell, IT Consultant
ITAS Solutions
royce3 at itas-solutions.com


On Fri, Aug 16, 2013 at 3:35 PM, Lawrence Conroy <lconroy at insensate.co.uk>wrote:

> Hi there,
>  Are you sure about fS allowing PCMU?
> According to the remote SDP, your incall is proposing PCMU as its top
> choice; g729 is 2nd best (quite right too :).
> Forcing the adtran to offer only PCMU should not make ay difference to
> that -- it'll still propose PCMU so no change.
> Either your setup is somehow blocking PCMU on the b-leg (but I'd expect to
> see that on the log), OR is trying to transcode because the b-leg requires
> some (non-PCMU) codec and can't (again, I'd expect that to be logged), OR
> fS is not accepting PCMU.
> Assuming that PCMU is in the fS vars codec lists, does your dialplan do
> anything funky with the codec list for an incall?
> all the best,
>   Lawrence
>
> On 16 Aug 2013, at 21:06, Royce Mitchell III wrote:
> > My FreeSWITCH is configured to prefer PCMU, and the devices it is talking
> > to are Adtran 908e's. The Adtrans are configured for the default codec
> > group which is supposed to be PCMU, but I can reconfigure them to
> > explicitly allow only PCMU. I will try that and see if it makes a
> > difference.
> >
> >
> >
> > Royce Mitchell, IT Consultant
> > ITAS Solutions
> > royce3 at itas-solutions.com
> >
> >
> > On Fri, Aug 16, 2013 at 11:04 AM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >
> >> INCOMPATIBLE_DESTINATION means a codec problem.
> >>
> >> The remote SDP sends they're offerring PCMU and G729.
> >>
> >> What codecs are you allowing, what codecs are you bridging with, and
> since
> >> G729 is on the list are you perhaps trying to transcode without using
> >> mod_com_g729+licenses?
> >>
> >>
> >> On 16 August 2013 16:07, Royce Mitchell III <royce3 at gmail.com> wrote:
> >>
> >>> Hi,
> >>>
> >>> My client has been running FreeSWITCH with low call volume and has been
> >>> stable for several months now. This Tuesday morning we switched all
> call
> >>> volume over to FreeSWITCH and everything was running fine until this
> >>> morning.
> >>>
> >>> Out of the blue, *all* calls started rejecting
> >>> with INCOMPATIBLE_DESTINATION.
> >>>
> >>> Restarting FreeSWITCH fixed the problem. The version of FreeSWITCH is a
> >>> trunk build from approximately 7/11/2013. Is this an issue that an
> update
> >>> could fix? Could this be port starvation? Please assist, thanks.
> >>>
> >>> Here's a snippet from the freeswitch log where the first occurrence of
> >>> that error:
> >>>
> >>> 2013-08-16 09:08:20.048687 [NOTICE] switch_channel.c:1030 New Channel
> >>> sofia/external/******9333 at 192.168.1.212:5080
> [bfa3419b-0f79-4220-8807-773e3f680751]
> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send
> signal
> >>> sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send
> signal
> >>> sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:416
> >>> (sofia/external/******9333 at 192.168.1.212:5080) Running State Change
> >>> CS_NEW
> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:434
> >>> (sofia/external/******9333 at 192.168.1.212:5080) State NEW
> >>> 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5793 Channel sofia/external/
> >>> ******9333 at 192.168.1.212:5080 entering state [received][100]
> >>> 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5802 Remote SDP:
> >>> v=0
> >>> o=- 1376662098 1 IN IP4 192.168.1.193
> >>> s=-
> >>> c=IN IP4 192.168.1.193
> >>> t=0 0
> >>> m=audio 0 RTP/AVP 0 18 101
> >>> a=rtpmap:0 PCMU/8000
> >>> a=rtpmap:18 G729/8000
> >>> a=fmtp:18 annexb=no
> >>> a=rtpmap:101 telephone-event/8000
> >>> a=fmtp:101 0-15
> >>> a=silenceSupp:off - - - -
> >>>
> >>> 2013-08-16 09:08:20.068687 [NOTICE] sofia.c:6093 Hangup sofia/external/
> >>> ******9333 at 192.168.1.212:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION]
> >>>
> >>>
> >>>
> >>> Royce Mitchell, IT Consultant
> >>> ITAS Solutions
> >>> royce3 at itas-solutions.com
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.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
> >>>
> >>>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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
> >>
> >>
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveayre at gmail.com  Sat Aug 17 01:12:47 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 16 Aug 2013 22:12:47 +0100
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <CAHwzD9ncYxxJxaMYnu2qx78Q_SSLp3onz+5=kN=DSz+vRVQGKA@mail.gmail.com>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
	<CAFiqYumBFafVazhjXUf1CdXgVaYqvbU4QzrPEc2svCLJu0BaFw@mail.gmail.com>
	<CAHwzD9n+CfCbCYNvwPcSiwYQoQe9bved3uK6q1=okh+2ravZoA@mail.gmail.com>
	<A46F2C5A-8082-4155-A4E0-96CE65AF3431@insensate.co.uk>
	<CAHwzD9ncYxxJxaMYnu2qx78Q_SSLp3onz+5=kN=DSz+vRVQGKA@mail.gmail.com>
Message-ID: <CAFiqYunDhXi+v1H4P=vQnrGW2WRd61w1=nnGwVPo+Pv9CWG3Dw@mail.gmail.com>

Can you paste a debug log of the entire lifetime of the call?


On 16 August 2013 21:57, Royce Mitchell III <royce3 at gmail.com> wrote:

> The only thing I'm doing in FS regarding codecs is I force PCMU in certain
> conditions ( ran into a transcoding bug in FreeSwitch between G722 HD and
> G711 with the Polycom phones )
>
>
>
> Royce Mitchell, IT Consultant
> ITAS Solutions
> royce3 at itas-solutions.com
>
>
> On Fri, Aug 16, 2013 at 3:35 PM, Lawrence Conroy <lconroy at insensate.co.uk>wrote:
>
>> Hi there,
>>  Are you sure about fS allowing PCMU?
>> According to the remote SDP, your incall is proposing PCMU as its top
>> choice; g729 is 2nd best (quite right too :).
>> Forcing the adtran to offer only PCMU should not make ay difference to
>> that -- it'll still propose PCMU so no change.
>> Either your setup is somehow blocking PCMU on the b-leg (but I'd expect
>> to see that on the log), OR is trying to transcode because the b-leg
>> requires some (non-PCMU) codec and can't (again, I'd expect that to be
>> logged), OR fS is not accepting PCMU.
>> Assuming that PCMU is in the fS vars codec lists, does your dialplan do
>> anything funky with the codec list for an incall?
>> all the best,
>>   Lawrence
>>
>> On 16 Aug 2013, at 21:06, Royce Mitchell III wrote:
>> > My FreeSWITCH is configured to prefer PCMU, and the devices it is
>> talking
>> > to are Adtran 908e's. The Adtrans are configured for the default codec
>> > group which is supposed to be PCMU, but I can reconfigure them to
>> > explicitly allow only PCMU. I will try that and see if it makes a
>> > difference.
>> >
>> >
>> >
>> > Royce Mitchell, IT Consultant
>> > ITAS Solutions
>> > royce3 at itas-solutions.com
>> >
>> >
>> > On Fri, Aug 16, 2013 at 11:04 AM, Steven Ayre <steveayre at gmail.com>
>> wrote:
>> >
>> >> INCOMPATIBLE_DESTINATION means a codec problem.
>> >>
>> >> The remote SDP sends they're offerring PCMU and G729.
>> >>
>> >> What codecs are you allowing, what codecs are you bridging with, and
>> since
>> >> G729 is on the list are you perhaps trying to transcode without using
>> >> mod_com_g729+licenses?
>> >>
>> >>
>> >> On 16 August 2013 16:07, Royce Mitchell III <royce3 at gmail.com> wrote:
>> >>
>> >>> Hi,
>> >>>
>> >>> My client has been running FreeSWITCH with low call volume and has
>> been
>> >>> stable for several months now. This Tuesday morning we switched all
>> call
>> >>> volume over to FreeSWITCH and everything was running fine until this
>> >>> morning.
>> >>>
>> >>> Out of the blue, *all* calls started rejecting
>> >>> with INCOMPATIBLE_DESTINATION.
>> >>>
>> >>> Restarting FreeSWITCH fixed the problem. The version of FreeSWITCH is
>> a
>> >>> trunk build from approximately 7/11/2013. Is this an issue that an
>> update
>> >>> could fix? Could this be port starvation? Please assist, thanks.
>> >>>
>> >>> Here's a snippet from the freeswitch log where the first occurrence of
>> >>> that error:
>> >>>
>> >>> 2013-08-16 09:08:20.048687 [NOTICE] switch_channel.c:1030 New Channel
>> >>> sofia/external/******9333 at 192.168.1.212:5080
>> [bfa3419b-0f79-4220-8807-773e3f680751]
>> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send
>> signal
>> >>> sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
>> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send
>> signal
>> >>> sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
>> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:416
>> >>> (sofia/external/******9333 at 192.168.1.212:5080) Running State Change
>> >>> CS_NEW
>> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:434
>> >>> (sofia/external/******9333 at 192.168.1.212:5080) State NEW
>> >>> 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5793 Channel
>> sofia/external/
>> >>> ******9333 at 192.168.1.212:5080 entering state [received][100]
>> >>> 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5802 Remote SDP:
>> >>> v=0
>> >>> o=- 1376662098 1 IN IP4 192.168.1.193
>> >>> s=-
>> >>> c=IN IP4 192.168.1.193
>> >>> t=0 0
>> >>> m=audio 0 RTP/AVP 0 18 101
>> >>> a=rtpmap:0 PCMU/8000
>> >>> a=rtpmap:18 G729/8000
>> >>> a=fmtp:18 annexb=no
>> >>> a=rtpmap:101 telephone-event/8000
>> >>> a=fmtp:101 0-15
>> >>> a=silenceSupp:off - - - -
>> >>>
>> >>> 2013-08-16 09:08:20.068687 [NOTICE] sofia.c:6093 Hangup
>> sofia/external/
>> >>> ******9333 at 192.168.1.212:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>> >>>
>> >>>
>> >>>
>> >>> Royce Mitchell, IT Consultant
>> >>> ITAS Solutions
>> >>> royce3 at itas-solutions.com
>> >>>
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://wiki.freeswitch.org
>> >>> http://www.cluecon.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
>> >>>
>> >>>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.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
>> >>
>> >>
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 tru083 at yahoo.com  Sat Aug 17 02:34:21 2013
From: tru083 at yahoo.com (D D)
Date: Fri, 16 Aug 2013 15:34:21 -0700 (PDT)
Subject: [Freeswitch-users] Should WebRTC work in a double-NAT environment?
In-Reply-To: <CAKbxfG-qYZcVGr80K8x3jO4V0mfQfjcDKcTFMeWDt1MOMKUDFw@mail.gmail.com>
References: <CAK59s4y6YHecUr-Te-TMvJoS=ASxthL8FCuYmNYhEmkfQioFmQ@mail.gmail.com>	<CAKbxfG96LTXzrB92TGNEkFvmTU9bZz1WX=VFK9p+a_Hra9ZDBg@mail.gmail.com>	<CAK59s4yjYoEDVJJyEx2CE0z=tqn_aSNvg76TXBVrFUPR3as=gQ@mail.gmail.com>	<CAK59s4yH0-B8nxK30k7TBgDgivDwJzPfDTeE0kOXiuKKdrFvXQ@mail.gmail.com>
	<CAKbxfG-qYZcVGr80K8x3jO4V0mfQfjcDKcTFMeWDt1MOMKUDFw@mail.gmail.com>
Message-ID: <1376692461.52975.YahooMailNeo@web120704.mail.ne1.yahoo.com>

Hi,

We have a Freeswitch running in a double-NAT environment, where the 

server is in a NAT network, and the client is in a remote NAT network.

In the remote network, using a SIP softphone, we can make calls into the server 

and hearthe media.

Using a web browser on the same network as the switch, we can hear
media in WebRTC (using JSSIP).

But when using a web browser in the remote network, we can see the signaling
but cannot hear the media.

Should WebRTC work in a double-NAT environment?? Any ideas why the
media is not working in this environment?

Thanks,
David
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From jpyle at fidelityvoice.com  Sat Aug 17 02:56:51 2013
From: jpyle at fidelityvoice.com (Jeff Pyle)
Date: Fri, 16 Aug 2013 18:56:51 -0400
Subject: [Freeswitch-users] Codec order flipped in proxy-media mode
	re-INVITE
In-Reply-To: <CACYJG3+dgHd2MMhGdt02BC64ym3X9=wF6KDSyfQwvQTodMouzQ@mail.gmail.com>
References: <CACYJG3+M+KVH30Roj_MB2zo7a3hxQOw+NkBsSWkNZe=OS-jqPg@mail.gmail.com>
	<CAKbxfG9m1xU8t2X7iZ60WLQu5JCWZCit=xtKeGGzomDDaG840Q@mail.gmail.com>
	<CACYJG3+dgHd2MMhGdt02BC64ym3X9=wF6KDSyfQwvQTodMouzQ@mail.gmail.com>
Message-ID: <CACYJG3+Dh93RpbTCcb1gGfxVDT-mfztbw1Ry0jau3UZC_Ak4UA@mail.gmail.com>

It seems even with bypass media the codecs are flipped in the reinvite.
 Setting enable-soa=false breaks reinvites for t.38 and such.

FreeSWITCH Version
1.2.11-n20130816T111409Z-1~wheezy+1+git~20130816T135609Z~64ade54a73
(-n20130816T111409Z-1~wheezy+1git 64ade54 2013-08-16 13:56:09Z)


- Jeff



On Thu, Aug 1, 2013 at 3:38 PM, Jeff Pyle <jpyle at fidelityvoice.com> wrote:

> This seems to break other things.  I lose media, possibly when the far end
> sends 200 OK.  I didn't have an opportunity to do much troubleshooting.
>
> I'm working a different approach that includes removing FS from the media
> path altogether.
>
>
> - Jeff
>
>
> On Thu, Jul 25, 2013 at 8:49 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Try enable-soa=false in the Sofia profile to disable any auto parsing.
>> On Jul 25, 2013 6:41 AM, "Jeff Pyle" <jpyle at fidelityvoice.com> wrote:
>>
>>> Hello,
>>>
>>> FreeSWITCH Version 1.3.17+git~20130318T211211Z~2dc3b47db1.  The config
>>> is a basic SBC to handle NAT traversal between public and private
>>> interfaces.  Proxy-media mode is configured.
>>>
>>> Inbound call from carrier hits us with G711u and G729 in the SDP, in
>>> that order.  Inside PBX chooses G711u and call is established.  30 minutes
>>> later the carrier does a session refresh with a re-INVITE with an identical
>>> SDP as the initial INVITE.  When FreeSWITCH the re-INVITE leaves FreeSWITCH
>>> towards the inside PBX, the attached SDP has G711u and G729 are flipped
>>> such that G729 is now preferred.  The PBX accepts the G729 request and the
>>> session switches to G729.
>>>
>>> There are no SDP smarts in the dialplan.  Codecs are not handled at all
>>> by design.
>>>
>>> Is this something that could be caused by a misconfiguration?  Or, is my
>>> next step a jira?
>>>
>>>
>>>
>>> - Jeff
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 bob.mccarthy at experient.com  Sat Aug 17 02:58:00 2013
From: bob.mccarthy at experient.com (Bob McCarthy)
Date: Fri, 16 Aug 2013 16:58:00 -0600
Subject: [Freeswitch-users] mod_sms endpoints
Message-ID: <06df01ce9ad4$1018d450$304a7cf0$@experient.com>

Is it possible to create a fictious endpoint to route sms messages to ?
using mod_sms to send messages,  I am intercepting the messages and
disseminating them via the event socket. By sending them to nowhere I get a
nuisance error message.

 

Bob McCarthy

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From covici at ccs.covici.com  Sat Aug 17 03:29:40 2013
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 16 Aug 2013 19:29:40 -0400
Subject: [Freeswitch-users] does fs have problems with a . in the password
	field?
Message-ID: <23874.1376695780@ccs.covici.com>

Hi.  I was having problems with one of my carriers -- they changed the
password for my account without notice!  The password they changed to
had a . and they kept saying I had the wrong password or anyway an
authentication error, till at their suggestion, I removed the  on both
sides, and bingo I was able to register.  I am using
FreeSWITCH Version 1.5.3b+git~20130709T135336Z~267ef728e1 (git 267ef72
2013-07-09 13:53:36Z) .

Should a Jira be filed?

-- 
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 sos at sokhapkin.dyndns.org  Sat Aug 17 03:41:04 2013
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Fri, 16 Aug 2013 19:41:04 -0400
Subject: [Freeswitch-users] does fs have problems with a . in the
	password field?
In-Reply-To: <23874.1376695780@ccs.covici.com>
References: <23874.1376695780@ccs.covici.com>
Message-ID: <3296867.u5ZUApYDvv@sos>

"I removed the . on both sides". "." was missed in your email.

On Friday 16 August 2013 19:29:40 covici at ccs.covici.com wrote:
> Hi.  I was having problems with one of my carriers -- they changed the
> password for my account without notice!  The password they changed to
> had a . and they kept saying I had the wrong password or anyway an
> authentication error, till at their suggestion, I removed the  on both
> sides, and bingo I was able to register.  I am using
> FreeSWITCH Version 1.5.3b+git~20130709T135336Z~267ef728e1 (git 267ef72
> 2013-07-09 13:53:36Z) .
> 
> Should a Jira be filed?


From dujinfang at gmail.com  Sat Aug 17 04:04:24 2013
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 17 Aug 2013 08:04:24 +0800
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <CAOP4xcy9p1PvL6ggmWK=betiGb=N=2DENvMA=ieiSNo0mZF9Rw@mail.gmail.com>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>
	<CE311BFC.78EA0%krice@freeswitch.org>
	<045801ce990e$205b4f90$6111eeb0$@207me.com>
	<CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>
	<049101ce9915$e1cbb5f0$a56321d0$@207me.com>
	<CAOP4xcy6NZG0JKPMYhJp_wxaLqnAjsEkGMgTD5itvBPNdSCPzQ@mail.gmail.com>
	<050301ce993b$4dc37b60$e94a7220$@207me.com>
	<CAKbxfG80vr3b+BULjn5ZhKW6ZjxY6Khy4Ymhk8RWGVFLpuSKoA@mail.gmail.com>
	<0E0D5ED7-E97F-4F17-8290-2D893649A8F8@jerris.com>
	<CAOP4xcwBNPxaxsrFBr48-02QriO_baDv-=o+QhoDsiMKUnq4ug@mail.gmail.com>
	<CAKbxfG8zu0+_AM4zzmAq01BFHxpLygb8-2JdXNujOw6X6UMesw@mail.gmail.com>
	<CADk4Sr1g2c3toiU7+ikM_WByfUzvxwXKQwLrFXF8KnrtZoXWnQ@mail.gmail.com>
	<CAOP4xcy9p1PvL6ggmWK=betiGb=N=2DENvMA=ieiSNo0mZF9Rw@mail.gmail.com>
Message-ID: <2ECFB1E034254C48ACBDD31C7E189BA9@gmail.com>

Another solution would to use flash with mod_rtmp.  And to make that work better you need to change the flash code to remove the sound input device since you don't need it and this also helps to remove the confirmation prompt anytime you use a sound input in flash.  

I think I had written a plugin for flowplayer long time ago which works well.


--  
Seven Du
http://www.freeswitch.org.cn
http://about.me/dujinfang
http://www.dujinfang.com


Sent with Sparrow (http://www.sparrowmailapp.com/?sig)


On Saturday, August 17, 2013 at 2:22 AM, Jorge N??ez wrote:

> So Can anybody tell any other solution instead of webrtc and icecast to broadcast a conference in realtime without delay or maybe just a little bit of delay 1s or 2s?
>  
> Thank you very much for your help
>  
> Jorge
>  
>  
> 2013/8/15 Jeff Leung <jleung at v10networks.ca (mailto:jleung at v10networks.ca)>
> > Icecast is known to have a relatively high latency for audio.
> >  
> >  
> > On Thu, Aug 15, 2013 at 10:34 AM, Anthony Minessale <anthony.minessale at gmail.com (mailto:anthony.minessale at gmail.com)> wrote:
> > > enable_file_write_buffering=false
> > > stream_prebuffer=0
> > >  
> > > On the leg doing the recording.  It does't stop the iceast server or libshout from buffering it more.  
> > >  
> > >  
> > >  
> > >  
> > >  
> > > On Thu, Aug 15, 2013 at 12:07 PM, Jorge N??ez <jorgefren12 at gmail.com (mailto:jorgefren12 at gmail.com)> wrote:
> > > > Anthony and just for test how can I reduce the latency? and thanks Michael for use webrtc Do I need to use Sipml5 or JSSIP o are there any othe way to use because I was trying to find some information about mod_html5 but I didn?t find
> > > >  
> > > > thanks
> > > >  
> > > > Jorge
> > > >  
> > > >  
> > > > 2013/8/15 Michael Jerris <mike at jerris.com (mailto:mike at jerris.com)>
> > > > > The alternative if you want in sync audio would be to use webrtc endpoints to listen.
> > > > >  
> > > > > On Aug 14, 2013, at 7:19 PM, Anthony Minessale <anthony.minessale at gmail.com (mailto:anthony.minessale at gmail.com)> wrote:
> > > > > > Icecast is not designed for low-latency and realtime audio.  Its designed for higher quality reliable audio.  The servers and the client libs both use latency and buffering to guarantee this.  There are some techniques to reduce the buffering on at least the FS side but then you will start getting skips and resets if you miss any packets.
> > > > > >  
> > > > > > This comes up all the time when people first try doing this.  Either you need to just accept the delay since most people will not even know its there or use some other method.
> > > > > >  
> > > > > >  
> > > > > >  
> > > > > >  
> > > > > > On Wed, Aug 14, 2013 at 5:12 PM, Stephen Dame <sdame at 207me.com (mailto:sdame at 207me.com)> wrote:
> > > > > > > Did you restart the freeswitch server after changes to the config files,   I?m not sure 100% if reloadxml resets the conference settings.  
> > > > > > >   
> > > > > > > Also I use some flash based players, I think there where client settiings for buffering.   Can you set buffer in HTML tag?
> > > > > > >   
> > > > > > > Regards,
> > > > > > > Stephen
> > > > > > >   
> > > > > > > 207 Technology Group Inc.   1-888-229-9756 (tel:1-888-229-9756)  skype: Stephen_Dame
> > > > > > >   
> > > > > > > From: 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 Jorge N??ez
> > > > > > > Sent: Wednesday, August 14, 2013 4:18 PM
> > > > > > >  
> > > > > > > To: FreeSWITCH Users Help
> > > > > > > Subject: Re: [Freeswitch-users] Mod_shout using with icecast Big delay
> > > > > > >  
> > > > > > >   
> > > > > > > Hi thanks, I set your configuration but nothing changed, I reduced the burst size and it takes me just 11s and also I tried with 8k, 16k and 32k and nothing change
> > > > > > >   
> > > > > > >  
> > > > > > >     <limits>
> > > > > > >  
> > > > > > >         <clients>100</clients>
> > > > > > >  
> > > > > > >         <sources>10</sources>
> > > > > > >  
> > > > > > >         <threadpool>5</threadpool>
> > > > > > >  
> > > > > > >         <queue-size>524288</queue-size>
> > > > > > >  
> > > > > > >         <!-- <queue-size>524288</queue-size> -->
> > > > > > >  
> > > > > > >         <client-timeout>30</client-timeout>
> > > > > > >  
> > > > > > >         <header-timeout>15</header-timeout>
> > > > > > >  
> > > > > > >         <source-timeout>10</source-timeout>
> > > > > > >  
> > > > > > >         <!-- If enabled, this will provide a burst of data when a client  
> > > > > > >  
> > > > > > >              first connects, thereby significantly reducing the startup  
> > > > > > >  
> > > > > > >              time for listeners that do substantial buffering. However,
> > > > > > >  
> > > > > > >              it also significantly increases latency between the source
> > > > > > >  
> > > > > > >              client and listening client.  For low-latency setups, you
> > > > > > >  
> > > > > > >              might want to disable this. -->
> > > > > > >  
> > > > > > >         <burst-on-connect>0</burst-on-connect>
> > > > > > >  
> > > > > > >         <!-- same as burst-on-connect, but this allows for being more
> > > > > > >  
> > > > > > >              specific on how much to burst. Most people won't need to
> > > > > > >  
> > > > > > >              change from the default 64k. Applies to all mountpoints  
> > > > > > >  
> > > > > > >   
> > > > > > >  
> > > > > > >  -->
> > > > > > >  
> > > > > > >         <burst-size>4096</burst-size>
> > > > > > >  
> > > > > > >     </limits>
> > > > > > >  
> > > > > > >   
> > > > > > >  
> > > > > > > Regards
> > > > > > >  
> > > > > > >   
> > > > > > >  
> > > > > > > Jorge
> > > > > > >  
> > > > > > >   
> > > > > > >   
> > > > > > > 2013/8/14 Stephen Dame <sdame at 207me.com (mailto:sdame at 207me.com)>
> > > > > > > > Jorge,  
> > > > > > > >   
> > > > > > > > Play around with the burst size and queue size?  here is my xml config around 3-4 second delay from 16k freeswitch conference.  
> > > > > > > > To small a buffer and the players disconnect? Im also running icecast on same server.
> > > > > > > >   
> > > > > > > > <icecast>
> > > > > > > >     <limits>
> > > > > > > >         <clients>100</clients>
> > > > > > > >         <sources>10</sources>
> > > > > > > >         <threadpool>5</threadpool>
> > > > > > > >         <queue-size>524288</queue-size>
> > > > > > > >         <client-timeout>30</client-timeout>
> > > > > > > >         <header-timeout>15</header-timeout>
> > > > > > > >         <source-timeout>10</source-timeout>
> > > > > > > >         <burst-on-connect>1</burst-on-connect>
> > > > > > > >         <!-- same as burst-on-connect, but this allows for being more
> > > > > > > >              specific on how much to burst. Most people won't need to
> > > > > > > >              change from the default 64k. Applies to all mountpoints  -->
> > > > > > > >         <burst-size>65535</burst-size>
> > > > > > > >     </limits>
> > > > > > > >   
> > > > > > > > Regards,
> > > > > > > > Stephen
> > > > > > > >  
> > > > > > > >   
> > > > > > > > From: 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 Ken Rice
> > > > > > > > Sent: Wednesday, August 14, 2013 12:35 PM
> > > > > > > > To: FreeSWITCH Users Help
> > > > > > > > Subject: Re: [Freeswitch-users] Mod_shout using with icecast Big delay
> > > > > > > >   
> > > > > > > > You have to keep in mind that icecast itself has a fair bit of buffering involved with it and theres not a lot you can do about that
> > > > > > > >  
> > > > > > > >  
> > > > > > > > On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com (http://jorgefren12 at gmail.com/)> wrote:
> > > > > > > > > Hi I am using mod_shout to send a conference call to icecast and hear it from a tag audio in html5 in realtime, but it has a big delay like 12 seconds. How can I reduce the latency of the audio sent from freeswitch or what can I do to improve this.
> > > > > > > > >  
> > > > > > > > > Regards
> > > > > > > > >  
> > > > > > > > > Jorge
> > > > > > > > >  
> > > > > > > > > _________________________________________________________________________
> > > > > > > > > Professional FreeSWITCH Consulting Services:
> > > > > > > > > consulting at freeswitch.org (http://consulting at freeswitch.org/)
> > > > > > > > > http://www.freeswitchsolutions.com (http://www.freeswitchsolutions.com/)
> > > > > > > > >  
> > > > > > > > > 
> > > > > > > > >  (/)
> > > > > > > > >  
> > > > > > > > > Official FreeSWITCH Sites
> > > > > > > > > http://www.freeswitch.org (http://www.freeswitch.org/)
> > > > > > > > > http://wiki.freeswitch.org (http://wiki.freeswitch.org/)
> > > > > > > > > http://www.cluecon.com (http://www.cluecon.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 (http://www.freeswitch.org/)  
> > > > > > > >  
> > > > > > > > --  
> > > > > > > > Ken
> > > > > > > > http://www.FreeSWITCH.org (http://www.freeswitch.org/)
> > > > > > > > http://www.ClueCon.com (http://www.cluecon.com/)
> > > > > > > > http://www.OSTAG.org (http://www.ostag.org/)
> > > > > > > > irc.freenode.net (http://irc.freenode.net/) #freeswitch  
> > > > > > > >  
> > > > > > > > _________________________________________________________________________
> > > > > > > > Professional FreeSWITCH Consulting Services:
> > > > > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > > > > > > > http://www.freeswitchsolutions.com (http://www.freeswitchsolutions.com/)
> > > > > > > >  
> > > > > > > > 
> > > > > > > >  (/)
> > > > > > > >  
> > > > > > > > Official FreeSWITCH Sites
> > > > > > > > http://www.freeswitch.org (http://www.freeswitch.org/)
> > > > > > > > http://wiki.freeswitch.org (http://wiki.freeswitch.org/)
> > > > > > > > http://www.cluecon.com (http://www.cluecon.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 (http://www.freeswitch.org/)  
> > > > > > >   
> > > > > > >  
> > > > > > >  
> > > > > > >  
> > > > > > >  
> > > > > > >  
> > > > > > >  
> > > > > > >  
> > > > > > >  
> > > > > > > _________________________________________________________________________
> > > > > > > Professional FreeSWITCH Consulting Services:
> > > > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > > > > > > http://www.freeswitchsolutions.com (http://www.freeswitchsolutions.com/)
> > > > > > >  
> > > > > > > 
> > > > > > >  (/)
> > > > > > >  
> > > > > > > Official FreeSWITCH Sites
> > > > > > > http://www.freeswitch.org (http://www.freeswitch.org/)
> > > > > > > http://wiki.freeswitch.org (http://wiki.freeswitch.org/)
> > > > > > > http://www.cluecon.com (http://www.cluecon.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 (http://www.freeswitch.org/)
> > > > > > >  
> > > > > >  
> > > > > >  
> > > > > >  
> > > > > > --  
> > > > > > Anthony Minessale II
> > > > > >  
> > > > > > FreeSWITCH http://www.freeswitch.org/
> > > > > > ClueCon http://www.cluecon.com/
> > > > > > Twitter: http://twitter.com/FreeSWITCH_wire
> > > > > >  
> > > > > > AIM: anthm
> > > > > > MSN:anthony_minessale at hotmail.com (mailto:MSN%3Aanthony_minessale at hotmail.com)
> > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:PAYPAL%3Aanthony.minessale at gmail.com)
> > > > > > IRC: irc.freenode.net (http://irc.freenode.net/) #freeswitch
> > > > > >  
> > > > > > FreeSWITCH Developer Conference
> > > > > > sip:888 at conference.freeswitch.org (mailto:sip%3A888 at conference.freeswitch.org)
> > > > > > googletalk:conf+888 at conference.freeswitch.org (mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org)
> > > > > > pstn:+19193869900 (tel:%2B19193869900) _________________________________________________________________________
> > > > > > Professional FreeSWITCH Consulting Services:
> > > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > > > > > http://www.freeswitchsolutions.com
> > > > > >  
> > > > > > 
> > > > > > 
> > > > > >  
> > > > > > Official FreeSWITCH Sites
> > > > > > http://www.freeswitch.org
> > > > > > http://wiki.freeswitch.org
> > > > > > http://www.cluecon.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
> > > > >  
> > > > >  
> > > > > _________________________________________________________________________
> > > > > Professional FreeSWITCH Consulting Services:
> > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > > > > http://www.freeswitchsolutions.com
> > > > >  
> > > > > 
> > > > > 
> > > > >  
> > > > > Official FreeSWITCH Sites
> > > > > http://www.freeswitch.org
> > > > > http://wiki.freeswitch.org
> > > > > http://www.cluecon.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
> > > > >  
> > > >  
> > > >  
> > > > _________________________________________________________________________
> > > > Professional FreeSWITCH Consulting Services:
> > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > > > http://www.freeswitchsolutions.com
> > > >  
> > > > 
> > > > 
> > > >  
> > > > Official FreeSWITCH Sites
> > > > http://www.freeswitch.org
> > > > http://wiki.freeswitch.org
> > > > http://www.cluecon.com
> > > >  
> > > > FreeSWITCH-users mailing list
> > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org)
> > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > > http://www.freeswitch.org
> > > >  
> > >  
> > >  
> > >  
> > > --  
> > > Anthony Minessale II
> > >  
> > > FreeSWITCH http://www.freeswitch.org/
> > > ClueCon http://www.cluecon.com/
> > > Twitter: http://twitter.com/FreeSWITCH_wire
> > >  
> > > AIM: anthm
> > > MSN:anthony_minessale at hotmail.com (mailto:MSN%3Aanthony_minessale at hotmail.com)
> > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com (mailto:PAYPAL%3Aanthony.minessale at gmail.com)
> > > IRC: irc.freenode.net (http://irc.freenode.net) #freeswitch
> > >  
> > > FreeSWITCH Developer Conference
> > > sip:888 at conference.freeswitch.org (mailto:sip%3A888 at conference.freeswitch.org)
> > > googletalk:conf+888 at conference.freeswitch.org (mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org)
> > > pstn:+19193869900 (tel:%2B19193869900)  
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > > http://www.freeswitchsolutions.com
> > >  
> > > 
> > > 
> > >  
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.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
> > >  
> >  
> >  
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > http://www.freeswitchsolutions.com
> >  
> > 
> > 
> >  
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>  
>  


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From covici at ccs.covici.com  Sat Aug 17 04:59:40 2013
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Fri, 16 Aug 2013 20:59:40 -0400
Subject: [Freeswitch-users] does fs have problems with a . in the
	password field?
In-Reply-To: <3296867.u5ZUApYDvv@sos>
References: <23874.1376695780@ccs.covici.com> <3296867.u5ZUApYDvv@sos>
Message-ID: <7266.1376701180@ccs.covici.com>

Amazing what Email clients can do.

Sergey Okhapkin <sos at sokhapkin.dyndns.org> wrote:

> "I removed the . on both sides". "." was missed in your email.
> 
> On Friday 16 August 2013 19:29:40 covici at ccs.covici.com wrote:
> > Hi.  I was having problems with one of my carriers -- they changed the
> > password for my account without notice!  The password they changed to
> > had a . and they kept saying I had the wrong password or anyway an
> > authentication error, till at their suggestion, I removed the  on both
> > sides, and bingo I was able to register.  I am using
> > FreeSWITCH Version 1.5.3b+git~20130709T135336Z~267ef728e1 (git 267ef72
> > 2013-07-09 13:53:36Z) .
> > 
> > Should a Jira be filed?
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
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 sertys at gmail.com  Sat Aug 17 11:07:02 2013
From: sertys at gmail.com (Daniel Ivanov)
Date: Sat, 17 Aug 2013 09:07:02 +0200
Subject: [Freeswitch-users] mod_sms endpoints
In-Reply-To: <06df01ce9ad4$1018d450$304a7cf0$@experient.com>
References: <06df01ce9ad4$1018d450$304a7cf0$@experient.com>
Message-ID: <CAMeZp63hV6_QQQJ52ooc7+0dprYRyXcX4ZUa4yM8WtAx3p0ykw@mail.gmail.com>

I haven't seen fs complain about a message not been sent in the chatplan.
Just process it via a lua handler and do whatever you want with it.
On Aug 17, 2013 2:03 AM, "Bob McCarthy" <bob.mccarthy at experient.com> wrote:

> Is it possible to create a fictious endpoint to route sms messages to ?
> using mod_sms to send messages,  I am intercepting the messages and
> disseminating them via the event socket. By sending them to nowhere I get a
> nuisance error message.****
>
> ** **
>
> Bob McCarthy****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 gmangudai at gmail.com  Sat Aug 17 14:45:07 2013
From: gmangudai at gmail.com (Vincent Xia)
Date: Sat, 17 Aug 2013 18:45:07 +0800
Subject: [Freeswitch-users] something related to channel variables
Message-ID: <CAKQo6tkGjoTQU0bbEKo4uMjCrOW5v7sUy4nUJm7-sTjWq-bfRw@mail.gmail.com>

how to get the contact field of the sip message "invite" or "200 ok" in the
dialplan xml?

how to get the registering ip address of an endpoint in the dialplan xml
(as we have sofia_contact in the command line)?
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From bob at experient.com  Sat Aug 17 20:57:46 2013
From: bob at experient.com (Bob McCarthy)
Date: Sat, 17 Aug 2013 10:57:46 -0600
Subject: [Freeswitch-users] mod_sms endpoints
In-Reply-To: <CAMeZp63hV6_QQQJ52ooc7+0dprYRyXcX4ZUa4yM8WtAx3p0ykw@mail.gmail.com>
References: <06df01ce9ad4$1018d450$304a7cf0$@experient.com>
	<CAMeZp63hV6_QQQJ52ooc7+0dprYRyXcX4ZUa4yM8WtAx3p0ykw@mail.gmail.com>
Message-ID: <071601ce9b6a$e7219720$b564c560$@experient.com>

What I end up doing is deleteing the to header and replacing it with a
registered user.  But I would rather not.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Daniel
Ivanov
Sent: Saturday, August 17, 2013 1:07 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] mod_sms endpoints

 

I haven't seen fs complain about a message not been sent in the chatplan.
Just process it via a lua handler and do whatever you want with it. 

On Aug 17, 2013 2:03 AM, "Bob McCarthy" <bob.mccarthy at experient.com
<mailto:bob.mccarthy at experient.com> > wrote:

Is it possible to create a fictious endpoint to route sms messages to ?
using mod_sms to send messages,  I am intercepting the messages and
disseminating them via the event socket. By sending them to nowhere I get a
nuisance error message.

 

Bob McCarthy


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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

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From smontour at verizon.net  Sat Aug 17 23:04:05 2013
From: smontour at verizon.net (Sami Montour)
Date: Sat, 17 Aug 2013 14:04:05 -0500
Subject: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua
Message-ID: <00fc01ce9b7c$8b9aa1a0$a2cfe4e0$@verizon.net>

I am using Mod lua for call processing on the FreeSwitch. I would like to
wait for some events such as CHANNEL_ANSWER and CHANNEL_HANGUP  in order to
get the answer time and hangup time. Basically, something like

 

Wait for event CHANNEL_ANSWER or CHANNEL_HANGUP

If (event_name = "CHANNEL_ANSWER") then answer_timestamp =
getHeader("Caller-Channel-Answered-Time") 

If (event_name = "CHANNEL_HANGUP") then hangup_timestamp =
getHeader("Event-Date-Timestamp")

 

The session:getState() described in module lua gets the call state but not
the channel state. For instance, 'CS_EXECUTE' is the channel state when it
is executing a dialplan not when the call is answered. 

 

Is there a way to wait for channel events such as CHANNEL_ANSWER and
CHANNEL_HANGUP in order to retrieve answer and hangup timestamps for a call?

 

Any input is very much appreciated.

 

Thanks.

 

Sam

 

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From karl at xtronics.com  Sun Aug 18 04:40:30 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Sat, 17 Aug 2013 19:40:30 -0500
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <00000140874a3df1-e0a86002-7ad9-485d-b86b-200a83300469-000000@email.amazonses.com>
References: <CE33869E.78FAE%krice@freeswitch.org>
	<520E25CB.7030404@puzzled.xs4all.nl>
	<00000140874a3df1-e0a86002-7ad9-485d-b86b-200a83300469-000000@email.amazonses.com>
Message-ID: <521017FE.7070106@xtronics.com>

On 08/16/2013 08:22 AM, Avi Marcus wrote:
> Just btw -- this is hardly the first time this question has been asked, and nothing has been
> summarized on the wiki for future reference (that I know of).
> So, someone that has the time....
>
> -Avi

I'm actually summarizing the response here (as it seemed a bit off topic re freeswitch) :

http://wiki.xtronics.com/index.php/SIP_Phones

Work in processes - feel free grab and post on wiki.freeswitch.org





--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

The society that puts equality before freedom will
end up with neither. The society that puts freedom
before equality will end up with a great measure of both.
- Milton Freidman
--------------------------------------------------------------------------------


From karl at xtronics.com  Sun Aug 18 04:43:27 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Sat, 17 Aug 2013 19:43:27 -0500
Subject: [Freeswitch-users] freetdm rxgain txgain
In-Reply-To: <CAA4nhyDp8AyFN4zV03P4chac33cv+cP9Xw35z53-RYdcwdkqSw@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
	<CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
	<CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
	<520C4938.4010202@xtronics.com>
	<CAHKqMb7uoZKy0O4NAdSnFxdCzxv5sTJ5S4rJYDncv6dgyjPNEA@mail.gmail.com>
	<520D44BB.5050805@xtronics.com>
	<CAA4nhyDp8AyFN4zV03P4chac33cv+cP9Xw35z53-RYdcwdkqSw@mail.gmail.com>
Message-ID: <521018AF.20603@xtronics.com>

On 08/16/2013 08:27 AM, Moises Silva wrote:

> freetdm.conf is mandatory, zt.conf is optional. The gain parameters specified in freetdm.conf are
> common to all board manufacturers, the gains are applied *after* reading from the driver and
> *before* writing to the driver, so if you changed the gains in freetdm.conf, is no surprise DAHDI
> debug logs did not indicate any gain change.

This is not from a normal log - this is bumping the kernel driver to dump it's current settings into 
/var/log/syslog

> The DAHDI drivers have their own kernel-level gain
> adjustment. You will have problems if you're tweaking gains at both the kernel level and the freetdm
> level (user space).

In other words, the zt.conf could override the freetdm.conf ?

I think this is due to overlapping documentation over the time of the evolution of the software.

I'm

>
> Post your freetdm.conf, freetdm.conf.xml, zt.conf, system.conf files using
> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Using_The_Pastebin
>
> Note you can tweak the gain live during the call using the fs_cli command 'ftdm gains'

I also found there was yet another setup file that mattered:
/etc/modprobe.d/dahdi.conf

Looks like adding :
options wctdm24xxp Boostringer=1

Will bring the ring signal up to something that modern phone detect  (60Vrms) and you won't find 
this setting in the manual.

Haven't tested it yet.

I'm also writing this up ( 7 conf files involved so far ) at

http://wiki.xtronics.com/index.php/Freetdm



--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

The society that puts equality before freedom will
end up with neither. The society that puts freedom
before equality will end up with a great measure of both.
- Milton Freidman
--------------------------------------------------------------------------------


From william.king at quentustech.com  Sun Aug 18 05:18:03 2013
From: william.king at quentustech.com (William King)
Date: Sat, 17 Aug 2013 18:18:03 -0700
Subject: [Freeswitch-users] mod_sms endpoints
In-Reply-To: <071601ce9b6a$e7219720$b564c560$@experient.com>
References: <06df01ce9ad4$1018d450$304a7cf0$@experient.com>
	<CAMeZp63hV6_QQQJ52ooc7+0dprYRyXcX4ZUa4yM8WtAx3p0ykw@mail.gmail.com>
	<071601ce9b6a$e7219720$b564c560$@experient.com>
Message-ID: <521020CB.3000101@quentustech.com>

What's the error message you are getting?

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 08/17/2013 09:57 AM, Bob McCarthy wrote:
> What I end up doing is deleteing the to header and replacing it with a
> registered user.  But I would rather not.
> 
>  
> 
> *From:*freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of
> *Daniel Ivanov
> *Sent:* Saturday, August 17, 2013 1:07 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mod_sms endpoints
> 
>  
> 
> I haven't seen fs complain about a message not been sent in the
> chatplan. Just process it via a lua handler and do whatever you want
> with it.
> 
> On Aug 17, 2013 2:03 AM, "Bob McCarthy" <bob.mccarthy at experient.com
> <mailto:bob.mccarthy at experient.com>> wrote:
> 
>     Is it possible to create a fictious endpoint to route sms messages
>     to ? using mod_sms to send messages,  I am intercepting the messages
>     and disseminating them via the event socket. By sending them to
>     nowhere I get a nuisance error message.
> 
>      
> 
>     Bob McCarthy
> 
> 
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
> 
>     
>     
> 
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sravi123 at yahoo.com  Sun Aug 18 16:01:23 2013
From: sravi123 at yahoo.com (Ravi)
Date: Sun, 18 Aug 2013 05:01:23 -0700 (PDT)
Subject: [Freeswitch-users] Transfer / Forward calls - from an operator to
	an extension after the call is picked
Message-ID: <1376827283.11657.YahooMailNeo@web160501.mail.bf1.yahoo.com>

Hello Everyone !

I have successfully configured my PRI connection. I am able to send and receive calls. Now I am trying to have an operator handle one number - say 4302000. When any call is received the operator will talk to the caller and need to transfer the call to an extension. I tried looking at dialplan documentation, and I think I find only instances where the transfer is done within the program and not when some one picks up the call. 

Can some one please point me to right place/ documents where I can find information regarding this. 

I did find a link with some similar request, but the information is limited. 
http://lists.freeswitch.org/pipermail/freeswitch-users/2012-October/088579.html

Thanks for your help. 
Ravi
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From khorsmann at gmail.com  Sun Aug 18 19:11:14 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Sun, 18 Aug 2013 17:11:14 +0200
Subject: [Freeswitch-users] something related to channel variables
In-Reply-To: <CAKQo6tkGjoTQU0bbEKo4uMjCrOW5v7sUy4nUJm7-sTjWq-bfRw@mail.gmail.com>
References: <CAKQo6tkGjoTQU0bbEKo4uMjCrOW5v7sUy4nUJm7-sTjWq-bfRw@mail.gmail.com>
Message-ID: <CAFArqsbmMZwMg=AfmmDDdZWx6tY7=S6vv08Unb+5oONAqPg5PQ@mail.gmail.com>

Hi Vincent.

i am not sure if i get the first point of your question. It feels like a "i
want to manipulate sip packets directly". If so, then an sip-proxy like
kamailio or opensips i more what you want.

For the second part with the remote-ip take a look at the "info" app and
the xml-cdrs for debugging and research for the correct Channel-Variables.
Also http://wiki.freeswitch.org/wiki/Channel_Variables is a good point to
read.




2013/8/17 Vincent Xia <gmangudai at gmail.com>

>
> how to get the contact field of the sip message "invite" or "200 ok" in
> the dialplan xml?
>
> how to get the registering ip address of an endpoint in the dialplan xml
> (as we have sofia_contact in the command line)?
>
>

-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From khorsmann at gmail.com  Sun Aug 18 19:19:24 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Sun, 18 Aug 2013 17:19:24 +0200
Subject: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua
In-Reply-To: <00fc01ce9b7c$8b9aa1a0$a2cfe4e0$@verizon.net>
References: <00fc01ce9b7c$8b9aa1a0$a2cfe4e0$@verizon.net>
Message-ID: <CAFArqsaVJrxcQgPNMFrtigBtc90-YJMDyRkeOgYw1MSLr_h5ww@mail.gmail.com>

Hi Sam,

you should be more verbose what you currently doing. There a many ways to
call an lua script in FS (in the dialplan, as startup-script, as
dialplan-binding etc).

Sounds like you want to fetch some kind of billing informations.

2013/8/17 Sami Montour <smontour at verizon.net>

> I am using Mod lua for call processing on the FreeSwitch. I would like to
> wait for some events such as CHANNEL_ANSWER and CHANNEL_HANGUP  in order to
> get the answer time and hangup time. Basically, something like****
>
> ** **
>
> Wait for event CHANNEL_ANSWER or CHANNEL_HANGUP****
>
> If (event_name = ?CHANNEL_ANSWER?) then answer_timestamp =
> getHeader(?Caller-Channel-Answered-Time?) ****
>
> If (event_name = ?CHANNEL_HANGUP?) then hangup_timestamp =
> getHeader(?Event-Date-Timestamp?)****
>
> ** **
>
> The session:getState() described in module lua gets the call state but not
> the channel state. For instance, ?CS_EXECUTE? is the channel state when it
> is executing a dialplan not when the call is answered. ****
>
> ** **
>
> Is there a way to wait for channel events such as CHANNEL_ANSWER and
> CHANNEL_HANGUP in order to retrieve answer and hangup timestamps for a call?
> ****
>
> ** **
>
> Any input is very much appreciated.****
>
> ** **
>
> Thanks.****
>
> ** **
>
> Sam****
>
> ** **
>
>

-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From bob.mccarthy at experient.com  Sun Aug 18 19:53:04 2013
From: bob.mccarthy at experient.com (Bob McCarthy)
Date: Sun, 18 Aug 2013 09:53:04 -0600
Subject: [Freeswitch-users] mod_sms endpoints
In-Reply-To: <521020CB.3000101@quentustech.com>
References: <06df01ce9ad4$1018d450$304a7cf0$@experient.com>	<CAMeZp63hV6_QQQJ52ooc7+0dprYRyXcX4ZUa4yM8WtAx3p0ykw@mail.gmail.com>	<071601ce9b6a$e7219720$b564c560$@experient.com>
	<521020CB.3000101@quentustech.com>
Message-ID: <075501ce9c2b$07edb9f0$17c92dd0$@experient.com>

This is the error->

2013-08-18 09:51:46.317229 [ERR] sofia_presence.c:265 Chat proto [sip]
from ["Win 7" <sip:3038183060 at 192.168.1.8:5060>;tag=f402884a]
to [911 at 192.168.1.212]
<font face="Arial, sans-serif"
size="2">hello</font><L><http://lis.ddti.net:56074/?LbyR=39b93693-a22a-45bd-
ab42-7c91c3ad2f6e><l><B>C0-12-34-56-78-90<b>
Nobody to send to: Profile internal


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William
King
Sent: Saturday, August 17, 2013 7:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_sms endpoints

What's the error message you are getting?

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 08/17/2013 09:57 AM, Bob McCarthy wrote:
> What I end up doing is deleteing the to header and replacing it with a 
> registered user.  But I would rather not.
> 
>  
> 
> *From:*freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Daniel Ivanov
> *Sent:* Saturday, August 17, 2013 1:07 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] mod_sms endpoints
> 
>  
> 
> I haven't seen fs complain about a message not been sent in the 
> chatplan. Just process it via a lua handler and do whatever you want 
> with it.
> 
> On Aug 17, 2013 2:03 AM, "Bob McCarthy" <bob.mccarthy at experient.com 
> <mailto:bob.mccarthy at experient.com>> wrote:
> 
>     Is it possible to create a fictious endpoint to route sms messages
>     to ? using mod_sms to send messages,  I am intercepting the messages
>     and disseminating them via the event socket. By sending them to
>     nowhere I get a nuisance error message.
> 
>      
> 
>     Bob McCarthy
> 
> 
>
_________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
> 
>     
>     
> 
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
> 
> 
> 
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
>  
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-use
> rs
> http://www.freeswitch.org
> 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 smontour at verizon.net  Sun Aug 18 19:54:24 2013
From: smontour at verizon.net (Sami Montour)
Date: Sun, 18 Aug 2013 10:54:24 -0500
Subject: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua
In-Reply-To: <CAFArqsaVJrxcQgPNMFrtigBtc90-YJMDyRkeOgYw1MSLr_h5ww@mail.gmail.com>
References: <00fc01ce9b7c$8b9aa1a0$a2cfe4e0$@verizon.net>
	<CAFArqsaVJrxcQgPNMFrtigBtc90-YJMDyRkeOgYw1MSLr_h5ww@mail.gmail.com>
Message-ID: <001501ce9c2b$3638c520$a2aa4f60$@verizon.net>

Hi Karsten,

Thank you for the reply. You are correct. I am trying to retrieve billing
information from a basic call. I am calling lua script in the dialplan. When
a call comes in, FS hands it over to lua script for processing.

 

<include>

<extension>

  <condition field="destination_number" expression="^(15[0-9][0-9])$">

   <action application="lua" data="fs_billing.lua"/>

  </condition>

</extension>

</include>

 

I did the same thing with ESL where I listened for CHANNEL_ANSWER and
CHANNEL_HANGUP events then fetched answer and hangup times and that worked
just fine. I would like to do the same thing with lua. 

 

Thanks

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Karsten
Horsmann
Sent: Sunday, August 18, 2013 10:19 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua

 

Hi Sam,

 

you should be more verbose what you currently doing. There a many ways to
call an lua script in FS (in the dialplan, as startup-script, as
dialplan-binding etc).

Sounds like you want to fetch some kind of billing informations.

 

2013/8/17 Sami Montour <smontour at verizon.net>

I am using Mod lua for call processing on the FreeSwitch. I would like to
wait for some events such as CHANNEL_ANSWER and CHANNEL_HANGUP  in order to
get the answer time and hangup time. Basically, something like

 

Wait for event CHANNEL_ANSWER or CHANNEL_HANGUP

If (event_name = ?CHANNEL_ANSWER?) then answer_timestamp =
getHeader(?Caller-Channel-Answered-Time?) 

If (event_name = ?CHANNEL_HANGUP?) then hangup_timestamp =
getHeader(?Event-Date-Timestamp?)

 

The session:getState() described in module lua gets the call state but not
the channel state. For instance, ?CS_EXECUTE? is the channel state when it
is executing a dialplan not when the call is answered. 

 

Is there a way to wait for channel events such as CHANNEL_ANSWER and
CHANNEL_HANGUP in order to retrieve answer and hangup timestamps for a call?

 

Any input is very much appreciated.

 

Thanks.

 

Sam

 

 




 

-- 
Mit freundlichen Gr??en
*Karsten Horsmann*

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From krice at freeswitch.org  Sun Aug 18 19:54:56 2013
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 18 Aug 2013 10:54:56 -0500
Subject: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua
In-Reply-To: <CAFArqsaVJrxcQgPNMFrtigBtc90-YJMDyRkeOgYw1MSLr_h5ww@mail.gmail.com>
References: <00fc01ce9b7c$8b9aa1a0$a2cfe4e0$@verizon.net>
	<CAFArqsaVJrxcQgPNMFrtigBtc90-YJMDyRkeOgYw1MSLr_h5ww@mail.gmail.com>
Message-ID: <2D04E07F-95B8-48AC-852B-65DEC193CEA1@freeswitch.org>

billing in lua is asking for issues... why not do something like triggiering the billing from the xml cdr or such this avoids loss od cdrs as it uses the filesystem as a backup to incase your websvices or billing processor is down

Ken
Sent from my iPad

On Aug 18, 2013, at 10:19, Karsten Horsmann <khorsmann at gmail.com> wrote:

> Hi Sam,
> 
> you should be more verbose what you currently doing. There a many ways to call an lua script in FS (in the dialplan, as startup-script, as dialplan-binding etc).
> 
> Sounds like you want to fetch some kind of billing informations.
> 
> 2013/8/17 Sami Montour <smontour at verizon.net>
>> I am using Mod lua for call processing on the FreeSwitch. I would like to wait for some events such as CHANNEL_ANSWER and CHANNEL_HANGUP  in order to get the answer time and hangup time. Basically, something like
>> 
>>  
>> 
>> Wait for event CHANNEL_ANSWER or CHANNEL_HANGUP
>> 
>> If (event_name = ?CHANNEL_ANSWER?) then answer_timestamp = getHeader(?Caller-Channel-Answered-Time?)
>> 
>> If (event_name = ?CHANNEL_HANGUP?) then hangup_timestamp = getHeader(?Event-Date-Timestamp?)
>> 
>>  
>> 
>> The session:getState() described in module lua gets the call state but not the channel state. For instance, ?CS_EXECUTE? is the channel state when it is executing a dialplan not when the call is answered.
>> 
>>  
>> 
>> Is there a way to wait for channel events such as CHANNEL_ANSWER and CHANNEL_HANGUP in order to retrieve answer and hangup timestamps for a call?
>> 
>>  
>> 
>> Any input is very much appreciated.
>> 
>>  
>> 
>> Thanks.
>> 
>>  
>> 
>> Sam
>> 
> 
> 
> -- 
> Mit freundlichen Gr??en
> *Karsten Horsmann*
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 smontour at verizon.net  Sun Aug 18 20:15:04 2013
From: smontour at verizon.net (Sami Montour)
Date: Sun, 18 Aug 2013 11:15:04 -0500
Subject: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua
In-Reply-To: <2D04E07F-95B8-48AC-852B-65DEC193CEA1@freeswitch.org>
References: <00fc01ce9b7c$8b9aa1a0$a2cfe4e0$@verizon.net>
	<CAFArqsaVJrxcQgPNMFrtigBtc90-YJMDyRkeOgYw1MSLr_h5ww@mail.gmail.com>
	<2D04E07F-95B8-48AC-852B-65DEC193CEA1@freeswitch.org>
Message-ID: <002601ce9c2e$190e2870$4b2a7950$@verizon.net>

The billing I would like to use with lua is for real-time billing where subscribers are debited real-time or at the end of the call.

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Sunday, August 18, 2013 10:55 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua

 

billing in lua is asking for issues... why not do something like triggiering the billing from the xml cdr or such this avoids loss od cdrs as it uses the filesystem as a backup to incase your websvices or billing processor is down

Ken

Sent from my iPad


On Aug 18, 2013, at 10:19, Karsten Horsmann <khorsmann at gmail.com> wrote:

Hi Sam,

 

you should be more verbose what you currently doing. There a many ways to call an lua script in FS (in the dialplan, as startup-script, as dialplan-binding etc).

Sounds like you want to fetch some kind of billing informations.

 

2013/8/17 Sami Montour <smontour at verizon.net>

I am using Mod lua for call processing on the FreeSwitch. I would like to wait for some events such as CHANNEL_ANSWER and CHANNEL_HANGUP  in order to get the answer time and hangup time. Basically, something like

 

Wait for event CHANNEL_ANSWER or CHANNEL_HANGUP

If (event_name = ?CHANNEL_ANSWER?) then answer_timestamp = getHeader(?Caller-Channel-Answered-Time?) 

If (event_name = ?CHANNEL_HANGUP?) then hangup_timestamp = getHeader(?Event-Date-Timestamp?)

 

The session:getState() described in module lua gets the call state but not the channel state. For instance, ?CS_EXECUTE? is the channel state when it is executing a dialplan not when the call is answered. 

 

Is there a way to wait for channel events such as CHANNEL_ANSWER and CHANNEL_HANGUP in order to retrieve answer and hangup timestamps for a call?

 

Any input is very much appreciated.

 

Thanks.

 

Sam

 

 




 

-- 
Mit freundlichen Gr??en
*Karsten Horsmann*

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 khorsmann at gmail.com  Sun Aug 18 20:18:11 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Sun, 18 Aug 2013 18:18:11 +0200
Subject: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua
In-Reply-To: <001501ce9c2b$3638c520$a2aa4f60$@verizon.net>
References: <00fc01ce9b7c$8b9aa1a0$a2cfe4e0$@verizon.net>
	<CAFArqsaVJrxcQgPNMFrtigBtc90-YJMDyRkeOgYw1MSLr_h5ww@mail.gmail.com>
	<001501ce9c2b$3638c520$a2aa4f60$@verizon.net>
Message-ID: <CAFArqsbCY6zMRFPuLmTX3FXiBUU5VnC-Fng8=DXXGdb2MrqjFw@mail.gmail.com>

Hi Sam,

as Ken wrote. That maybe not a good idea to fetch billing informations with
lua.
And i am not sure, if you call the lua-script within the dialplan - if it
lives anymore at hangup to do something. May some of the freeswitch-jedis
knows that.

Cheers.

2013/8/18 Sami Montour <smontour at verizon.net>

> Hi Karsten,****
>
> Thank you for the reply. You are correct. I am trying to retrieve billing
> information from a basic call. I am calling lua script in the dialplan.
> When a call comes in, FS hands it over to lua script for processing.****
>
> ** **
>
> <include>****
>
> <extension>****
>
>   <condition field="destination_number" expression="^(15[0-9][0-9])$">****
>
>    <action application="lua" data="fs_billing.lua"/>****
>
>   </condition>****
>
> </extension>****
>
> </include>****
>
> ** **
>
> I did the same thing with ESL where I listened for CHANNEL_ANSWER and
> CHANNEL_HANGUP events then fetched answer and hangup times and that worked
> just fine. I would like to do the same thing with lua. ****
>
> ** **
>
> Thanks****
>
> ** **
>
>
>

-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From steveayre at gmail.com  Sun Aug 18 20:18:13 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Sun, 18 Aug 2013 17:18:13 +0100
Subject: [Freeswitch-users] something related to channel variables
In-Reply-To: <CAKQo6tkGjoTQU0bbEKo4uMjCrOW5v7sUy4nUJm7-sTjWq-bfRw@mail.gmail.com>
References: <CAKQo6tkGjoTQU0bbEKo4uMjCrOW5v7sUy4nUJm7-sTjWq-bfRw@mail.gmail.com>
Message-ID: <CAFiqYu=1BwVki1f-J-fPmDX5D4HnK5OY-f0eXyTFz0oE8JoD+w@mail.gmail.com>

sip_contact_params
sip_contact_user
sip_contact_port
sip_contact_uri
sip_contact_host

If you're looking for something in the variables try checking
http://wiki.freeswitch.org/wiki/Channel_Variables or
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_info


On 17 August 2013 11:45, Vincent Xia <gmangudai at gmail.com> wrote:

>
> how to get the contact field of the sip message "invite" or "200 ok" in
> the dialplan xml?
>
> how to get the registering ip address of an endpoint in the dialplan xml
> (as we have sofia_contact in the command line)?
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sos at sokhapkin.dyndns.org  Sun Aug 18 20:26:43 2013
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Sun, 18 Aug 2013 12:26:43 -0400
Subject: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua
In-Reply-To: <002601ce9c2e$190e2870$4b2a7950$@verizon.net>
References: <00fc01ce9b7c$8b9aa1a0$a2cfe4e0$@verizon.net>
	<2D04E07F-95B8-48AC-852B-65DEC193CEA1@freeswitch.org>
	<002601ce9c2e$190e2870$4b2a7950$@verizon.net>
Message-ID: <1709460.UbJ5LCPbry@sos>

Look at mod_nibblebill. It already does what you want and perfectly handles 
concurrent calls from the same subscriber.

On Sunday 18 August 2013 11:15:04 Sami Montour wrote:
> The billing I would like to use with lua is for real-time billing where
> subscribers are debited real-time or at the end of the call.
> 
> 
> 
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken
> Rice Sent: Sunday, August 18, 2013 10:55 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua
> 
> 
> 
> billing in lua is asking for issues... why not do something like triggiering
> the billing from the xml cdr or such this avoids loss od cdrs as it uses
> the filesystem as a backup to incase your websvices or billing processor is
> down
> 
> Ken
> 
> Sent from my iPad
> 
> 
> On Aug 18, 2013, at 10:19, Karsten Horsmann <khorsmann at gmail.com> wrote:
> 
> Hi Sam,
> 
> 
> 
> you should be more verbose what you currently doing. There a many ways to
> call an lua script in FS (in the dialplan, as startup-script, as
> dialplan-binding etc).
> 
> Sounds like you want to fetch some kind of billing informations.
> 
> 
> 
> 2013/8/17 Sami Montour <smontour at verizon.net>
> 
> I am using Mod lua for call processing on the FreeSwitch. I would like to
> wait for some events such as CHANNEL_ANSWER and CHANNEL_HANGUP  in order to
> get the answer time and hangup time. Basically, something like
> 
> 
> 
> Wait for event CHANNEL_ANSWER or CHANNEL_HANGUP
> 
> If (event_name = ?CHANNEL_ANSWER?) then answer_timestamp =
> getHeader(?Caller-Channel-Answered-Time?)
> 
> If (event_name = ?CHANNEL_HANGUP?) then hangup_timestamp =
> getHeader(?Event-Date-Timestamp?)
> 
> 
> 
> The session:getState() described in module lua gets the call state but not
> the channel state. For instance, ?CS_EXECUTE? is the channel state when it
> is executing a dialplan not when the call is answered.
> 
> 
> 
> Is there a way to wait for channel events such as CHANNEL_ANSWER and
> CHANNEL_HANGUP in order to retrieve answer and hangup timestamps for a
> call?
> 
> 
> 
> Any input is very much appreciated.
> 
> 
> 
> Thanks.
> 
> 
> 
> Sam


From krice at freeswitch.org  Sun Aug 18 20:28:54 2013
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 18 Aug 2013 11:28:54 -0500
Subject: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua
In-Reply-To: <002601ce9c2e$190e2870$4b2a7950$@verizon.net>
References: <00fc01ce9b7c$8b9aa1a0$a2cfe4e0$@verizon.net>
	<CAFArqsaVJrxcQgPNMFrtigBtc90-YJMDyRkeOgYw1MSLr_h5ww@mail.gmail.com>
	<2D04E07F-95B8-48AC-852B-65DEC193CEA1@freeswitch.org>
	<002601ce9c2e$190e2870$4b2a7950$@verizon.net>
Message-ID: <CF857FEA-B1A8-4144-8DCC-FCE97F9AE7C5@freeswitch.org>

realtime billing can be accomplished with nibble whichs deducts as the call is in progress... folowed up with bill validation via the cdr. if your lua script crashes no billing...

Ken
Sent from my iPad

On Aug 18, 2013, at 11:15, "Sami Montour" <smontour at verizon.net> wrote:

> The billing I would like to use with lua is for real-time billing where subscribers are debited real-time or at the end of the call.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
> Sent: Sunday, August 18, 2013 10:55 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua
>  
> billing in lua is asking for issues... why not do something like triggiering the billing from the xml cdr or such this avoids loss od cdrs as it uses the filesystem as a backup to incase your websvices or billing processor is down
> 
> Ken
> Sent from my iPad
> 
> On Aug 18, 2013, at 10:19, Karsten Horsmann <khorsmann at gmail.com> wrote:
> 
> Hi Sam,
>  
> you should be more verbose what you currently doing. There a many ways to call an lua script in FS (in the dialplan, as startup-script, as dialplan-binding etc).
> 
> Sounds like you want to fetch some kind of billing informations.
>  
> 2013/8/17 Sami Montour <smontour at verizon.net>
> I am using Mod lua for call processing on the FreeSwitch. I would like to wait for some events such as CHANNEL_ANSWER and CHANNEL_HANGUP  in order to get the answer time and hangup time. Basically, something like
>  
> Wait for event CHANNEL_ANSWER or CHANNEL_HANGUP
> If (event_name = ?CHANNEL_ANSWER?) then answer_timestamp = getHeader(?Caller-Channel-Answered-Time?)
> If (event_name = ?CHANNEL_HANGUP?) then hangup_timestamp = getHeader(?Event-Date-Timestamp?)
>  
> The session:getState() described in module lua gets the call state but not the channel state. For instance, ?CS_EXECUTE? is the channel state when it is executing a dialplan not when the call is answered.
>  
> Is there a way to wait for channel events such as CHANNEL_ANSWER and CHANNEL_HANGUP in order to retrieve answer and hangup timestamps for a call?
>  
> Any input is very much appreciated.
>  
> Thanks.
>  
> Sam
>  
>  
> 
>  
> -- 
> Mit freundlichen Gr??en
> *Karsten Horsmann*
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 smontour at verizon.net  Sun Aug 18 21:11:56 2013
From: smontour at verizon.net (Sami Montour)
Date: Sun, 18 Aug 2013 12:11:56 -0500
Subject: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua
In-Reply-To: <CF857FEA-B1A8-4144-8DCC-FCE97F9AE7C5@freeswitch.org>
References: <00fc01ce9b7c$8b9aa1a0$a2cfe4e0$@verizon.net>
	<CAFArqsaVJrxcQgPNMFrtigBtc90-YJMDyRkeOgYw1MSLr_h5ww@mail.gmail.com>
	<2D04E07F-95B8-48AC-852B-65DEC193CEA1@freeswitch.org>
	<002601ce9c2e$190e2870$4b2a7950$@verizon.net>
	<CF857FEA-B1A8-4144-8DCC-FCE97F9AE7C5@freeswitch.org>
Message-ID: <004301ce9c36$0b227e20$21677a60$@verizon.net>

Thank you guys for your input. I will look into nibble bill module and try it. It seems the module is designed for real-time billing.

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Sunday, August 18, 2013 11:29 AM
To: FreeSWITCH Users Help
Cc: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua

 

realtime billing can be accomplished with nibble whichs deducts as the call is in progress... folowed up with bill validation via the cdr. if your lua script crashes no billing...

Ken

Sent from my iPad


On Aug 18, 2013, at 11:15, "Sami Montour" <smontour at verizon.net> wrote:

The billing I would like to use with lua is for real-time billing where subscribers are debited real-time or at the end of the call.

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Sunday, August 18, 2013 10:55 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Wait for ANSWER/HANGUP events in Mod lua

 

billing in lua is asking for issues... why not do something like triggiering the billing from the xml cdr or such this avoids loss od cdrs as it uses the filesystem as a backup to incase your websvices or billing processor is down

Ken

Sent from my iPad


On Aug 18, 2013, at 10:19, Karsten Horsmann <khorsmann at gmail.com> wrote:

Hi Sam,

 

you should be more verbose what you currently doing. There a many ways to call an lua script in FS (in the dialplan, as startup-script, as dialplan-binding etc).

Sounds like you want to fetch some kind of billing informations.

 

2013/8/17 Sami Montour <smontour at verizon.net>

I am using Mod lua for call processing on the FreeSwitch. I would like to wait for some events such as CHANNEL_ANSWER and CHANNEL_HANGUP  in order to get the answer time and hangup time. Basically, something like

 

Wait for event CHANNEL_ANSWER or CHANNEL_HANGUP

If (event_name = ?CHANNEL_ANSWER?) then answer_timestamp = getHeader(?Caller-Channel-Answered-Time?) 

If (event_name = ?CHANNEL_HANGUP?) then hangup_timestamp = getHeader(?Event-Date-Timestamp?)

 

The session:getState() described in module lua gets the call state but not the channel state. For instance, ?CS_EXECUTE? is the channel state when it is executing a dialplan not when the call is answered. 

 

Is there a way to wait for channel events such as CHANNEL_ANSWER and CHANNEL_HANGUP in order to retrieve answer and hangup timestamps for a call?

 

Any input is very much appreciated.

 

Thanks.

 

Sam

 

 




 

-- 
Mit freundlichen Gr??en
*Karsten Horsmann*

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




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http://wiki.freeswitch.org
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From fdelawarde at wirelessmundi.com  Sun Aug 18 22:19:09 2013
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?=)
Date: Sun, 18 Aug 2013 20:19:09 +0200
Subject: [Freeswitch-users] Issue when receiving INVITE with both audio
 and image (T38)
In-Reply-To: <CAKbxfG_qKgrng5Fc29EBJzvNkkrhh_az6c5i2AjuSxna77vwJA@mail.gmail.com>
References: <1376661366.3068.47.camel@luna.madrid.commsmundi.com>
	<CAKbxfG8gO9pg_8umr591Ejjgt80txYgy=fCfpQxfkJG9g4O0MA@mail.gmail.com>
	<1376669353.3068.67.camel@luna.madrid.commsmundi.com>
	<CAKbxfG_qKgrng5Fc29EBJzvNkkrhh_az6c5i2AjuSxna77vwJA@mail.gmail.com>
Message-ID: <1376849949.27726.6.camel@salon.delawarde.com>

There you go:

http://jira.freeswitch.org/browse/FS-5714

Thanks,
Fran?ois.

On Fri, 2013-08-16 at 13:01 -0500, Anthony Minessale wrote:

> Sure
> 
> 
> On Aug 16, 2013 11:14 AM, "Fran?ois" <fdelawarde at wirelessmundi.com>
> wrote:
> 
>         Thanks for the tip. Should I add a JIRA ticket?
>         
>         Fran?ois.
>         
>         On Fri, 2013-08-16 at 10:00 -0500, Anthony Minessale wrote:
>         > We can investigate that,  Meanwhile you could use regex in
>         your
>         > dialplan for this situation and set
>         "sip_ignore_reinvites=true" to
>         > mitigate.
>         >
>         >
>         >
>         >
>         > On Fri, Aug 16, 2013 at 8:56 AM, Fran?ois
>         > <fdelawarde at wirelessmundi.com> wrote:
>         >         Hello list,
>         >
>         >         I've seen several gateways (Patton, Audiocodes,
>         Nortel CS2K),
>         >         sending
>         >         both audio+image (T38) in the initial INVITE. In a
>         simple
>         >         bridge
>         >         scenario, FS will just ignore the T38 part and setup
>         an audio
>         >         connection
>         >         only with b-leg.
>         >
>         >         My issue is later on if/when we receive a re-INVITE
>         (ex: SIP
>         >         timers
>         >         refresh) with the same SDP (audio+T38), FS now
>         considers it as
>         >         FAX and
>         >         sends a re-INVITE to b-leg with T38 only! In case
>         the
>         >         destination is
>         >         voice only, the call hangs up.
>         >
>         >         ---
>         >
>         >         Asterisk apparently had the same issue, they solved
>         it by
>         >         setting up
>         >         both streams on a-leg (RTP + UDPTL). If they receive
>         packets
>         >         on UDPTL
>         >         they just re-INVITE b-leg with T38.
>         >
>         >         See: https://reviewboard.asterisk.org/r/208/
>         >         Also:
>         https://issues.asterisk.org/jira/browse/ASTERISK-11843
>         >
>         >         Is this the right way to go? (Also is it feasible in
>         >         FreeSwitch)? I can
>         >         add a JIRA if necessary.
>         >
>         >         Thanks,
>         >         Fran?ois.
>         >
>         >
>         >
>         _________________________________________________________________________
>         >         Professional FreeSWITCH Consulting Services:
>         >         consulting at freeswitch.org
>         >         http://www.freeswitchsolutions.com
>         >
>         >         FreeSWITCH-powered IP PBX: The CudaTel Communication
>         Server
>         >         
>         >
>         >         Official FreeSWITCH Sites
>         >         http://www.freeswitch.org
>         >         http://wiki.freeswitch.org
>         >         http://www.cluecon.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
>         >
>         _________________________________________________________________________
>         > Professional FreeSWITCH Consulting Services:
>         > consulting at freeswitch.org
>         > http://www.freeswitchsolutions.com
>         >
>         > 
>         > 
>         >
>         > Official FreeSWITCH Sites
>         > http://www.freeswitch.org
>         > http://wiki.freeswitch.org
>         > http://www.cluecon.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
>         
>         
>         
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org
>         http://www.freeswitchsolutions.com
>         
>         
>         
>         
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 moises.silva at gmail.com  Mon Aug 19 02:41:08 2013
From: moises.silva at gmail.com (Moises Silva)
Date: Sun, 18 Aug 2013 18:41:08 -0400
Subject: [Freeswitch-users] Ademco signaling with mod_spandsp
In-Reply-To: <CAHgk4BXwkQdwmCX7y=K8Vv03m2U6xpnO0gTCbKgKyRPdFmr0eg@mail.gmail.com>
References: <CAHgk4BXwkQdwmCX7y=K8Vv03m2U6xpnO0gTCbKgKyRPdFmr0eg@mail.gmail.com>
Message-ID: <CAA4nhyAWrzt6cjncAJHiXty37MS-twvm2AYJOHxpiwUhdSRjTA@mail.gmail.com>

It is only present on the spandsp library, but no application or component
in FreeSWITCH makes use of it. You'd require to write some C code to
integrate it.

*Moises Silva
**Manager, Software Engineering***

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



**<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>

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On Fri, Aug 16, 2013 at 12:52 PM, Brian Foster <bdfoster at davri.com> wrote:

> I've been tasked with securing a storage facility with an alarm system.
> Basically we'd use POTS or GSM on the site and a POTS line on the
> monitoring station. The alarm system is capable of using the ademco
> standard. I see there's some code in git in regards to the ademco standard.
> Is this usable inside Freeswitch? If so, how can I access it? I'd like to
> pay the wiki tax to doxument it if anyone can get me started.
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana
>
> Sent from a mobile device.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Mon Aug 19 03:15:23 2013
From: eidevm5 at gmail.com (Peter)
Date: Mon, 19 Aug 2013 09:15:23 +1000
Subject: [Freeswitch-users] Softphone not getting connected
In-Reply-To: <CAMvhjJs_YqJB6q11ADhKiPY-OdAiGZ9euzjja45VoXtwcycBXw@mail.gmail.com>
References: <CAMvhjJvSmpPJt2E6H-Xi2HbRiRvsvmuv-y9xuvx16aawgh+dcg@mail.gmail.com>
	<CABkS=xpdopCj8iihOqupYYH9XWqaj32gfhqwspd5gm0HRu6D0w@mail.gmail.com>
	<CAMvhjJs8JRnB3hCYFxoW7muc7xG4XTkM+4_j0MFxh=yHCrNYvg@mail.gmail.com>
	<CABkS=xo_utAEwsz4cNW8O0B+AYNpAiSGJfuP+y2fSbE1wNJdow@mail.gmail.com>
	<CAMvhjJsBWo9AejesvNH6PXAEUhzpZMqeLm-k425d3KhdCygAaQ@mail.gmail.com>
	<CAMvhjJsusoLzBAjigwiVsh0Ppn+a23ec6mUG7VVzWONenSa3Xw@mail.gmail.com>
	<CABkS=xpjHwqBPWb1JJNT4LCeviqyXhbs96FrB6=GxW9UyQ+D1w@mail.gmail.com>
	<CALn233ovuLR8rgD==bqiOi+Fa=yDwS8azsrD3cWkBz9sz7Mw7w@mail.gmail.com>
	<CAMvhjJut4W_hwnAG45o2wFWCOKknwPzE8vFGrkNv6JoZZA8o1Q@mail.gmail.com>
	<CABkS=xorE-TTwdKb=m0pO-C4stk4JC5KBJGcpbrhpFhdeMYKbQ@mail.gmail.com>
	<CAMvhjJs_YqJB6q11ADhKiPY-OdAiGZ9euzjja45VoXtwcycBXw@mail.gmail.com>
Message-ID: <CABkS=xq4AMkedk_TpLts3i3SXRL0V7mcDfBJ+nfV-7MZiMVOYQ@mail.gmail.com>

Take your pick of Linux SIP clients from

https://en.wikipedia.org/wiki/List_of_SIP_software#Clients


On Fri, Aug 16, 2013 at 3:54 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Any recommendations for a softphone that works on ubuntu linux. As xlite
> will not work on linux.
> Thanks
> Ashish Mishra
> On Aug 16, 2013 4:55 AM, "Peter" <eidevm5 at gmail.com> wrote:
>
>> Of course you can use a softphone on the freeswitch server.   It would
>> eliminate any network issues.
>>
>>
>> On Thu, Aug 15, 2013 at 7:12 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> I also think that there may be some network problem as my windows pc is
>>> not showing that some linux machine is connected but the linux machine
>>> shows the windows system connected to it. Also can i install the softphone
>>> on the same linux machine on which my freeswitch runs and can check whether
>>> it gets connected or not...???
>>>  On Aug 15, 2013 5:10 AM, "Mitch Capper" <mitch.capper at gmail.com> wrote:
>>>
>>>> You may want to try FSClient its a freeswitch based windows soft phone
>>>> and you can connect fs_cli (or fs_logger.pl) to it and see what
>>>> happens when it tries to connect to your server.
>>>>
>>>> ~mitch
>>>>
>>>>
>>>> On Wed, Aug 14, 2013 at 4:06 PM, Peter <eidevm5 at gmail.com> wrote:
>>>>
>>>>> My mistake, I meant loglevel rather than logging.
>>>>>
>>>>> So when you say "no progress" are you seeing nothing at all on the FS
>>>>> console?
>>>>>
>>>>> If so, then that indicates a network problem as FS isn't even seeing
>>>>> the registration requests.
>>>>>
>>>>> Can you ping the windows box from the FS server?
>>>>>
>>>>>
>>>>> On Wed, Aug 14, 2013 at 7:04 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>>
>>>>>> I went for "sofia loglevel 5" instead of "sofia logging 5" but still
>>>>>> no progress in connection of softphone...:-(
>>>>>> Thanks
>>>>>> Ashish Mishra
>>>>>> On Aug 14, 2013 2:14 PM, "Ashish Mishra" <itsme.kunnu at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Also the command that you had mentioned "sofia logging all 5" gives
>>>>>>> me an error message :
>>>>>>> Unknown command [logging]
>>>>>>> On Aug 14, 2013 7:24 AM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>>>>
>>>>>>>> Connect to the FS console with fs_cli and do
>>>>>>>>
>>>>>>>> sofia logging all 5
>>>>>>>>
>>>>>>>> then try to register your softphone.    If you don't even see a
>>>>>>>> registration attempt on the FS console, it means you have a network problem.
>>>>>>>>
>>>>>>>> If the network isn't a problem, the debug output should give you a
>>>>>>>> clue as to what the problem is.
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Tue, Aug 13, 2013 at 5:34 PM, Ashish Mishra <
>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Thank you Peter...you mean to say that i should first turn on the
>>>>>>>>> fs_cli and then retry to connect the softphone...??? I would also like to
>>>>>>>>> remind you that fs_cli and freeswitch are running on the same machine and i
>>>>>>>>> am trying to connect thru a network cable my windows 8 pc that has the
>>>>>>>>> softphone on it...
>>>>>>>>> Regards
>>>>>>>>> Ashish Mishra
>>>>>>>>> On Aug 13, 2013 1:00 PM, "Peter" <eidevm5 at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Have you turned on debugging from the FS cli and seen if any
>>>>>>>>>> registration requests come in?
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Tue, Aug 13, 2013 at 5:13 PM, Ashish Mishra <
>>>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> I installed freeswitch on my ubuntu 12.04 machine...but when i
>>>>>>>>>>> am trying to connect the softphone installed on my windows 8 pc with ubuntu
>>>>>>>>>>> machine the softphone gives me an error that account not enabled...i have
>>>>>>>>>>> used a network cable to connect the two machines...also the firewall in
>>>>>>>>>>> both the machines is disabled...
>>>>>>>>>>> Kindly help..
>>>>>>>>>>>
>>>>>>>>>>> Ashish Mishra
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>>
>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.com
>>>>>
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>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.com
>>>>
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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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.com
>>>
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
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>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
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>> http://www.cluecon.com
>>
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>> http://www.freeswitch.org
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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From bdfoster at davri.com  Mon Aug 19 04:18:41 2013
From: bdfoster at davri.com (Brian Foster)
Date: Sun, 18 Aug 2013 20:18:41 -0400
Subject: [Freeswitch-users] Ademco signaling with mod_spandsp
In-Reply-To: <CAA4nhyAWrzt6cjncAJHiXty37MS-twvm2AYJOHxpiwUhdSRjTA@mail.gmail.com>
References: <CAHgk4BXwkQdwmCX7y=K8Vv03m2U6xpnO0gTCbKgKyRPdFmr0eg@mail.gmail.com>
	<CAA4nhyAWrzt6cjncAJHiXty37MS-twvm2AYJOHxpiwUhdSRjTA@mail.gmail.com>
Message-ID: <CAHgk4BX7jP-M=GLxcCSXaJ9y-SVQYg5Dkf8Qza-s66b1B00fJw@mail.gmail.com>

It's probably easier to write something in LUA to handle this. Seems pretty
straightforward looking at the standard. Might be wrong though.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
It is only present on the spandsp library, but no application or component
in FreeSWITCH makes use of it. You'd require to write some C code to
integrate it.

 *Moises Silva
**Manager, Software Engineering***

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



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On Fri, Aug 16, 2013 at 12:52 PM, Brian Foster <bdfoster at davri.com> wrote:

> I've been tasked with securing a storage facility with an alarm system.
> Basically we'd use POTS or GSM on the site and a POTS line on the
> monitoring station. The alarm system is capable of using the ademco
> standard. I see there's some code in git in regards to the ademco standard.
> Is this usable inside Freeswitch? If so, how can I access it? I'd like to
> pay the wiki tax to doxument it if anyone can get me started.
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana
>
> Sent from a mobile device.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

FreeSWITCH-users mailing list
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From nandy1925 at gmail.com  Mon Aug 19 07:54:16 2013
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Mon, 19 Aug 2013 11:54:16 +0800
Subject: [Freeswitch-users] Transfer / Forward calls - from an operator
 to an extension after the call is picked
In-Reply-To: <1376827283.11657.YahooMailNeo@web160501.mail.bf1.yahoo.com>
References: <1376827283.11657.YahooMailNeo@web160501.mail.bf1.yahoo.com>
Message-ID: <CAHKqMb5SK8V5h-CYf8uaivj_LkvMR-xjQ1k+gG7UA_0z_5g44w@mail.gmail.com>

Hi,

1. Copy your working dialplan entry.
2. Add <condition field="caller_id_number" data="4302000"/>
3. Change the bridge/transfer application to connect to the operator

Take note to place the above entry ahead of your working entry. I hope this
helps.
/Nandy


On Sun, Aug 18, 2013 at 8:01 PM, Ravi <sravi123 at yahoo.com> wrote:

> Hello Everyone !
>
> I have successfully configured my PRI connection. I am able to send and
> receive calls. Now I am trying to have an operator handle one number - say
> 4302000. When any call is received the operator will talk to the caller and
> need to transfer the call to an extension. I tried looking at dialplan
> documentation, and I think I find only instances where the transfer is done
> within the program and not when some one picks up the call.
>
> Can some one please point me to right place/ documents where I can find
> information regarding this.
>
> I did find a link with some similar request, but the information is
> limited.
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2012-October/088579.html
>
> Thanks for your help.
> Ravi
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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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From moises.silva at gmail.com  Mon Aug 19 07:54:46 2013
From: moises.silva at gmail.com (Moises Silva)
Date: Sun, 18 Aug 2013 23:54:46 -0400
Subject: [Freeswitch-users] Ademco signaling with mod_spandsp
In-Reply-To: <CAHgk4BX7jP-M=GLxcCSXaJ9y-SVQYg5Dkf8Qza-s66b1B00fJw@mail.gmail.com>
References: <CAHgk4BXwkQdwmCX7y=K8Vv03m2U6xpnO0gTCbKgKyRPdFmr0eg@mail.gmail.com>
	<CAA4nhyAWrzt6cjncAJHiXty37MS-twvm2AYJOHxpiwUhdSRjTA@mail.gmail.com>
	<CAHgk4BX7jP-M=GLxcCSXaJ9y-SVQYg5Dkf8Qza-s66b1B00fJw@mail.gmail.com>
Message-ID: <CAA4nhyAOzPOpg5N+qsKCSDhuFZaX2AyZwW4F5bXJgtOa4Zr1wA@mail.gmail.com>

On Sun, Aug 18, 2013 at 8:18 PM, Brian Foster <bdfoster at davri.com> wrote:

> It's probably easier to write something in LUA to handle this. Seems
> pretty straightforward looking at the standard. Might be wrong though.
>
> Although not rocket-science, I believe you might be underestimating it a
bit :-)

I have not looked at the standard and have no experience at all with the
protocol, but the code is in spandsp for a reason, there are some
per-sample digital audio calculations, tone/silence detection etc, with
strict rules/timing that most likely you won't be able to do from LUA.

See libs/spandsp/src/ademco_contactid.c comments

I believe the right approach is to write a mod_ademco wrapper that exposes
the functionality to the dialplan (either through dialplan applications or
functions), then you can write LUA scripts :-)

Moy

*Moises Silva
**Manager, Software Engineering***

msilva at sangoma.com

Sangoma Technologies

100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada


t.   +1 800 388 2475 (N. America)

t.   +1 905 474 1990 x128

f.   +1 905 474 9223



**<http://www.sangoma.com/contact?utm_source=signature&utm_medium=email&utm_campaign=email+signatures>

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From babak.freeswitch at gmail.com  Mon Aug 19 09:49:14 2013
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Mon, 19 Aug 2013 10:19:14 +0430
Subject: [Freeswitch-users] setting transfer context
Message-ID: <CAEVfcaaVmePZ3YB1GYvWK+7qAKrMFR0DYsVY0Zq2343mdE0XdQ@mail.gmail.com>

hi
when a call is attended transferred, for example, to a number like 222,
first bleg is hold and caller initiates a new channel with destination 222
which hits default context. if this new call is transferred to another
context like public using transfer application with a different destination
number like 333 after attended transfer completes the bleg is dropped.
because it goes in routing state with destination number 333 in default
dialplan. to solve this problem I'm using an extension in all contexts
after unloop extension:

<extension name="transfer_adjustment">
  <condition field="${transfer_adjusted}" expression="^$"/>
      <condition field="${sip_refer_to}" expression="^<sip:(.+)@.*$">
<action application="log" data="DEBUG original tranfer was:$1"/>
<action application="set" data="transfer_adjusted=true"/>
<action application="transfer" data="$1 ${dialplan} ${context}"/>
      </condition>
</extension>

I check if transfer_adjusted is not set and the channel is
transferred,first and second conditions, then I transfer it to the original
number which caller called to transfer the call to correct destination. Is
this the right way? or I'm missing something
thanks
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From eidevm5 at gmail.com  Mon Aug 19 10:02:07 2013
From: eidevm5 at gmail.com (Peter)
Date: Mon, 19 Aug 2013 16:02:07 +1000
Subject: [Freeswitch-users] Call not hanging up
Message-ID: <CABkS=xoKBtF6zP4BzWb8tiNzpK2N-yLM6F1Jqhfz26fPcnO34Q@mail.gmail.com>

Running FS 1.2.12 in the following environment:

FS Ext  <--->  FS SBC <---> FS Int

I can successfully make calls in both directions.

If the caller hangs up, the other end correctly hangs up.

However, if the callee hangs up, the caller session remains active until
the RTP timeout kicks in.

Here's a snippet from the FS log where this occurs (1004 is the caller and
2010 is the callee)

[NOTICE] sofia.c:716 Hangup
sofia/internal/2010 at 10.1.1.206:5061[CS_HIBERNATE] [NORMAL_CLEARING]
[NOTICE] switch_ivr_bridge.c:1109 Hangup
sofia/internal/1004 at 10.1.1.204[CS_HIBERNATE] [NORMAL_CLEARING]
[NOTICE] switch_core_session.c:1560 Session 4 (sofia/internal/
2010 at 10.1.1.206:5061) Ended
[NOTICE] switch_core_session.c:1564 Close Channel sofia/internal/
2010 at 10.1.1.206:5061 [CS_DESTROY]
[NOTICE] switch_core_session.c:1560 Session 3 (sofia/internal/
1004 at 10.1.1.104) Ended
[NOTICE] switch_core_session.c:1564 Close Channel sofia/internal/
1004 at 10.1.1.104 [CS_DESTROY]

I've tried setting hangup_after_bridge=true in the dialplan on FS Ext.    I
wasn't 100% sure whether it should go on FS Ext or FS SBC, so I tried both
and it made no difference.

What else should I be checking?

Thanks

Peter
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From andrew at cassidywebservices.co.uk  Mon Aug 19 12:25:55 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Mon, 19 Aug 2013 09:25:55 +0100
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <521017FE.7070106@xtronics.com>
References: <CE33869E.78FAE%krice@freeswitch.org>
	<520E25CB.7030404@puzzled.xs4all.nl>
	<00000140874a3df1-e0a86002-7ad9-485d-b86b-200a83300469-000000@email.amazonses.com>
	<521017FE.7070106@xtronics.com>
Message-ID: <CAMbW9oJ3SG=n0-RdFdtMkksEVOQ-F82KdwvdCiTVh0K=e_7MMA@mail.gmail.com>

Good stuff, that's pretty useful to me too. Like I say, the newer Cisco
SPA50x's are nice, the old 79xx series are horrible for SIP (as someone
else mentioned)

Yealink have been mentioned, has anyone got any thoughts on those? I've
been thinking about investigating them.


On 18 August 2013 01:40, Karl Schmidt <karl at xtronics.com> wrote:

> On 08/16/2013 08:22 AM, Avi Marcus wrote:
> > Just btw -- this is hardly the first time this question has been asked,
> and nothing has been
> > summarized on the wiki for future reference (that I know of).
> > So, someone that has the time....
> >
> > -Avi
>
> I'm actually summarizing the response here (as it seemed a bit off topic
> re freeswitch) :
>
> http://wiki.xtronics.com/index.php/SIP_Phones
>
> Work in processes - feel free grab and post on wiki.freeswitch.org
>
>
>
>
>
>
> --------------------------------------------------------------------------------
> Karl Schmidt                                  EMail Karl at xtronics.com
> Transtronics, Inc.                              WEB
> http://secure.transtronics.com
> 3209 West 9th Street                             Ph (785) 841-3089
> Lawrence, KS 66049                              FAX (785) 841-0434
>
> The society that puts equality before freedom will
> end up with neither. The society that puts freedom
> before equality will end up with a great measure of both.
> - Milton Freidman
>
> --------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
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From steveu at coppice.org  Mon Aug 19 15:47:36 2013
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 19 Aug 2013 19:47:36 +0800
Subject: [Freeswitch-users] Ademco signaling with mod_spandsp
In-Reply-To: <CAA4nhyAOzPOpg5N+qsKCSDhuFZaX2AyZwW4F5bXJgtOa4Zr1wA@mail.gmail.com>
References: <CAHgk4BXwkQdwmCX7y=K8Vv03m2U6xpnO0gTCbKgKyRPdFmr0eg@mail.gmail.com>
	<CAA4nhyAWrzt6cjncAJHiXty37MS-twvm2AYJOHxpiwUhdSRjTA@mail.gmail.com>
	<CAHgk4BX7jP-M=GLxcCSXaJ9y-SVQYg5Dkf8Qza-s66b1B00fJw@mail.gmail.com>
	<CAA4nhyAOzPOpg5N+qsKCSDhuFZaX2AyZwW4F5bXJgtOa4Zr1wA@mail.gmail.com>
Message-ID: <521205D8.1010705@coppice.org>

On 08/19/2013 11:54 AM, Moises Silva wrote:
> On Sun, Aug 18, 2013 at 8:18 PM, Brian Foster <bdfoster at davri.com 
> <mailto:bdfoster at davri.com>> wrote:
>
>     It's probably easier to write something in LUA to handle this.
>     Seems pretty straightforward looking at the standard. Might be
>     wrong though.
>
> Although not rocket-science, I believe you might be underestimating it 
> a bit :-)
>
> I have not looked at the standard and have no experience at all with 
> the protocol, but the code is in spandsp for a reason, there are some 
> per-sample digital audio calculations, tone/silence detection etc, 
> with strict rules/timing that most likely you won't be able to do from 
> LUA.
>
> See libs/spandsp/src/ademco_contactid.c comments
>
> I believe the right approach is to write a mod_ademco wrapper that 
> exposes the functionality to the dialplan (either through dialplan 
> applications or functions), then you can write LUA scripts :-)
>
> Moy
Moises is right. You can get close to an Ademco receiver without special 
code, but you won't really meet the detailed timing of the spec without 
doing a proper job. You won't get anywhere trying to make a decent 
Ademco sender without specialised DSP code. spandsp implements both 
sides properly. Why not use it?

Steve


From royce3 at gmail.com  Mon Aug 19 16:59:40 2013
From: royce3 at gmail.com (Royce Mitchell III)
Date: Mon, 19 Aug 2013 07:59:40 -0500
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <CAFiqYunDhXi+v1H4P=vQnrGW2WRd61w1=nnGwVPo+Pv9CWG3Dw@mail.gmail.com>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
	<CAFiqYumBFafVazhjXUf1CdXgVaYqvbU4QzrPEc2svCLJu0BaFw@mail.gmail.com>
	<CAHwzD9n+CfCbCYNvwPcSiwYQoQe9bved3uK6q1=okh+2ravZoA@mail.gmail.com>
	<A46F2C5A-8082-4155-A4E0-96CE65AF3431@insensate.co.uk>
	<CAHwzD9ncYxxJxaMYnu2qx78Q_SSLp3onz+5=kN=DSz+vRVQGKA@mail.gmail.com>
	<CAFiqYunDhXi+v1H4P=vQnrGW2WRd61w1=nnGwVPo+Pv9CWG3Dw@mail.gmail.com>
Message-ID: <CAHwzD9=gpZjsuVxdj3O04PMPCh_mqoxj8SudvXak=ejWZJiohw@mail.gmail.com>

Here is one example, thanks



Royce Mitchell, IT Consultant
ITAS Solutions
royce3 at itas-solutions.com


On Fri, Aug 16, 2013 at 4:12 PM, Steven Ayre <steveayre at gmail.com> wrote:

> Can you paste a debug log of the entire lifetime of the call?
>
>
> On 16 August 2013 21:57, Royce Mitchell III <royce3 at gmail.com> wrote:
>
>> The only thing I'm doing in FS regarding codecs is I force PCMU in
>> certain conditions ( ran into a transcoding bug in FreeSwitch between G722
>> HD and G711 with the Polycom phones )
>>
>>
>>
>> Royce Mitchell, IT Consultant
>> ITAS Solutions
>> royce3 at itas-solutions.com
>>
>>
>> On Fri, Aug 16, 2013 at 3:35 PM, Lawrence Conroy <lconroy at insensate.co.uk
>> > wrote:
>>
>>> Hi there,
>>>  Are you sure about fS allowing PCMU?
>>> According to the remote SDP, your incall is proposing PCMU as its top
>>> choice; g729 is 2nd best (quite right too :).
>>> Forcing the adtran to offer only PCMU should not make ay difference to
>>> that -- it'll still propose PCMU so no change.
>>> Either your setup is somehow blocking PCMU on the b-leg (but I'd expect
>>> to see that on the log), OR is trying to transcode because the b-leg
>>> requires some (non-PCMU) codec and can't (again, I'd expect that to be
>>> logged), OR fS is not accepting PCMU.
>>> Assuming that PCMU is in the fS vars codec lists, does your dialplan do
>>> anything funky with the codec list for an incall?
>>> all the best,
>>>   Lawrence
>>>
>>> On 16 Aug 2013, at 21:06, Royce Mitchell III wrote:
>>> > My FreeSWITCH is configured to prefer PCMU, and the devices it is
>>> talking
>>> > to are Adtran 908e's. The Adtrans are configured for the default codec
>>> > group which is supposed to be PCMU, but I can reconfigure them to
>>> > explicitly allow only PCMU. I will try that and see if it makes a
>>> > difference.
>>> >
>>> >
>>> >
>>> > Royce Mitchell, IT Consultant
>>> > ITAS Solutions
>>> > royce3 at itas-solutions.com
>>> >
>>> >
>>> > On Fri, Aug 16, 2013 at 11:04 AM, Steven Ayre <steveayre at gmail.com>
>>> wrote:
>>> >
>>> >> INCOMPATIBLE_DESTINATION means a codec problem.
>>> >>
>>> >> The remote SDP sends they're offerring PCMU and G729.
>>> >>
>>> >> What codecs are you allowing, what codecs are you bridging with, and
>>> since
>>> >> G729 is on the list are you perhaps trying to transcode without using
>>> >> mod_com_g729+licenses?
>>> >>
>>> >>
>>> >> On 16 August 2013 16:07, Royce Mitchell III <royce3 at gmail.com> wrote:
>>> >>
>>> >>> Hi,
>>> >>>
>>> >>> My client has been running FreeSWITCH with low call volume and has
>>> been
>>> >>> stable for several months now. This Tuesday morning we switched all
>>> call
>>> >>> volume over to FreeSWITCH and everything was running fine until this
>>> >>> morning.
>>> >>>
>>> >>> Out of the blue, *all* calls started rejecting
>>> >>> with INCOMPATIBLE_DESTINATION.
>>> >>>
>>> >>> Restarting FreeSWITCH fixed the problem. The version of FreeSWITCH
>>> is a
>>> >>> trunk build from approximately 7/11/2013. Is this an issue that an
>>> update
>>> >>> could fix? Could this be port starvation? Please assist, thanks.
>>> >>>
>>> >>> Here's a snippet from the freeswitch log where the first occurrence
>>> of
>>> >>> that error:
>>> >>>
>>> >>> 2013-08-16 09:08:20.048687 [NOTICE] switch_channel.c:1030 New Channel
>>> >>> sofia/external/******9333 at 192.168.1.212:5080
>>> [bfa3419b-0f79-4220-8807-773e3f680751]
>>> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send
>>> signal
>>> >>> sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
>>> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send
>>> signal
>>> >>> sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
>>> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:416
>>> >>> (sofia/external/******9333 at 192.168.1.212:5080) Running State Change
>>> >>> CS_NEW
>>> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:434
>>> >>> (sofia/external/******9333 at 192.168.1.212:5080) State NEW
>>> >>> 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5793 Channel
>>> sofia/external/
>>> >>> ******9333 at 192.168.1.212:5080 entering state [received][100]
>>> >>> 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5802 Remote SDP:
>>> >>> v=0
>>> >>> o=- 1376662098 1 IN IP4 192.168.1.193
>>> >>> s=-
>>> >>> c=IN IP4 192.168.1.193
>>> >>> t=0 0
>>> >>> m=audio 0 RTP/AVP 0 18 101
>>> >>> a=rtpmap:0 PCMU/8000
>>> >>> a=rtpmap:18 G729/8000
>>> >>> a=fmtp:18 annexb=no
>>> >>> a=rtpmap:101 telephone-event/8000
>>> >>> a=fmtp:101 0-15
>>> >>> a=silenceSupp:off - - - -
>>> >>>
>>> >>> 2013-08-16 09:08:20.068687 [NOTICE] sofia.c:6093 Hangup
>>> sofia/external/
>>> >>> ******9333 at 192.168.1.212:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION]
>>> >>>
>>> >>>
>>> >>>
>>> >>> Royce Mitchell, IT Consultant
>>> >>> ITAS Solutions
>>> >>> royce3 at itas-solutions.com
>>> >>>
>>> >>>
>>> _________________________________________________________________________
>>> >>> Professional FreeSWITCH Consulting Services:
>>> >>> consulting at freeswitch.org
>>> >>> http://www.freeswitchsolutions.com
>>> >>>
>>> >>> 
>>> >>> 
>>> >>>
>>> >>> Official FreeSWITCH Sites
>>> >>> http://www.freeswitch.org
>>> >>> http://wiki.freeswitch.org
>>> >>> http://www.cluecon.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
>>> >>>
>>> >>>
>>> >>
>>> >>
>>> _________________________________________________________________________
>>> >> Professional FreeSWITCH Consulting Services:
>>> >> consulting at freeswitch.org
>>> >> http://www.freeswitchsolutions.com
>>> >>
>>> >> 
>>> >> 
>>> >>
>>> >> Official FreeSWITCH Sites
>>> >> http://www.freeswitch.org
>>> >> http://wiki.freeswitch.org
>>> >> http://www.cluecon.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
>>> >>
>>> >>
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.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
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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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2013-08-16 09:24:28.808704 [NOTICE] switch_channel.c:1030 New Channel sofia/external/******5853 at 192.168.1.212:5080 [42ceff14-b8f7-49d3-81b5-baf29ebb3920]
2013-08-16 09:24:28.808704 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/******5853 at 192.168.1.212:5080 [BREAK]
2013-08-16 09:24:28.808704 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/******5853 at 192.168.1.212:5080 [BREAK]
2013-08-16 09:24:28.808704 [DEBUG] switch_core_state_machine.c:416 (sofia/external/******5853 at 192.168.1.212:5080) Running State Change CS_NEW
2013-08-16 09:24:28.808704 [DEBUG] switch_core_state_machine.c:434 (sofia/external/******5853 at 192.168.1.212:5080) State NEW
2013-08-16 09:24:28.828706 [DEBUG] sofia.c:5793 Channel sofia/external/******5853 at 192.168.1.212:5080 entering state [received][100]
2013-08-16 09:24:28.828706 [DEBUG] sofia.c:5802 Remote SDP:
v=0
o=- 1376663067 1 IN IP4 192.168.1.193
s=-
c=IN IP4 192.168.1.193
t=0 0
m=audio 0 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -

2013-08-16 09:24:28.828706 [NOTICE] sofia.c:6093 Hangup sofia/external/******5853 at 192.168.1.212:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION]
2013-08-16 09:24:28.828706 [DEBUG] switch_channel.c:3135 Send signal sofia/external/******5853 at 192.168.1.212:5080 [KILL]
2013-08-16 09:24:28.828706 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/******5853 at 192.168.1.212:5080 [BREAK]
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:416 (sofia/external/******5853 at 192.168.1.212:5080) Running State Change CS_HANGUP
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:679 (sofia/external/******5853 at 192.168.1.212:5080) State HANGUP
2013-08-16 09:24:28.828706 [DEBUG] mod_sofia.c:463 Channel sofia/external/******5853 at 192.168.1.212:5080 hanging up, cause: INCOMPATIBLE_DESTINATION
2013-08-16 09:24:28.828706 [DEBUG] mod_sofia.c:597 Responding to INVITE with: 488
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:48 sofia/external/******5853 at 192.168.1.212:5080 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:679 (sofia/external/******5853 at 192.168.1.212:5080) State HANGUP going to sleep
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:692 (sofia/external/******5853 at 192.168.1.212:5080) Callstate Change DOWN -> HANGUP
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:447 (sofia/external/******5853 at 192.168.1.212:5080) State Change CS_HANGUP -> CS_REPORTING
2013-08-16 09:24:28.828706 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/******5853 at 192.168.1.212:5080 [BREAK]
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:416 (sofia/external/******5853 at 192.168.1.212:5080) Running State Change CS_REPORTING
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:764 (sofia/external/******5853 at 192.168.1.212:5080) State REPORTING
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:92 sofia/external/******5853 at 192.168.1.212:5080 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:764 (sofia/external/******5853 at 192.168.1.212:5080) State REPORTING going to sleep
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:441 (sofia/external/******5853 at 192.168.1.212:5080) State Change CS_REPORTING -> CS_DESTROY
2013-08-16 09:24:28.828706 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/******5853 at 192.168.1.212:5080 [BREAK]
2013-08-16 09:24:28.828706 [DEBUG] switch_core_session.c:1549 Session 134678 (sofia/external/******5853 at 192.168.1.212:5080) Locked, Waiting on external entities
2013-08-16 09:24:28.828706 [NOTICE] switch_core_session.c:1567 Session 134678 (sofia/external/******5853 at 192.168.1.212:5080) Ended
2013-08-16 09:24:28.828706 [NOTICE] switch_core_session.c:1571 Close Channel sofia/external/******5853 at 192.168.1.212:5080 [CS_DESTROY]
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:566 (sofia/external/******5853 at 192.168.1.212:5080) Callstate Change HANGUP -> DOWN
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:569 (sofia/external/******5853 at 192.168.1.212:5080) Running State Change CS_DESTROY
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:579 (sofia/external/******5853 at 192.168.1.212:5080) State DESTROY
2013-08-16 09:24:28.828706 [DEBUG] mod_sofia.c:373 sofia/external/******5853 at 192.168.1.212:5080 SOFIA DESTROY
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:99 sofia/external/******5853 at 192.168.1.212:5080 Standard DESTROY
2013-08-16 09:24:28.828706 [DEBUG] switch_core_state_machine.c:579 (sofia/external/******5853 at 192.168.1.212:5080) State DESTROY going to sleep
2013-08-16 09:24:38.188642 [NOTICE] switch_channel.c:1030 New Channel sofia/external/******5853 at 192.168.1.212:5080 [cacb743c-9b29-4184-bdd8-9037bd41330e]
2013-08-16 09:24:38.188642 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/******5853 at 192.168.1.212:5080 [BREAK]
2013-08-16 09:24:38.188642 [DEBUG] switch_core_session.c:1006 Send signal sofia/external/******5853 at 192.168.1.212:5080 [BREAK]
2013-08-16 09:24:38.209644 [DEBUG] switch_core_state_machine.c:416 (sofia/external/******5853 at 192.168.1.212:5080) Running State Change CS_NEW
2013-08-16 09:24:38.209644 [DEBUG] switch_core_state_machine.c:434 (sofia/external/******5853 at 192.168.1.212:5080) State NEW
2013-08-16 09:24:38.228646 [DEBUG] sofia.c:5793 Channel sofia/external/******5853 at 192.168.1.212:5080 entering state [received][100]
2013-08-16 09:24:38.228646 [DEBUG] sofia.c:5802 Remote SDP:
v=0
o=- 1376663076 1 IN IP4 192.168.1.193
s=-
c=IN IP4 192.168.1.193
t=0 0
m=audio 0 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=silenceSupp:off - - - -

2013-08-16 09:24:38.228646 [NOTICE] sofia.c:6093 Hangup sofia/external/******5853 at 192.168.1.212:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION]
2013-08-16 09:24:38.228646 [DEBUG] switch_channel.c:3135 Send signal sofia/external/******5853 at 192.168.1.212:5080 [KILL]
2013-08-16 09:24:38.228646 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/******5853 at 192.168.1.212:5080 [BREAK]
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:416 (sofia/external/******5853 at 192.168.1.212:5080) Running State Change CS_HANGUP
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:679 (sofia/external/******5853 at 192.168.1.212:5080) State HANGUP
2013-08-16 09:24:38.228646 [DEBUG] mod_sofia.c:463 Channel sofia/external/******5853 at 192.168.1.212:5080 hanging up, cause: INCOMPATIBLE_DESTINATION
2013-08-16 09:24:38.228646 [DEBUG] mod_sofia.c:597 Responding to INVITE with: 488
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:48 sofia/external/******5853 at 192.168.1.212:5080 Standard HANGUP, cause: INCOMPATIBLE_DESTINATION
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:679 (sofia/external/******5853 at 192.168.1.212:5080) State HANGUP going to sleep
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:692 (sofia/external/******5853 at 192.168.1.212:5080) Callstate Change DOWN -> HANGUP
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:447 (sofia/external/******5853 at 192.168.1.212:5080) State Change CS_HANGUP -> CS_REPORTING
2013-08-16 09:24:38.228646 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/******5853 at 192.168.1.212:5080 [BREAK]
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:416 (sofia/external/******5853 at 192.168.1.212:5080) Running State Change CS_REPORTING
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:764 (sofia/external/******5853 at 192.168.1.212:5080) State REPORTING
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:92 sofia/external/******5853 at 192.168.1.212:5080 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:764 (sofia/external/******5853 at 192.168.1.212:5080) State REPORTING going to sleep
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:441 (sofia/external/******5853 at 192.168.1.212:5080) State Change CS_REPORTING -> CS_DESTROY
2013-08-16 09:24:38.228646 [DEBUG] switch_core_session.c:1341 Send signal sofia/external/******5853 at 192.168.1.212:5080 [BREAK]
2013-08-16 09:24:38.228646 [DEBUG] switch_core_session.c:1549 Session 134679 (sofia/external/******5853 at 192.168.1.212:5080) Locked, Waiting on external entities
2013-08-16 09:24:38.228646 [NOTICE] switch_core_session.c:1567 Session 134679 (sofia/external/******5853 at 192.168.1.212:5080) Ended
2013-08-16 09:24:38.228646 [NOTICE] switch_core_session.c:1571 Close Channel sofia/external/******5853 at 192.168.1.212:5080 [CS_DESTROY]
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:566 (sofia/external/******5853 at 192.168.1.212:5080) Callstate Change HANGUP -> DOWN
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:569 (sofia/external/******5853 at 192.168.1.212:5080) Running State Change CS_DESTROY
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:579 (sofia/external/******5853 at 192.168.1.212:5080) State DESTROY
2013-08-16 09:24:38.228646 [DEBUG] mod_sofia.c:373 sofia/external/******5853 at 192.168.1.212:5080 SOFIA DESTROY
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:99 sofia/external/******5853 at 192.168.1.212:5080 Standard DESTROY
2013-08-16 09:24:38.228646 [DEBUG] switch_core_state_machine.c:579 (sofia/external/******5853 at 192.168.1.212:5080) State DESTROY going to sleep

From alex at digitalmail.com  Mon Aug 19 16:35:03 2013
From: alex at digitalmail.com (Alex Lake)
Date: Mon, 19 Aug 2013 13:35:03 +0100
Subject: [Freeswitch-users] Congratulations and Wanted: Single Consultant
Message-ID: <521210F7.5040308@digitalmail.com>

A little bird tells me that Avi Marcus is getting married (or 
"Bridged"?) which is lovely, but that means that there's a little 
project that we're looking for a Freeswitch guru to do for us....

It's essentially that we want a DTMF testing rig setup (on a 
"cloud-based" server of ours) that we can use as a model to determine 
why we're getting poor dtmf recognition from some phones/networks when 
they come to our current setup, but that Avi's test switch picked up 
just fine.

So a consultant who's either young, free and single or one that's old, 
married and bored ;-)

Email me if you're interested and available nowish (we'd like to get 
this done this week if possible)

Alex


From kworm at sofnet.com  Mon Aug 19 17:28:51 2013
From: kworm at sofnet.com (Kevin Wormington)
Date: Mon, 19 Aug 2013 08:28:51 -0500
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <CAMbW9oJ3SG=n0-RdFdtMkksEVOQ-F82KdwvdCiTVh0K=e_7MMA@mail.gmail.com>
References: <CE33869E.78FAE%krice@freeswitch.org>
	<520E25CB.7030404@puzzled.xs4all.nl>
	<00000140874a3df1-e0a86002-7ad9-485d-b86b-200a83300469-000000@email.amazonses.com>
	<521017FE.7070106@xtronics.com>
	<CAMbW9oJ3SG=n0-RdFdtMkksEVOQ-F82KdwvdCiTVh0K=e_7MMA@mail.gmail.com>
Message-ID: <52121D93.5050809@sofnet.com>

I have some Yealink T28Ps and a T22P in the lab and they seem to work 
fine.  Call quality is fine.  I have been working my way through all the 
FS features on them and so far BLF is working other than the light going 
dim when a monitored extension is not registered.  I think this is 
probably in FreeSWITCH though and I haven't had time to get to the 
bottom of it.

Kevin

On 08/19/2013 03:25 AM, Andrew Cassidy wrote:
> Good stuff, that's pretty useful to me too. Like I say, the newer Cisco
> SPA50x's are nice, the old 79xx series are horrible for SIP (as someone
> else mentioned)
>
> Yealink have been mentioned, has anyone got any thoughts on those? I've
> been thinking about investigating them.
>
>
> On 18 August 2013 01:40, Karl Schmidt <karl at xtronics.com
> <mailto:karl at xtronics.com>> wrote:
>
>     On 08/16/2013 08:22 AM, Avi Marcus wrote:
>      > Just btw -- this is hardly the first time this question has been
>     asked, and nothing has been
>      > summarized on the wiki for future reference (that I know of).
>      > So, someone that has the time....
>      >
>      > -Avi
>
>     I'm actually summarizing the response here (as it seemed a bit off
>     topic re freeswitch) :
>
>     http://wiki.xtronics.com/index.php/SIP_Phones
>
>     Work in processes - feel free grab and post on wiki.freeswitch.org
>     <http://wiki.freeswitch.org>
>
>
>
>
>
>     --------------------------------------------------------------------------------
>     Karl Schmidt                                  EMail
>     Karl at xtronics.com <mailto:Karl at xtronics.com>
>     Transtronics, Inc.                              WEB
>     http://secure.transtronics.com
>     3209 West 9th Street                             Ph (785) 841-3089
>     Lawrence, KS 66049                              FAX (785) 841-0434
>
>     The society that puts equality before freedom will
>     end up with neither. The society that puts freedom
>     before equality will end up with a great measure of both.
>     - Milton Freidman
>     --------------------------------------------------------------------------------
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <mailto:info at cassidywebservices.co.uk> *03300 100 960 *F
> <mailto:info at cassidywebservices.co.uk> *03300 100 961
> *E <mailto:info at cassidywebservices.co.uk>
> *andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk>
> *W <mailto:info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
> <http://www.cassidywebservices.co.uk>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mike at jerris.com  Mon Aug 19 17:38:52 2013
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Aug 2013 09:38:52 -0400
Subject: [Freeswitch-users] Should WebRTC work in a double-NAT
	environment?
In-Reply-To: <1376692461.52975.YahooMailNeo@web120704.mail.ne1.yahoo.com>
References: <CAK59s4y6YHecUr-Te-TMvJoS=ASxthL8FCuYmNYhEmkfQioFmQ@mail.gmail.com>	<CAKbxfG96LTXzrB92TGNEkFvmTU9bZz1WX=VFK9p+a_Hra9ZDBg@mail.gmail.com>	<CAK59s4yjYoEDVJJyEx2CE0z=tqn_aSNvg76TXBVrFUPR3as=gQ@mail.gmail.com>	<CAK59s4yH0-B8nxK30k7TBgDgivDwJzPfDTeE0kOXiuKKdrFvXQ@mail.gmail.com>
	<CAKbxfG-qYZcVGr80K8x3jO4V0mfQfjcDKcTFMeWDt1MOMKUDFw@mail.gmail.com>
	<1376692461.52975.YahooMailNeo@web120704.mail.ne1.yahoo.com>
Message-ID: <62BE6DC6-F078-4A71-BAB2-8D468F705997@jerris.com>

As long as its natted to a static address on the server side and you have ext-rtp-ip set, it should work.  It may not work with some nasty nats on the server side where it changes which port number too.

On Aug 16, 2013, at 6:34 PM, D D <tru083 at yahoo.com> wrote:

> Hi,
> 
> We have a Freeswitch running in a double-NAT environment, where the 
> server is in a NAT network, and the client is in a remote NAT network.
> 
> In the remote network, using a SIP softphone, we can make calls into the server 
> and hear the media.
> 
> Using a web browser on the same network as the switch, we can hear
> media in WebRTC (using JSSIP).
> 
> But when using a web browser in the remote network, we can see the signaling
> but cannot hear the media.
> 
> Should WebRTC work in a double-NAT environment?  Any ideas why the
> media is not working in this environment?
> 
> Thanks,
> David

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From mike at jerris.com  Mon Aug 19 17:40:07 2013
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Aug 2013 09:40:07 -0400
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <2ECFB1E034254C48ACBDD31C7E189BA9@gmail.com>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>
	<CE311BFC.78EA0%krice@freeswitch.org>
	<045801ce990e$205b4f90$6111eeb0$@207me.com>
	<CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>
	<049101ce9915$e1cbb5f0$a56321d0$@207me.com>
	<CAOP4xcy6NZG0JKPMYhJp_wxaLqnAjsEkGMgTD5itvBPNdSCPzQ@mail.gmail.com>
	<050301ce993b$4dc37b60$e94a7220$@207me.com>
	<CAKbxfG80vr3b+BULjn5ZhKW6ZjxY6Khy4Ymhk8RWGVFLpuSKoA@mail.gmail.com>
	<0E0D5ED7-E97F-4F17-8290-2D893649A8F8@jerris.com>
	<CAOP4xcwBNPxaxsrFBr48-02QriO_baDv-=o+QhoDsiMKUnq4ug@mail.gmail.com>
	<CAKbxfG8zu0+_AM4zzmAq01BFHxpLygb8-2JdXNujOw6X6UMesw@mail.gmail.com>
	<CADk4Sr1g2c3toiU7+ikM_WByfUzvxwXKQwLrFXF8KnrtZoXWnQ@mail.gmail.com>
	<CAOP4xcy9p1PvL6ggmWK=betiGb=N=2DENvMA=ieiSNo0mZF9Rw@mail.gmail.com>
	<2ECFB1E034254C48ACBDD31C7E189BA9@gmail.com>
Message-ID: <3B81AEAC-802B-4B56-A7E3-739B78895D42@jerris.com>

But this would still include buffering? although it should be less than 12 seconds.

On Aug 16, 2013, at 8:04 PM, Seven Du <dujinfang at gmail.com> wrote:

> Another solution would to use flash with mod_rtmp.  And to make that work better you need to change the flash code to remove the sound input device since you don't need it and this also helps to remove the confirmation prompt anytime you use a sound input in flash.
> 
> I think I had written a plugin for flowplayer long time ago which works well.
> 
> 
> -- 
> Seven Du
> http://www.freeswitch.org.cn
> http://about.me/dujinfang
> http://www.dujinfang.com
> 
> Sent with Sparrow
> 
> On Saturday, August 17, 2013 at 2:22 AM, Jorge N??ez wrote:
> 
>> So Can anybody tell any other solution instead of webrtc and icecast to broadcast a conference in realtime without delay or maybe just a little bit of delay 1s or 2s?
>> 
>> Thank you very much for your help
>> 
>> Jorge
>> 
>> 
>> 2013/8/15 Jeff Leung <jleung at v10networks.ca>
>>> Icecast is known to have a relatively high latency for audio.
>>> 
>>> 
>>> On Thu, Aug 15, 2013 at 10:34 AM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>>> enable_file_write_buffering=false
>>>> stream_prebuffer=0
>>>> 
>>>> On the leg doing the recording.  It does't stop the iceast server or libshout from buffering it more.
>>>> 
>>>> 
>>>> 
>>>> 
>>>> 
>>>> On Thu, Aug 15, 2013 at 12:07 PM, Jorge N??ez <jorgefren12 at gmail.com> wrote:
>>>>> Anthony and just for test how can I reduce the latency? and thanks Michael for use webrtc Do I need to use Sipml5 or JSSIP o are there any othe way to use because I was trying to find some information about mod_html5 but I didn?t find
>>>>> 
>>>>> thanks
>>>>> 
>>>>> Jorge
>>>>> 
>>>>> 
>>>>> 2013/8/15 Michael Jerris <mike at jerris.com>
>>>>>> The alternative if you want in sync audio would be to use webrtc endpoints to listen.
>>>>>> 
>>>>>> On Aug 14, 2013, at 7:19 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:
>>>>>> 
>>>>>>> Icecast is not designed for low-latency and realtime audio.  Its designed for higher quality reliable audio.  The servers and the client libs both use latency and buffering to guarantee this.  There are some techniques to reduce the buffering on at least the FS side but then you will start getting skips and resets if you miss any packets.
>>>>>>> 
>>>>>>> This comes up all the time when people first try doing this.  Either you need to just accept the delay since most people will not even know its there or use some other method.
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> On Wed, Aug 14, 2013 at 5:12 PM, Stephen Dame <sdame at 207me.com> wrote:
>>>>>>>> Did you restart the freeswitch server after changes to the config files,   I?m not sure 100% if reloadxml resets the conference settings.
>>>>>>>>  
>>>>>>>> Also I use some flash based players, I think there where client settiings for buffering.   Can you set buffer in HTML tag?
>>>>>>>>  
>>>>>>>> Regards,
>>>>>>>> Stephen
>>>>>>>>  
>>>>>>>> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame
>>>>>>>>  
>>>>>>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jorge N??ez
>>>>>>>> Sent: Wednesday, August 14, 2013 4:18 PM
>>>>>>>> 
>>>>>>>> To: FreeSWITCH Users Help
>>>>>>>> Subject: Re: [Freeswitch-users] Mod_shout using with icecast Big delay
>>>>>>>> 
>>>>>>>>  
>>>>>>>> Hi thanks, I set your configuration but nothing changed, I reduced the burst size and it takes me just 11s and also I tried with 8k, 16k and 32k and nothing change
>>>>>>>>  
>>>>>>>>     <limits>
>>>>>>>>         <clients>100</clients>
>>>>>>>>         <sources>10</sources>
>>>>>>>>         <threadpool>5</threadpool>
>>>>>>>>         <queue-size>524288</queue-size>
>>>>>>>>         <!-- <queue-size>524288</queue-size> -->
>>>>>>>>         <client-timeout>30</client-timeout>
>>>>>>>>         <header-timeout>15</header-timeout>
>>>>>>>>         <source-timeout>10</source-timeout>
>>>>>>>>         <!-- If enabled, this will provide a burst of data when a client 
>>>>>>>>              first connects, thereby significantly reducing the startup 
>>>>>>>>              time for listeners that do substantial buffering. However,
>>>>>>>>              it also significantly increases latency between the source
>>>>>>>>              client and listening client.  For low-latency setups, you
>>>>>>>>              might want to disable this. -->
>>>>>>>>         <burst-on-connect>0</burst-on-connect>
>>>>>>>>         <!-- same as burst-on-connect, but this allows for being more
>>>>>>>>              specific on how much to burst. Most people won't need to
>>>>>>>>              change from the default 64k. Applies to all mountpoints 
>>>>>>>>  
>>>>>>>>  -->
>>>>>>>>         <burst-size>4096</burst-size>
>>>>>>>>     </limits>
>>>>>>>>  
>>>>>>>> Regards
>>>>>>>>  
>>>>>>>> Jorge
>>>>>>>>  
>>>>>>>>  
>>>>>>>> 2013/8/14 Stephen Dame <sdame at 207me.com>
>>>>>>>> Jorge,
>>>>>>>>  
>>>>>>>> Play around with the burst size and queue size?  here is my xml config around 3-4 second delay from 16k freeswitch conference.
>>>>>>>> To small a buffer and the players disconnect? Im also running icecast on same server.
>>>>>>>>  
>>>>>>>> <icecast>
>>>>>>>>     <limits>
>>>>>>>>         <clients>100</clients>
>>>>>>>>         <sources>10</sources>
>>>>>>>>         <threadpool>5</threadpool>
>>>>>>>>         <queue-size>524288</queue-size>
>>>>>>>>         <client-timeout>30</client-timeout>
>>>>>>>>         <header-timeout>15</header-timeout>
>>>>>>>>         <source-timeout>10</source-timeout>
>>>>>>>>         <burst-on-connect>1</burst-on-connect>
>>>>>>>>         <!-- same as burst-on-connect, but this allows for being more
>>>>>>>>              specific on how much to burst. Most people won't need to
>>>>>>>>              change from the default 64k. Applies to all mountpoints  -->
>>>>>>>>         <burst-size>65535</burst-size>
>>>>>>>>     </limits>
>>>>>>>>  
>>>>>>>> Regards,
>>>>>>>> Stephen
>>>>>>>>  
>>>>>>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
>>>>>>>> Sent: Wednesday, August 14, 2013 12:35 PM
>>>>>>>> To: FreeSWITCH Users Help
>>>>>>>> Subject: Re: [Freeswitch-users] Mod_shout using with icecast Big delay
>>>>>>>>  
>>>>>>>> You have to keep in mind that icecast itself has a fair bit of buffering involved with it and theres not a lot you can do about that
>>>>>>>> 
>>>>>>>> 
>>>>>>>> On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com> wrote:
>>>>>>>> Hi I am using mod_shout to send a conference call to icecast and hear it from a tag audio in html5 in realtime, but it has a big delay like 12 seconds. How can I reduce the latency of the audio sent from freeswitch or what can I do to improve this.
>>>>>>>> 
>>>>>>>> Regards
>>>>>>>> 
>>>>>>>> Jorge
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>> 
>>>>>>>> -- 
>>>>>>>> Ken
>>>>>>>> http://www.FreeSWITCH.org
>>>>>>>> http://www.ClueCon.com
>>>>>>>> http://www.OSTAG.org
>>>>>>>> irc.freenode.net #freeswitch
>>>>>>>> 
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>  
>>>>>>>> 
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> 
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> 
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>> 
>>>>>> 
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> 
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>> 
>>>>> 
>>>>> 
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mike at jerris.com  Mon Aug 19 17:41:42 2013
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Aug 2013 09:41:42 -0400
Subject: [Freeswitch-users] does fs have problems with a . in the
	password field?
In-Reply-To: <23874.1376695780@ccs.covici.com>
References: <23874.1376695780@ccs.covici.com>
Message-ID: <0D69EC7C-F53C-44C1-BBA1-6CDA4CD768A8@jerris.com>

I guess it depends if we were screwing it up or they were.  This would be a strange problem for us to have never seen before.  Can anyone else confirm if this is our problem or not?

On Aug 16, 2013, at 7:29 PM, covici at ccs.covici.com wrote:

> Hi.  I was having problems with one of my carriers -- they changed the
> password for my account without notice!  The password they changed to
> had a . and they kept saying I had the wrong password or anyway an
> authentication error, till at their suggestion, I removed the  on both
> sides, and bingo I was able to register.  I am using
> FreeSWITCH Version 1.5.3b+git~20130709T135336Z~267ef728e1 (git 267ef72
> 2013-07-09 13:53:36Z) .
> 
> Should a Jira be filed?
> 



From freeswitch-list at puzzled.xs4all.nl  Mon Aug 19 17:55:57 2013
From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists)
Date: Mon, 19 Aug 2013 15:55:57 +0200
Subject: [Freeswitch-users] Congratulations and Wanted: Single Consultant
In-Reply-To: <521210F7.5040308@digitalmail.com>
References: <521210F7.5040308@digitalmail.com>
Message-ID: <521223ED.4030408@puzzled.xs4all.nl>

On 08/19/2013 02:35 PM, Alex Lake wrote:
> A little bird tells me that Avi Marcus is getting married (or

Congratulations Avi!

Regards,
Patrick




From fdelawarde at wirelessmundi.com  Mon Aug 19 17:59:39 2013
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?=)
Date: Mon, 19 Aug 2013 15:59:39 +0200
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <52121D93.5050809@sofnet.com>
References: <CE33869E.78FAE%krice@freeswitch.org>
	<520E25CB.7030404@puzzled.xs4all.nl>
	<00000140874a3df1-e0a86002-7ad9-485d-b86b-200a83300469-000000@email.amazonses.com>
	<521017FE.7070106@xtronics.com>
	<CAMbW9oJ3SG=n0-RdFdtMkksEVOQ-F82KdwvdCiTVh0K=e_7MMA@mail.gmail.com>
	<52121D93.5050809@sofnet.com>
Message-ID: <1376920779.2668.7.camel@luna.madrid.commsmundi.com>

Have been implementing Yealink for a couple of years: nice design, low
price, fine quality, enterprise feature set, easy to setup, responsive
support. For me the best quality/price for 2013, and by far...

BTW, BLF works fine, at least with git HEAD.

Fran?ois.

On Mon, 2013-08-19 at 08:28 -0500, Kevin Wormington wrote: 
> I have some Yealink T28Ps and a T22P in the lab and they seem to work 
> fine.  Call quality is fine.  I have been working my way through all the 
> FS features on them and so far BLF is working other than the light going 
> dim when a monitored extension is not registered.  I think this is 
> probably in FreeSWITCH though and I haven't had time to get to the 
> bottom of it.
> 
> Kevin
> 
> On 08/19/2013 03:25 AM, Andrew Cassidy wrote:
> > Good stuff, that's pretty useful to me too. Like I say, the newer Cisco
> > SPA50x's are nice, the old 79xx series are horrible for SIP (as someone
> > else mentioned)
> >
> > Yealink have been mentioned, has anyone got any thoughts on those? I've
> > been thinking about investigating them.
> >
> >
> > On 18 August 2013 01:40, Karl Schmidt <karl at xtronics.com
> > <mailto:karl at xtronics.com>> wrote:
> >
> >     On 08/16/2013 08:22 AM, Avi Marcus wrote:
> >      > Just btw -- this is hardly the first time this question has been
> >     asked, and nothing has been
> >      > summarized on the wiki for future reference (that I know of).
> >      > So, someone that has the time....
> >      >
> >      > -Avi
> >
> >     I'm actually summarizing the response here (as it seemed a bit off
> >     topic re freeswitch) :
> >
> >     http://wiki.xtronics.com/index.php/SIP_Phones
> >
> >     Work in processes - feel free grab and post on wiki.freeswitch.org
> >     <http://wiki.freeswitch.org>
> >
> >
> >
> >
> >
> >     --------------------------------------------------------------------------------
> >     Karl Schmidt                                  EMail
> >     Karl at xtronics.com <mailto:Karl at xtronics.com>
> >     Transtronics, Inc.                              WEB
> >     http://secure.transtronics.com
> >     3209 West 9th Street                             Ph (785) 841-3089
> >     Lawrence, KS 66049                              FAX (785) 841-0434
> >
> >     The society that puts equality before freedom will
> >     end up with neither. The society that puts freedom
> >     before equality will end up with a great measure of both.
> >     - Milton Freidman
> >     --------------------------------------------------------------------------------
> >
> >     _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com
> >
> >     
> >     
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org
> >     http://wiki.freeswitch.org
> >     http://www.cluecon.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
> >
> >
> >
> >
> > --
> > *Andrew Cassidy BSc (Hons) MBCS SSCA*
> > Managing Director
> >
> >
> > *T <mailto:info at cassidywebservices.co.uk> *03300 100 960 *F
> > <mailto:info at cassidywebservices.co.uk> *03300 100 961
> > *E <mailto:info at cassidywebservices.co.uk>
> > *andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk>
> > *W <mailto:info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
> > <http://www.cassidywebservices.co.uk>
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 covici at ccs.covici.com  Mon Aug 19 18:01:53 2013
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 19 Aug 2013 10:01:53 -0400
Subject: [Freeswitch-users] does fs have problems with a . in the
	password field?
In-Reply-To: <0D69EC7C-F53C-44C1-BBA1-6CDA4CD768A8@jerris.com>
References: <23874.1376695780@ccs.covici.com>
	<0D69EC7C-F53C-44C1-BBA1-6CDA4CD768A8@jerris.com>
Message-ID: <29300.1376920913@ccs.covici.com>

It is callwithus where I was having the problem -- they changed my
password without notice!!  It maybe them, I cannot tell from here.

Michael Jerris <mike at jerris.com> wrote:

> I guess it depends if we were screwing it up or they were.  This would be a strange problem for us to have never seen before.  Can anyone else confirm if this is our problem or not?
> 
> On Aug 16, 2013, at 7:29 PM, covici at ccs.covici.com wrote:
> 
> > Hi.  I was having problems with one of my carriers -- they changed the
> > password for my account without notice!  The password they changed to
> > had a . and they kept saying I had the wrong password or anyway an
> > authentication error, till at their suggestion, I removed the  on both
> > sides, and bingo I was able to register.  I am using
> > FreeSWITCH Version 1.5.3b+git~20130709T135336Z~267ef728e1 (git 267ef72
> > 2013-07-09 13:53:36Z) .
> > 
> > Should a Jira be filed?
> > 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
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 fs.user at fordior.net  Mon Aug 19 18:23:37 2013
From: fs.user at fordior.net (EL)
Date: Mon, 19 Aug 2013 16:23:37 +0200
Subject: [Freeswitch-users] Auto-detecting Local IP when no gateway set
In-Reply-To: <CAHgk4BVujRwVT55BfLLo8kcLU76Jr09hdQua01dvYYuL7tuVeg@mail.gmail.com>
References: <CAHccO0VMg6TUU-5SdL2voPeTAbFpJJTCo3cQLHu9ktuiJFTmwQ@mail.gmail.com>
	<CADk4Sr3G8AqAOHuXtVjYww-_9ZT4JWaVER-DzGAY4AAQjCF_-w@mail.gmail.com>
	<CAHccO0UJyRtVxyS4NFhaW+L=Oztbd2mgSNRCUNOzKvn5pjmnkg@mail.gmail.com>
	<CAFiqYunKSVHQPb_zHijLu-xWaCqeG=D6gOa1C5Adhb3P3uYvag@mail.gmail.com>
	<CAB+Kzh6g+uSu64AJ6k=p3S2Xux9hN8sVduR+zgppy_smGON88w@mail.gmail.com>
	<CAFiqYu=0FpR2q20BCQar1gxnwgEUCK1uZNKyLZKymRc2ukrbdg@mail.gmail.com>
	<CAFiqYumr7SYWP5WJWWfxsYDtKrj7dBBt8VcSEf2OaxUW5EEKKg@mail.gmail.com>
	<CAKbxfG-QQWxy9q6p3yvxPoVdGpeATgnza49YfCvmu=vPhtg-nw@mail.gmail.com>
	<CAHccO0X_Dr+fP8eHmkeTVwergLDaP+kjAbQ0n_p24x-NdXY1tQ@mail.gmail.com>
	<CAHgk4BVujRwVT55BfLLo8kcLU76Jr09hdQua01dvYYuL7tuVeg@mail.gmail.com>
Message-ID: <20130819142337.GE24164@0rdior.com>


> /sbin/ifconfig $1 | grep "inet addr" | awk -F: '{print $2}' | awk '{print
> $1}'

To make it a bit less ancient:

========
/bin/ip -o addr show | grep inet | awk '{print $4}' | awk -F/ '{print $1}'

or just for eth0:

/bin/ip -o addr show eth0 | grep inet | awk '{print $4}' | awk -F/ '{print $1}'
========
 
This might be an interesting article to read, regarding ifconfig:

http://inai.de/2008/02/19

Further interesting peace of information regarding linux/iptables 
(section 'Mythbusting time') see: http://inai.de/links/iptables/

I hope this information may be usefull to you.

-- 
EL


From sos at sokhapkin.dyndns.org  Mon Aug 19 18:25:32 2013
From: sos at sokhapkin.dyndns.org (Sergey Okhapkin)
Date: Mon, 19 Aug 2013 10:25:32 -0400
Subject: [Freeswitch-users] does fs have problems with a . in the
	password field?
In-Reply-To: <29300.1376920913@ccs.covici.com>
References: <23874.1376695780@ccs.covici.com>
	<0D69EC7C-F53C-44C1-BBA1-6CDA4CD768A8@jerris.com>
	<29300.1376920913@ccs.covici.com>
Message-ID: <3029537.fOd83fM3bv@sos>

Password with a dot character works fine with callwithus in linphone.

On Monday 19 August 2013 10:01:53 covici at ccs.covici.com wrote:
> It is callwithus where I was having the problem -- they changed my
> password without notice!!  It maybe them, I cannot tell from here.
> 
> Michael Jerris <mike at jerris.com> wrote:
> > I guess it depends if we were screwing it up or they were.  This would be
> > a strange problem for us to have never seen before.  Can anyone else
> > confirm if this is our problem or not?> 
> > On Aug 16, 2013, at 7:29 PM, covici at ccs.covici.com wrote:
> > > Hi.  I was having problems with one of my carriers -- they changed the
> > > password for my account without notice!  The password they changed to
> > > had a . and they kept saying I had the wrong password or anyway an
> > > authentication error, till at their suggestion, I removed the  on both
> > > sides, and bingo I was able to register.  I am using
> > > FreeSWITCH Version 1.5.3b+git~20130709T135336Z~267ef728e1 (git 267ef72
> > > 2013-07-09 13:53:36Z) .
> > > 
> > > Should a Jira be filed?
> > 
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> > 
> > 
> > 
> > 
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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 kworm at sofnet.com  Mon Aug 19 18:34:16 2013
From: kworm at sofnet.com (Kevin Wormington)
Date: Mon, 19 Aug 2013 09:34:16 -0500
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <1376920779.2668.7.camel@luna.madrid.commsmundi.com>
References: <CE33869E.78FAE%krice@freeswitch.org>
	<520E25CB.7030404@puzzled.xs4all.nl>
	<00000140874a3df1-e0a86002-7ad9-485d-b86b-200a83300469-000000@email.amazonses.com>
	<521017FE.7070106@xtronics.com>
	<CAMbW9oJ3SG=n0-RdFdtMkksEVOQ-F82KdwvdCiTVh0K=e_7MMA@mail.gmail.com>
	<52121D93.5050809@sofnet.com>
	<1376920779.2668.7.camel@luna.madrid.commsmundi.com>
Message-ID: <52122CE8.7080000@sofnet.com>

Fran?ois,

I'm running 1.2.12 from git 8/16/13 and when a monitored extension is 
not registered the light remains solid green.  On your system is the 
light going out for extensions that become un-registered or are not 
registered when the phone powers on?

On 08/19/2013 08:59 AM, Fran?ois wrote:
> Have been implementing Yealink for a couple of years: nice design, low
> price, fine quality, enterprise feature set, easy to setup, responsive
> support. For me the best quality/price for 2013, and by far...
>
> BTW, BLF works fine, at least with git HEAD.
>
> Fran?ois.
>
> On Mon, 2013-08-19 at 08:28 -0500, Kevin Wormington wrote:
>> I have some Yealink T28Ps and a T22P in the lab and they seem to work
>> fine.  Call quality is fine.  I have been working my way through all the
>> FS features on them and so far BLF is working other than the light going
>> dim when a monitored extension is not registered.  I think this is
>> probably in FreeSWITCH though and I haven't had time to get to the
>> bottom of it.
>>
>> Kevin
>>
>> On 08/19/2013 03:25 AM, Andrew Cassidy wrote:
>>> Good stuff, that's pretty useful to me too. Like I say, the newer Cisco
>>> SPA50x's are nice, the old 79xx series are horrible for SIP (as someone
>>> else mentioned)
>>>

>>> Yealink have been mentioned, has anyone got any thoughts on those? I've
>>> been thinking about investigating them.
>>>
>>>
>>> On 18 August 2013 01:40, Karl Schmidt <karl at xtronics.com
>>> <mailto:karl at xtronics.com>> wrote:
>>>
>>>      On 08/16/2013 08:22 AM, Avi Marcus wrote:
>>>       > Just btw -- this is hardly the first time this question has been
>>>      asked, and nothing has been
>>>       > summarized on the wiki for future reference (that I know of).
>>>       > So, someone that has the time....
>>>       >
>>>       > -Avi
>>>
>>>      I'm actually summarizing the response here (as it seemed a bit off
>>>      topic re freeswitch) :
>>>
>>>      http://wiki.xtronics.com/index.php/SIP_Phones
>>>
>>>      Work in processes - feel free grab and post on wiki.freeswitch.org
>>>      <http://wiki.freeswitch.org>
>>>
>>>
>>>
>>>
>>>
>>>      --------------------------------------------------------------------------------
>>>      Karl Schmidt                                  EMail
>>>      Karl at xtronics.com <mailto:Karl at xtronics.com>
>>>      Transtronics, Inc.                              WEB
>>>      http://secure.transtronics.com
>>>      3209 West 9th Street                             Ph (785) 841-3089
>>>      Lawrence, KS 66049                              FAX (785) 841-0434
>>>
>>>      The society that puts equality before freedom will
>>>      end up with neither. The society that puts freedom
>>>      before equality will end up with a great measure of both.
>>>      - Milton Freidman
>>>      --------------------------------------------------------------------------------
>>>
>>>      _________________________________________________________________________
>>>      Professional FreeSWITCH Consulting Services:
>>>      consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>>      http://www.freeswitchsolutions.com
>>>
>>>      
>>>      
>>>
>>>      Official FreeSWITCH Sites
>>>      http://www.freeswitch.org
>>>      http://wiki.freeswitch.org
>>>      http://www.cluecon.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
>>>
>>>
>>>
>>>
>>> --
>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>> Managing Director
>>>
>>>
>>> *T <mailto:info at cassidywebservices.co.uk> *03300 100 960 *F
>>> <mailto:info at cassidywebservices.co.uk> *03300 100 961
>>> *E <mailto:info at cassidywebservices.co.uk>
>>> *andrew at cassidywebservices.co.uk <mailto:andrew at cassidywebservices.co.uk>
>>> *W <mailto:info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>> <http://www.cassidywebservices.co.uk>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 vallimamod.abdullah at imtelecom.fr  Mon Aug 19 19:06:02 2013
From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH)
Date: Mon, 19 Aug 2013 17:06:02 +0200
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <CAHwzD9=gpZjsuVxdj3O04PMPCh_mqoxj8SudvXak=ejWZJiohw@mail.gmail.com>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
	<CAFiqYumBFafVazhjXUf1CdXgVaYqvbU4QzrPEc2svCLJu0BaFw@mail.gmail.com>
	<CAHwzD9n+CfCbCYNvwPcSiwYQoQe9bved3uK6q1=okh+2ravZoA@mail.gmail.com>
	<A46F2C5A-8082-4155-A4E0-96CE65AF3431@insensate.co.uk>
	<CAHwzD9ncYxxJxaMYnu2qx78Q_SSLp3onz+5=kN=DSz+vRVQGKA@mail.gmail.com>
	<CAFiqYunDhXi+v1H4P=vQnrGW2WRd61w1=nnGwVPo+Pv9CWG3Dw@mail.gmail.com>
	<CAHwzD9=gpZjsuVxdj3O04PMPCh_mqoxj8SudvXak=ejWZJiohw@mail.gmail.com>
Message-ID: <94920D4E-5645-4BCE-A6A9-C7780E0BDBC6@imtelecom.fr>

Hi,

It looks like the remote SDP does not contain correct rtp port information. You have:

m=audio 0 RTP/AVP 0 18 101

The first '0' should be the rtp port normally. I am not sure but it does not looks correct and may be the cause of the incompatible destination error.
Hope this helps.

-- 
Best Regards,
Vallimamod
.



On Aug 19, 2013, at 2:59 PM, Royce Mitchell III <royce3 at gmail.com> wrote:

> Here is one example, thanks
> 
> 
> 
> Royce Mitchell, IT Consultant
> ITAS Solutions
> royce3 at itas-solutions.com
> 
> 
> On Fri, Aug 16, 2013 at 4:12 PM, Steven Ayre <steveayre at gmail.com> wrote:
> Can you paste a debug log of the entire lifetime of the call?
> 
> 
> On 16 August 2013 21:57, Royce Mitchell III <royce3 at gmail.com> wrote:
> The only thing I'm doing in FS regarding codecs is I force PCMU in certain conditions ( ran into a transcoding bug in FreeSwitch between G722 HD and G711 with the Polycom phones )
> 
> 
> 
> Royce Mitchell, IT Consultant
> ITAS Solutions
> royce3 at itas-solutions.com
> 
> 
> On Fri, Aug 16, 2013 at 3:35 PM, Lawrence Conroy <lconroy at insensate.co.uk> wrote:
> Hi there,
>  Are you sure about fS allowing PCMU?
> According to the remote SDP, your incall is proposing PCMU as its top choice; g729 is 2nd best (quite right too :).
> Forcing the adtran to offer only PCMU should not make ay difference to that -- it'll still propose PCMU so no change.
> Either your setup is somehow blocking PCMU on the b-leg (but I'd expect to see that on the log), OR is trying to transcode because the b-leg requires some (non-PCMU) codec and can't (again, I'd expect that to be logged), OR fS is not accepting PCMU.
> Assuming that PCMU is in the fS vars codec lists, does your dialplan do anything funky with the codec list for an incall?
> all the best,
>   Lawrence
> 
> On 16 Aug 2013, at 21:06, Royce Mitchell III wrote:
> > My FreeSWITCH is configured to prefer PCMU, and the devices it is talking
> > to are Adtran 908e's. The Adtrans are configured for the default codec
> > group which is supposed to be PCMU, but I can reconfigure them to
> > explicitly allow only PCMU. I will try that and see if it makes a
> > difference.
> >
> >
> >
> > Royce Mitchell, IT Consultant
> > ITAS Solutions
> > royce3 at itas-solutions.com
> >
> >
> > On Fri, Aug 16, 2013 at 11:04 AM, Steven Ayre <steveayre at gmail.com> wrote:
> >
> >> INCOMPATIBLE_DESTINATION means a codec problem.
> >>
> >> The remote SDP sends they're offerring PCMU and G729.
> >>
> >> What codecs are you allowing, what codecs are you bridging with, and since
> >> G729 is on the list are you perhaps trying to transcode without using
> >> mod_com_g729+licenses?
> >>
> >>
> >> On 16 August 2013 16:07, Royce Mitchell III <royce3 at gmail.com> wrote:
> >>
> >>> Hi,
> >>>
> >>> My client has been running FreeSWITCH with low call volume and has been
> >>> stable for several months now. This Tuesday morning we switched all call
> >>> volume over to FreeSWITCH and everything was running fine until this
> >>> morning.
> >>>
> >>> Out of the blue, *all* calls started rejecting
> >>> with INCOMPATIBLE_DESTINATION.
> >>>
> >>> Restarting FreeSWITCH fixed the problem. The version of FreeSWITCH is a
> >>> trunk build from approximately 7/11/2013. Is this an issue that an update
> >>> could fix? Could this be port starvation? Please assist, thanks.
> >>>
> >>> Here's a snippet from the freeswitch log where the first occurrence of
> >>> that error:
> >>>
> >>> 2013-08-16 09:08:20.048687 [NOTICE] switch_channel.c:1030 New Channel
> >>> sofia/external/******9333 at 192.168.1.212:5080[bfa3419b-0f79-4220-8807-773e3f680751]
> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send signal
> >>> sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_session.c:1006 Send signal
> >>> sofia/external/******9333 at 192.168.1.212:5080 [BREAK]
> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:416
> >>> (sofia/external/******9333 at 192.168.1.212:5080) Running State Change
> >>> CS_NEW
> >>> 2013-08-16 09:08:20.048687 [DEBUG] switch_core_state_machine.c:434
> >>> (sofia/external/******9333 at 192.168.1.212:5080) State NEW
> >>> 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5793 Channel sofia/external/
> >>> ******9333 at 192.168.1.212:5080 entering state [received][100]
> >>> 2013-08-16 09:08:20.068687 [DEBUG] sofia.c:5802 Remote SDP:
> >>> v=0
> >>> o=- 1376662098 1 IN IP4 192.168.1.193
> >>> s=-
> >>> c=IN IP4 192.168.1.193
> >>> t=0 0
> >>> m=audio 0 RTP/AVP 0 18 101
> >>> a=rtpmap:0 PCMU/8000
> >>> a=rtpmap:18 G729/8000
> >>> a=fmtp:18 annexb=no
> >>> a=rtpmap:101 telephone-event/8000
> >>> a=fmtp:101 0-15
> >>> a=silenceSupp:off - - - -
> >>>
> >>> 2013-08-16 09:08:20.068687 [NOTICE] sofia.c:6093 Hangup sofia/external/
> >>> ******9333 at 192.168.1.212:5080 [CS_NEW] [INCOMPATIBLE_DESTINATION]
> >>>

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From anthony.minessale at gmail.com  Mon Aug 19 19:22:22 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 19 Aug 2013 10:22:22 -0500
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <3B81AEAC-802B-4B56-A7E3-739B78895D42@jerris.com>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>
	<CE311BFC.78EA0%krice@freeswitch.org>
	<045801ce990e$205b4f90$6111eeb0$@207me.com>
	<CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>
	<049101ce9915$e1cbb5f0$a56321d0$@207me.com>
	<CAOP4xcy6NZG0JKPMYhJp_wxaLqnAjsEkGMgTD5itvBPNdSCPzQ@mail.gmail.com>
	<050301ce993b$4dc37b60$e94a7220$@207me.com>
	<CAKbxfG80vr3b+BULjn5ZhKW6ZjxY6Khy4Ymhk8RWGVFLpuSKoA@mail.gmail.com>
	<0E0D5ED7-E97F-4F17-8290-2D893649A8F8@jerris.com>
	<CAOP4xcwBNPxaxsrFBr48-02QriO_baDv-=o+QhoDsiMKUnq4ug@mail.gmail.com>
	<CAKbxfG8zu0+_AM4zzmAq01BFHxpLygb8-2JdXNujOw6X6UMesw@mail.gmail.com>
	<CADk4Sr1g2c3toiU7+ikM_WByfUzvxwXKQwLrFXF8KnrtZoXWnQ@mail.gmail.com>
	<CAOP4xcy9p1PvL6ggmWK=betiGb=N=2DENvMA=ieiSNo0mZF9Rw@mail.gmail.com>
	<2ECFB1E034254C48ACBDD31C7E189BA9@gmail.com>
	<3B81AEAC-802B-4B56-A7E3-739B78895D42@jerris.com>
Message-ID: <CAKbxfG8R_SNuFAHdVODaF-Gw-6gA73v_xqT3bDruS_Wg7ndcpw@mail.gmail.com>

Is there a a real significance to it being 12 seconds of delay?  The people
listening will not know it.  I am just wondering if there is a tangible
problem with it or if its just obsession when listening to it in a test env.




On Mon, Aug 19, 2013 at 8:40 AM, Michael Jerris <mike at jerris.com> wrote:

> But this would still include buffering? although it should be less than 12
> seconds.
>
> On Aug 16, 2013, at 8:04 PM, Seven Du <dujinfang at gmail.com> wrote:
>
>  Another solution would to use flash with mod_rtmp.  And to make that work
> better you need to change the flash code to remove the sound input device
> since you don't need it and this also helps to remove the confirmation
> prompt anytime you use a sound input in flash.
>
> I think I had written a plugin for flowplayer long time ago which works
> well.
>
>
> --
> Seven Du
> http://www.freeswitch.org.cn
> http://about.me/dujinfang
> http://www.dujinfang.com
>
> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>
> On Saturday, August 17, 2013 at 2:22 AM, Jorge N??ez wrote:
>
> So Can anybody tell any other solution instead of webrtc and icecast to
> broadcast a conference in realtime without delay or maybe just a little bit
> of delay 1s or 2s?
>
> Thank you very much for your help
>
> Jorge
>
>
> 2013/8/15 Jeff Leung <jleung at v10networks.ca>
>
> Icecast is known to have a relatively high latency for audio.
>
>
> On Thu, Aug 15, 2013 at 10:34 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> enable_file_write_buffering=false
> stream_prebuffer=0
>
> On the leg doing the recording.  It does't stop the iceast server or
> libshout from buffering it more.
>
>
>
>
>
> On Thu, Aug 15, 2013 at 12:07 PM, Jorge N??ez <jorgefren12 at gmail.com>wrote:
>
> Anthony and just for test how can I reduce the latency? and thanks Michael
> for use webrtc Do I need to use Sipml5 or JSSIP o are there any othe way to
> use because I was trying to find some information about mod_html5 but I
> didn?t find
>
> thanks
>
> Jorge
>
>
> 2013/8/15 Michael Jerris <mike at jerris.com>
>
> The alternative if you want in sync audio would be to use webrtc endpoints
> to listen.
>
> On Aug 14, 2013, at 7:19 PM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
> Icecast is not designed for low-latency and realtime audio.  Its designed
> for higher quality reliable audio.  The servers and the client libs both
> use latency and buffering to guarantee this.  There are some techniques to
> reduce the buffering on at least the FS side but then you will start
> getting skips and resets if you miss any packets.
>
> This comes up all the time when people first try doing this.  Either you
> need to just accept the delay since most people will not even know its
> there or use some other method.
>
>
>
>
> On Wed, Aug 14, 2013 at 5:12 PM, Stephen Dame <sdame at 207me.com> wrote:
>
> Did you restart the freeswitch server after changes to the config files,
> I?m not sure 100% if reloadxml resets the conference settings. ****
> ** **
> Also I use some flash based players, I think there where client settiings
> for buffering.   Can you set buffer in HTML tag?****
> ** **
> Regards,****
> Stephen****
> ** **
> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
> ** **
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jorge N??ez
> *Sent:* Wednesday, August 14, 2013 4:18 PM
>
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big delay**
> **
>
> ** **
> Hi thanks, I set your configuration but nothing changed, I reduced the
> burst size and it takes me just 11s and also I tried with 8k, 16k and 32k
> and nothing change****
> ** **
>     <limits>****
>         <clients>100</clients>****
>         <sources>10</sources>****
>         <threadpool>5</threadpool>****
>         <queue-size>524288</queue-size>****
>         <!-- <queue-size>524288</queue-size> -->****
>         <client-timeout>30</client-timeout>****
>         <header-timeout>15</header-timeout>****
>         <source-timeout>10</source-timeout>****
>         <!-- If enabled, this will provide a burst of data when a client *
> ***
>              first connects, thereby significantly reducing the startup **
> **
>              time for listeners that do substantial buffering. However,***
> *
>              it also significantly increases latency between the source***
> *
>              client and listening client.  For low-latency setups, you****
>              might want to disable this. -->****
>         <burst-on-connect>0</burst-on-connect>****
>         <!-- same as burst-on-connect, but this allows for being more****
>              specific on how much to burst. Most people won't need to****
>              change from the default 64k. Applies to all mountpoints ****
> ** **
>  -->****
>         <burst-size>4096</burst-size>****
>     </limits>****
> ** **
> Regards****
> ** **
> Jorge****
> ** **
> ** **
> 2013/8/14 Stephen Dame <sdame at 207me.com>****
>
> Jorge, ****
>  ****
> Play around with the burst size and queue size?  here is my xml config
> around 3-4 second delay from 16k freeswitch conference. ****
> To small a buffer and the players disconnect? Im also running icecast on
> same server.****
>  ****
> <icecast>****
>     <limits>****
>         <clients>100</clients>****
>         <sources>10</sources>****
>         <threadpool>5</threadpool>****
>         <queue-size>524288</queue-size>****
>         <client-timeout>30</client-timeout>****
>         <header-timeout>15</header-timeout>****
>         <source-timeout>10</source-timeout>****
>         <burst-on-connect>1</burst-on-connect>****
>         <!-- same as burst-on-connect, but this allows for being more****
>              specific on how much to burst. Most people won't need to****
>              change from the default 64k. Applies to all mountpoints  -->*
> ***
>         <burst-size>65535</burst-size>****
>     </limits>****
>  ****
> Regards,****
> Stephen****
>  ****
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice
> *Sent:* Wednesday, August 14, 2013 12:35 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big delay**
> **
>  ****
> You have to keep in mind that icecast itself has a fair bit of buffering
> involved with it and theres not a lot you can do about that
>
>
> On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com> wrote:****
>
> Hi I am using mod_shout to send a conference call to icecast and hear it
> from a tag audio in html5 in realtime, but it has a big delay like 12
> seconds. How can I reduce the latency of the audio sent from freeswitch or
> what can I do to improve this.
>
> Regards
>
> Jorge****
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
>
> --
> Ken
> *http://www.FreeSWITCH.org <http://www.freeswitch.org/>
> http://www.ClueCon.com <http://www.cluecon.com/>
> http://www.OSTAG.org <http://www.ostag.org/>
> *irc.freenode.net #freeswitch****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
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>
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>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> 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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
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> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> 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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
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>
>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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> http://www.freeswitch.org
>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
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> http://www.freeswitch.org
>
>
>  _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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>
>
>
> _________________________________________________________________________
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> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
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>
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>


-- 
Anthony Minessale II

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 william.king at quentustech.com  Mon Aug 19 19:24:34 2013
From: william.king at quentustech.com (William King)
Date: Mon, 19 Aug 2013 08:24:34 -0700
Subject: [Freeswitch-users] mod_sms endpoints
In-Reply-To: <075501ce9c2b$07edb9f0$17c92dd0$@experient.com>
References: <06df01ce9ad4$1018d450$304a7cf0$@experient.com>	<CAMeZp63hV6_QQQJ52ooc7+0dprYRyXcX4ZUa4yM8WtAx3p0ykw@mail.gmail.com>	<071601ce9b6a$e7219720$b564c560$@experient.com>
	<521020CB.3000101@quentustech.com>
	<075501ce9c2b$07edb9f0$17c92dd0$@experient.com>
Message-ID: <521238B2.2090400@quentustech.com>

If you have a lua script in your chat plan that returns successful, that
counts as an endpoint delivery.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 08/18/2013 08:53 AM, Bob McCarthy wrote:
> This is the error->
> 
> 2013-08-18 09:51:46.317229 [ERR] sofia_presence.c:265 Chat proto [sip]
> from ["Win 7" <sip:3038183060 at 192.168.1.8:5060>;tag=f402884a]
> to [911 at 192.168.1.212]
> <font face="Arial, sans-serif"
> size="2">hello</font><L><http://lis.ddti.net:56074/?LbyR=39b93693-a22a-45bd-
> ab42-7c91c3ad2f6e><l><B>C0-12-34-56-78-90<b>
> Nobody to send to: Profile internal
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William
> King
> Sent: Saturday, August 17, 2013 7:18 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_sms endpoints
> 
> What's the error message you are getting?
> 
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
> 
> On 08/17/2013 09:57 AM, Bob McCarthy wrote:
>> What I end up doing is deleteing the to header and replacing it with a 
>> registered user.  But I would rather not.
>>
>>  
>>
>> *From:*freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
>> *Daniel Ivanov
>> *Sent:* Saturday, August 17, 2013 1:07 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] mod_sms endpoints
>>
>>  
>>
>> I haven't seen fs complain about a message not been sent in the 
>> chatplan. Just process it via a lua handler and do whatever you want 
>> with it.
>>
>> On Aug 17, 2013 2:03 AM, "Bob McCarthy" <bob.mccarthy at experient.com 
>> <mailto:bob.mccarthy at experient.com>> wrote:
>>
>>     Is it possible to create a fictious endpoint to route sms messages
>>     to ? using mod_sms to send messages,  I am intercepting the messages
>>     and disseminating them via the event socket. By sending them to
>>     nowhere I get a nuisance error message.
>>
>>      
>>
>>     Bob McCarthy
>>
>>
>>
> _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.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
>>
>>
>>
>> ______________________________________________________________________
>> ___ Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>  
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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-use
>> rs
>> http://www.freeswitch.org
>>
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mike at jerris.com  Mon Aug 19 19:51:05 2013
From: mike at jerris.com (Michael Jerris)
Date: Mon, 19 Aug 2013 11:51:05 -0400
Subject: [Freeswitch-users] Call not hanging up
In-Reply-To: <CABkS=xoKBtF6zP4BzWb8tiNzpK2N-yLM6F1Jqhfz26fPcnO34Q@mail.gmail.com>
References: <CABkS=xoKBtF6zP4BzWb8tiNzpK2N-yLM6F1Jqhfz26fPcnO34Q@mail.gmail.com>
Message-ID: <210DAC89-E7C0-4417-ACFE-2A77E424C934@jerris.com>

Check out sip trace.. sounds like a BYE is not getting somewhere.

On Aug 19, 2013, at 2:02 AM, Peter <eidevm5 at gmail.com> wrote:

> Running FS 1.2.12 in the following environment:
> 
> FS Ext  <--->  FS SBC <---> FS Int
> 
> I can successfully make calls in both directions.
> 
> If the caller hangs up, the other end correctly hangs up.
> 
> However, if the callee hangs up, the caller session remains active until the RTP timeout kicks in.
> 
> Here's a snippet from the FS log where this occurs (1004 is the caller and 2010 is the callee)
> 
> [NOTICE] sofia.c:716 Hangup sofia/internal/2010 at 10.1.1.206:5061 [CS_HIBERNATE] [NORMAL_CLEARING]
> [NOTICE] switch_ivr_bridge.c:1109 Hangup sofia/internal/1004 at 10.1.1.204 [CS_HIBERNATE] [NORMAL_CLEARING]
> [NOTICE] switch_core_session.c:1560 Session 4 (sofia/internal/2010 at 10.1.1.206:5061) Ended
> [NOTICE] switch_core_session.c:1564 Close Channel sofia/internal/2010 at 10.1.1.206:5061 [CS_DESTROY]
> [NOTICE] switch_core_session.c:1560 Session 3 (sofia/internal/1004 at 10.1.1.104) Ended
> [NOTICE] switch_core_session.c:1564 Close Channel sofia/internal/1004 at 10.1.1.104 [CS_DESTROY]
> 
> I've tried setting hangup_after_bridge=true in the dialplan on FS Ext.    I wasn't 100% sure whether it should go on FS Ext or FS SBC, so I tried both and it made no difference.
> 
> What else should I be checking?
> 

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From michel.brabants at gmail.com  Mon Aug 19 19:53:11 2013
From: michel.brabants at gmail.com (Michel Brabants)
Date: Mon, 19 Aug 2013 17:53:11 +0200
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <57B0D003-A269-448E-A323-3E8F3B6DD8C0@insensate.co.uk>
References: <CE33869E.78FAE%krice@freeswitch.org>
	<57B0D003-A269-448E-A323-3E8F3B6DD8C0@insensate.co.uk>
Message-ID: <CAHt+dNH3sFf2So6EszoAZp3E74c5y+cqARG-_eadbPw71e5foQ@mail.gmail.com>

Polycoms work fine with freeswitch. The bootserver-config contains more
options than you can configure using the phone, but you also have the
builtin webtool of the phone, but I didn't use that a lot yet. A bootserver
with a configfile gives you all the "power"..., but it isn't need to
register a phone with freeswitch. You can do that from the phone or the
web-interface of the polycom.

Michel

On Fri, Aug 16, 2013 at 3:26 PM, Lawrence Conroy <lconroy at insensate.co.uk>wrote:

> Hi there again folks,
>  for clarification, I don't know about the "pure SIP" versions or Poly
> deskphones
> -- I whinged only about the Lync variants of their DECT phones.
> In my case, once you've had one of your users blame you for the setup and
> describe the
> Polycom kirk butterfly as looking like a sanitary towel (only less
> useful), it's hard
> to retain your enthusiasm.
>
> Do their pure SIP phones really need a separate boot server to be set up
> before they'll
> work properly? For central config sure, but for general operation of
> individual phones?
>
> Re. SIP Deskphones, the Aastras seem to be popular on site and work quite
> well.
> We have a couple of hundred of them, and with no failures (at all) yet, so
> they must be
> pretty robust.
>
> Older Cisco desk phones have left me with mental scars -- don't know about
> the new ones.
> [In the early days, Cisco's interpretation of SIP was kinda different,
> causing many debug
> sessions to find out what they needed]
> Grandstream 2K series worked, but suffered from unacceptable mains hum and
> odd little
> incompatibilities with fS.
> The Siemens/Gigaset SIP desk phones were pretty boring, IIRC, but had
> pretty good sound
>  quality.
> For the others, can't say.
>
> all the best,
>   Lawrence
>
>
> On 16 Aug 2013, at 13:34, Ken Rice wrote:
> > I?ve never found polycoms hard to debug or configure, you just need to
> set
> > up a proper boot server for them... Once you do that, the configs are
> > straight forward, and it?ll actually logs back to that server...
> >
> >
> > On 8/16/13 7:25 AM, "Peter Blackford" <peter at onemetric.com> wrote:
> >
> >> I HATE Polycoms they are hard to configure and hard to debug.
> >>
> >> I really like the Cisco SPA series as they are reliable however also
> hard to
> >> debug. Snoms and Yealinks are good and a lot easier to debug (Pulling
> packet
> >> captures directly from the phone makes life a lot easier)
> >>
> >> Just my 2c.
> >>
> >>
> >> On 16 August 2013 20:35, Lawrence Conroy <lconroy at insensate.co.uk>
> wrote:
> >>> Hi Folks,
> >>>  +3 for the Gigasets. I have a number of these setups -- with different
> >>> handsets.
> >>> These are working fine with fS in all of my servers.
> >>> TL;DR: I paid my own money for two of these for my own use. My wife
> doesn't
> >>> know
> >>>  she's using SIP for all calls -- it's just a 'phone.
> >>>
> >>> Voice Quality seems to be good, range is excellent, battery life is
> fine
> >>> (though
> >>> they spend most of their time in the desk stands, charging).
> >>> I have not used bluetooth headsets with them, so can't talk on that.
> The 2.5
> >>> mm
> >>> audio connector works fine (some folk prefer clipping the phone to
> their
> >>> waist
> >>> and wearing a headset -- YMMV).
> >>> The N300A (consumer DECT SIP base station) voicemail is OK, but IMHO
> >>> over-compresses.
> >>> The in-built web server by which the base is configured is OK if slow.
> >>> Configuring the base via a connected handset is possible, but like
> every
> >>> manufacturer's
> >>> version of that I've seen, it's not recommended for mere mortals
> without a
> >>> lot of patience.
> >>> [...and apart from the annoyance of the battery compartment door
> falling open
> >>> if you
> >>> drop them, and the handset NOT being waterproof/toiletproof, they're
> >>> remarkably robust]
> >>>
> >>> As for Polycom, can't speak for the "straight" SIP versions, but I have
> >>> experience
> >>>  of their Lync DECT multicell system & handsets; IMHO these don't even
> reach
> >>> the
> >>>  'blows goats' level. Work has 50 of them; user feedback is that voice
> >>> quality,
> >>>  coverage, battery life is "not as good as we might hope".
> >>> Opinion seems to be that these are a step back from the siemens
> cellular DECT
> >>> 'phones we had on the old "steam" PBX (being polite).
> >>> Sigh.  For all portable systems, Try before you buy.
> >>>
> >>> all the best,
> >>>  Lawrence
> >>>
> >>>
> >>> On 16 Aug 2013, at 10:15, Andrew Cassidy wrote:
> >>>>> Polycom is the name I see the most on this list, I've some some
> experience
> >>>>> with Snom and Grandstream. The do the job but my favourites are Cisco
> >>>>> SPA50x series.
> >>>>>
> >>>>>
> >>>>> On 16 August 2013 08:22, Stanislav Sinyagin <ssinyagin at yahoo.com>
> wrote:
> >>>>>
> >>>>>>> Gigaset 610IP is a great cordless ip-phone. Works perfectly with
> >>>>>>> FreeSWITCH. You can also have several handsets served from one
> base.
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>  ------------------------------
> >>>>>>> *From:* Karl Schmidt <karl at xtronics.com>
> >>>>>>> *To:* freeswitch-users at lists.freeswitch.org
> >>>>>>> *Sent:* Friday, August 16, 2013 5:36 AM
> >>>>>>> *Subject:* [Freeswitch-users] SIP phones -
> >>>>>>>
> >>>>>>> I am thinking of getting a couple of SIP phones for our system - I
> see
> >>>>>>> some used on ebay.
> >>>>>>>
> >>>>>>> Are there ones to avoid? Good ones to seek?
> >>>>>>>
> >>>>>>> I have a couple of places where there needs to be a backup phone
> in case
> >>>>>>> the workstation is
> >>>>>>> rebooting. I can set up POE if that sounds like a good idea.
> >>>>>>>
> >>>>>>> Depending on what happens, I could then start retiring some of the
> >>>>> analog
> >>>>>>> phones replacing with SIP
> >>>>>>> phones.
> >>>>>>>
> >>>>>>> The bits I'm finding on the web is almost all by someone selling
> and
> >>>>>>> mostly spammy information. A
> >>>>>>> few clues could go a long way.
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>
> ---------------------------------------------------------------------------
> >>>>> -----
> >>>>>>> Karl Schmidt                                  EMail
> Karl at xtronics.com
> >>>>>>> Transtronics, Inc.                              WEB
> >>>>>>> http://secure.transtronics.com
> >>>>>>> 3209 West 9th Street                            Ph (785) 841-3089
> >>>>> <tel:%28785%29%20841-3089>
> >>>>>>> Lawrence, KS 66049                              FAX (785) 841-0434
> >>>>> <tel:%28785%29%20841-0434>
> >>>>>>>
> >>>>>>> Never ruin an apology with an excuse.
> >>>>>>>
> >>>>>>>
> >>>>>
> ---------------------------------------------------------------------------
> >>>>> -----
> >>>>>>>
> >>>>>>>
> >>>>>
> _________________________________________________________________________
> >>>>>>> Professional FreeSWITCH Consulting Services:
> >>>>>>> consulting at freeswitch.org
> >>>>>>> http://www.freeswitchsolutions.com
> >>>>>>>
> >>>>>>> 
> >>>>>>> 
> >>>>>>>
> >>>>>>> Official FreeSWITCH Sites
> >>>>>>> http://www.freeswitch.org
> >>>>>>> http://wiki.freeswitch.org
> >>>>>>> http://www.cluecon.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
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>>>
> >>>>>
> _________________________________________________________________________
> >>>>>>> Professional FreeSWITCH Consulting Services:
> >>>>>>> consulting at freeswitch.org
> >>>>>>> http://www.freeswitchsolutions.com
> >>>>>>>
> >>>>>>> 
> >>>>>>> 
> >>>>>>>
> >>>>>>> Official FreeSWITCH Sites
> >>>>>>> http://www.freeswitch.org
> >>>>>>> http://wiki.freeswitch.org
> >>>>>>> http://www.cluecon.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
> >>>>>>>
> >>>>>>>
> >>>>>
> >>>>>
> >>>>> --
> >>>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> >>>>> Managing Director
> >>>>>
> >>>>>
> >>>>> *T <info at cassidywebservices.co.uk> *03300 100 960
> >>>>> *F<info at cassidywebservices.co.uk>
> >>>>> *03300 100 961
> >>>>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> >>>>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
> >>>> <http://www.cassidywebservices.co.uk>
> >>>>>
> _________________________________________________________________________
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>>
> >>>>> 
> >>>>> 
> >>>>>
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://wiki.freeswitch.org
> >>>>> http://www.cluecon.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
> >>>
> >>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.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
> >>
> >>
> >>
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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
> >
> > --
> > Ken
> > http://www.FreeSWITCH.org <http://www.freeswitch.org/>
> > http://www.ClueCon.com <http://www.cluecon.com/>
> > http://www.OSTAG.org <http://www.ostag.org/>
> > irc.freenode.net #freeswitch
> >
>  >
> _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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.brabants at gmail.com  Mon Aug 19 19:56:54 2013
From: michel.brabants at gmail.com (Michel Brabants)
Date: Mon, 19 Aug 2013 17:56:54 +0200
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <CAHt+dNH3sFf2So6EszoAZp3E74c5y+cqARG-_eadbPw71e5foQ@mail.gmail.com>
References: <CE33869E.78FAE%krice@freeswitch.org>
	<57B0D003-A269-448E-A323-3E8F3B6DD8C0@insensate.co.uk>
	<CAHt+dNH3sFf2So6EszoAZp3E74c5y+cqARG-_eadbPw71e5foQ@mail.gmail.com>
Message-ID: <CAHt+dNHYN85jK-BmZF-MvsqwWmp=nu07pM6XQM-c-xBXwqUE6A@mail.gmail.com>

Adding to that. I can give you some scenario's in which the polycoms can
crash in a HA-setup with 2 freeswitches (with firmware version 4), but
there are way around it without reducing functionality (as far as we needed
of course, but call-handling is fine. Didn't test BLF).

Michel



On Mon, Aug 19, 2013 at 5:53 PM, Michel Brabants
<michel.brabants at gmail.com>wrote:

> Polycoms work fine with freeswitch. The bootserver-config contains more
> options than you can configure using the phone, but you also have the
> builtin webtool of the phone, but I didn't use that a lot yet. A bootserver
> with a configfile gives you all the "power"..., but it isn't need to
> register a phone with freeswitch. You can do that from the phone or the
> web-interface of the polycom.
>
> Michel
>
>  On Fri, Aug 16, 2013 at 3:26 PM, Lawrence Conroy <lconroy at insensate.co.uk
> > wrote:
>
>> Hi there again folks,
>>  for clarification, I don't know about the "pure SIP" versions or Poly
>> deskphones
>> -- I whinged only about the Lync variants of their DECT phones.
>> In my case, once you've had one of your users blame you for the setup and
>> describe the
>> Polycom kirk butterfly as looking like a sanitary towel (only less
>> useful), it's hard
>> to retain your enthusiasm.
>>
>> Do their pure SIP phones really need a separate boot server to be set up
>> before they'll
>> work properly? For central config sure, but for general operation of
>> individual phones?
>>
>> Re. SIP Deskphones, the Aastras seem to be popular on site and work quite
>> well.
>> We have a couple of hundred of them, and with no failures (at all) yet,
>> so they must be
>> pretty robust.
>>
>> Older Cisco desk phones have left me with mental scars -- don't know
>> about the new ones.
>> [In the early days, Cisco's interpretation of SIP was kinda different,
>> causing many debug
>> sessions to find out what they needed]
>> Grandstream 2K series worked, but suffered from unacceptable mains hum
>> and odd little
>> incompatibilities with fS.
>> The Siemens/Gigaset SIP desk phones were pretty boring, IIRC, but had
>> pretty good sound
>>  quality.
>> For the others, can't say.
>>
>> all the best,
>>   Lawrence
>>
>>
>> On 16 Aug 2013, at 13:34, Ken Rice wrote:
>> > I?ve never found polycoms hard to debug or configure, you just need to
>> set
>> > up a proper boot server for them... Once you do that, the configs are
>> > straight forward, and it?ll actually logs back to that server...
>> >
>> >
>> > On 8/16/13 7:25 AM, "Peter Blackford" <peter at onemetric.com> wrote:
>> >
>> >> I HATE Polycoms they are hard to configure and hard to debug.
>> >>
>> >> I really like the Cisco SPA series as they are reliable however also
>> hard to
>> >> debug. Snoms and Yealinks are good and a lot easier to debug (Pulling
>> packet
>> >> captures directly from the phone makes life a lot easier)
>> >>
>> >> Just my 2c.
>> >>
>> >>
>> >> On 16 August 2013 20:35, Lawrence Conroy <lconroy at insensate.co.uk>
>> wrote:
>> >>> Hi Folks,
>> >>>  +3 for the Gigasets. I have a number of these setups -- with
>> different
>> >>> handsets.
>> >>> These are working fine with fS in all of my servers.
>> >>> TL;DR: I paid my own money for two of these for my own use. My wife
>> doesn't
>> >>> know
>> >>>  she's using SIP for all calls -- it's just a 'phone.
>> >>>
>> >>> Voice Quality seems to be good, range is excellent, battery life is
>> fine
>> >>> (though
>> >>> they spend most of their time in the desk stands, charging).
>> >>> I have not used bluetooth headsets with them, so can't talk on that.
>> The 2.5
>> >>> mm
>> >>> audio connector works fine (some folk prefer clipping the phone to
>> their
>> >>> waist
>> >>> and wearing a headset -- YMMV).
>> >>> The N300A (consumer DECT SIP base station) voicemail is OK, but IMHO
>> >>> over-compresses.
>> >>> The in-built web server by which the base is configured is OK if slow.
>> >>> Configuring the base via a connected handset is possible, but like
>> every
>> >>> manufacturer's
>> >>> version of that I've seen, it's not recommended for mere mortals
>> without a
>> >>> lot of patience.
>> >>> [...and apart from the annoyance of the battery compartment door
>> falling open
>> >>> if you
>> >>> drop them, and the handset NOT being waterproof/toiletproof, they're
>> >>> remarkably robust]
>> >>>
>> >>> As for Polycom, can't speak for the "straight" SIP versions, but I
>> have
>> >>> experience
>> >>>  of their Lync DECT multicell system & handsets; IMHO these don't
>> even reach
>> >>> the
>> >>>  'blows goats' level. Work has 50 of them; user feedback is that voice
>> >>> quality,
>> >>>  coverage, battery life is "not as good as we might hope".
>> >>> Opinion seems to be that these are a step back from the siemens
>> cellular DECT
>> >>> 'phones we had on the old "steam" PBX (being polite).
>> >>> Sigh.  For all portable systems, Try before you buy.
>> >>>
>> >>> all the best,
>> >>>  Lawrence
>> >>>
>> >>>
>> >>> On 16 Aug 2013, at 10:15, Andrew Cassidy wrote:
>> >>>>> Polycom is the name I see the most on this list, I've some some
>> experience
>> >>>>> with Snom and Grandstream. The do the job but my favourites are
>> Cisco
>> >>>>> SPA50x series.
>> >>>>>
>> >>>>>
>> >>>>> On 16 August 2013 08:22, Stanislav Sinyagin <ssinyagin at yahoo.com>
>> wrote:
>> >>>>>
>> >>>>>>> Gigaset 610IP is a great cordless ip-phone. Works perfectly with
>> >>>>>>> FreeSWITCH. You can also have several handsets served from one
>> base.
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>  ------------------------------
>> >>>>>>> *From:* Karl Schmidt <karl at xtronics.com>
>> >>>>>>> *To:* freeswitch-users at lists.freeswitch.org
>> >>>>>>> *Sent:* Friday, August 16, 2013 5:36 AM
>> >>>>>>> *Subject:* [Freeswitch-users] SIP phones -
>> >>>>>>>
>> >>>>>>> I am thinking of getting a couple of SIP phones for our system -
>> I see
>> >>>>>>> some used on ebay.
>> >>>>>>>
>> >>>>>>> Are there ones to avoid? Good ones to seek?
>> >>>>>>>
>> >>>>>>> I have a couple of places where there needs to be a backup phone
>> in case
>> >>>>>>> the workstation is
>> >>>>>>> rebooting. I can set up POE if that sounds like a good idea.
>> >>>>>>>
>> >>>>>>> Depending on what happens, I could then start retiring some of the
>> >>>>> analog
>> >>>>>>> phones replacing with SIP
>> >>>>>>> phones.
>> >>>>>>>
>> >>>>>>> The bits I'm finding on the web is almost all by someone selling
>> and
>> >>>>>>> mostly spammy information. A
>> >>>>>>> few clues could go a long way.
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>
>> ---------------------------------------------------------------------------
>> >>>>> -----
>> >>>>>>> Karl Schmidt                                  EMail
>> Karl at xtronics.com
>> >>>>>>> Transtronics, Inc.                              WEB
>> >>>>>>> http://secure.transtronics.com
>> >>>>>>> 3209 West 9th Street                            Ph (785) 841-3089
>> >>>>> <tel:%28785%29%20841-3089>
>> >>>>>>> Lawrence, KS 66049                              FAX (785) 841-0434
>> >>>>> <tel:%28785%29%20841-0434>
>> >>>>>>>
>> >>>>>>> Never ruin an apology with an excuse.
>> >>>>>>>
>> >>>>>>>
>> >>>>>
>> ---------------------------------------------------------------------------
>> >>>>> -----
>> >>>>>>>
>> >>>>>>>
>> >>>>>
>> _________________________________________________________________________
>> >>>>>>> Professional FreeSWITCH Consulting Services:
>> >>>>>>> consulting at freeswitch.org
>> >>>>>>> http://www.freeswitchsolutions.com
>> >>>>>>>
>> >>>>>>> 
>> >>>>>>> 
>> >>>>>>>
>> >>>>>>> Official FreeSWITCH Sites
>> >>>>>>> http://www.freeswitch.org
>> >>>>>>> http://wiki.freeswitch.org
>> >>>>>>> http://www.cluecon.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
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>>>
>> >>>>>
>> _________________________________________________________________________
>> >>>>>>> Professional FreeSWITCH Consulting Services:
>> >>>>>>> consulting at freeswitch.org
>> >>>>>>> http://www.freeswitchsolutions.com
>> >>>>>>>
>> >>>>>>> 
>> >>>>>>> 
>> >>>>>>>
>> >>>>>>> Official FreeSWITCH Sites
>> >>>>>>> http://www.freeswitch.org
>> >>>>>>> http://wiki.freeswitch.org
>> >>>>>>> http://www.cluecon.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
>> >>>>>>>
>> >>>>>>>
>> >>>>>
>> >>>>>
>> >>>>> --
>> >>>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> >>>>> Managing Director
>> >>>>>
>> >>>>>
>> >>>>> *T <info at cassidywebservices.co.uk> *03300 100 960
>> >>>>> *F<info at cassidywebservices.co.uk>
>> >>>>> *03300 100 961
>> >>>>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>> >>>>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>> >>>> <http://www.cassidywebservices.co.uk>
>> >>>>>
>> _________________________________________________________________________
>> >>>>> Professional FreeSWITCH Consulting Services:
>> >>>>> consulting at freeswitch.org
>> >>>>> http://www.freeswitchsolutions.com
>> >>>>>
>> >>>>> 
>> >>>>> 
>> >>>>>
>> >>>>> Official FreeSWITCH Sites
>> >>>>> http://www.freeswitch.org
>> >>>>> http://wiki.freeswitch.org
>> >>>>> http://www.cluecon.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
>> >>>
>> >>>
>> >>>
>> _________________________________________________________________________
>> >>> Professional FreeSWITCH Consulting Services:
>> >>> consulting at freeswitch.org
>> >>> http://www.freeswitchsolutions.com
>> >>>
>> >>> 
>> >>> 
>> >>>
>> >>> Official FreeSWITCH Sites
>> >>> http://www.freeswitch.org
>> >>> http://wiki.freeswitch.org
>> >>> http://www.cluecon.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
>> >>
>> >>
>> >>
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.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
>> >
>> > --
>> > Ken
>> > http://www.FreeSWITCH.org <http://www.freeswitch.org/>
>> > http://www.ClueCon.com <http://www.cluecon.com/>
>> > http://www.OSTAG.org <http://www.ostag.org/>
>> > irc.freenode.net #freeswitch
>> >
>>  >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 jorgefren12 at gmail.com  Mon Aug 19 20:11:58 2013
From: jorgefren12 at gmail.com (=?ISO-8859-1?Q?Jorge_N=FA=F1ez?=)
Date: Mon, 19 Aug 2013 11:11:58 -0500
Subject: [Freeswitch-users] Mod_shout using with icecast Big delay
In-Reply-To: <CAKbxfG8R_SNuFAHdVODaF-Gw-6gA73v_xqT3bDruS_Wg7ndcpw@mail.gmail.com>
References: <CAOP4xcxkZUjM58aKyb160GzD-_dK3J4NgT_nkbiuyEueN+dPUw@mail.gmail.com>
	<CE311BFC.78EA0%krice@freeswitch.org>
	<045801ce990e$205b4f90$6111eeb0$@207me.com>
	<CAOP4xcyf2yA1XguRT-wHRUu-QNfkeMCMNB4_666zLeC1yYs_4A@mail.gmail.com>
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	<CAOP4xcy6NZG0JKPMYhJp_wxaLqnAjsEkGMgTD5itvBPNdSCPzQ@mail.gmail.com>
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	<CAKbxfG80vr3b+BULjn5ZhKW6ZjxY6Khy4Ymhk8RWGVFLpuSKoA@mail.gmail.com>
	<0E0D5ED7-E97F-4F17-8290-2D893649A8F8@jerris.com>
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Message-ID: <CAOP4xcy_goaa0FmBADAmxpXM5fhHLTUg-9ss_n3N1LHvyNMdBg@mail.gmail.com>

Yes it is!! because I have a live chat and other stuff working in real
time. So the voice wont have a big delay respect the other stuff.

I am using node,js and websockets for the html5 client thats why I want to
use the audio tag instead a flash player or something else

Thanks

Jorge




2013/8/19 Anthony Minessale <anthony.minessale at gmail.com>

> Is there a a real significance to it being 12 seconds of delay?  The
> people listening will not know it.  I am just wondering if there is a
> tangible problem with it or if its just obsession when listening to it in a
> test env.
>
>
>
>
> On Mon, Aug 19, 2013 at 8:40 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> But this would still include buffering? although it should be less than
>> 12 seconds.
>>
>> On Aug 16, 2013, at 8:04 PM, Seven Du <dujinfang at gmail.com> wrote:
>>
>>  Another solution would to use flash with mod_rtmp.  And to make that
>> work better you need to change the flash code to remove the sound input
>> device since you don't need it and this also helps to remove the
>> confirmation prompt anytime you use a sound input in flash.
>>
>> I think I had written a plugin for flowplayer long time ago which works
>> well.
>>
>>
>> --
>> Seven Du
>> http://www.freeswitch.org.cn
>> http://about.me/dujinfang
>> http://www.dujinfang.com
>>
>> Sent with Sparrow <http://www.sparrowmailapp.com/?sig>
>>
>> On Saturday, August 17, 2013 at 2:22 AM, Jorge N??ez wrote:
>>
>> So Can anybody tell any other solution instead of webrtc and icecast to
>> broadcast a conference in realtime without delay or maybe just a little bit
>> of delay 1s or 2s?
>>
>> Thank you very much for your help
>>
>> Jorge
>>
>>
>> 2013/8/15 Jeff Leung <jleung at v10networks.ca>
>>
>> Icecast is known to have a relatively high latency for audio.
>>
>>
>> On Thu, Aug 15, 2013 at 10:34 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>> enable_file_write_buffering=false
>> stream_prebuffer=0
>>
>> On the leg doing the recording.  It does't stop the iceast server or
>> libshout from buffering it more.
>>
>>
>>
>>
>>
>> On Thu, Aug 15, 2013 at 12:07 PM, Jorge N??ez <jorgefren12 at gmail.com>wrote:
>>
>> Anthony and just for test how can I reduce the latency? and thanks
>> Michael for use webrtc Do I need to use Sipml5 or JSSIP o are there any
>> othe way to use because I was trying to find some information about
>> mod_html5 but I didn?t find
>>
>> thanks
>>
>> Jorge
>>
>>
>> 2013/8/15 Michael Jerris <mike at jerris.com>
>>
>> The alternative if you want in sync audio would be to use webrtc
>> endpoints to listen.
>>
>> On Aug 14, 2013, at 7:19 PM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>> Icecast is not designed for low-latency and realtime audio.  Its designed
>> for higher quality reliable audio.  The servers and the client libs both
>> use latency and buffering to guarantee this.  There are some techniques to
>> reduce the buffering on at least the FS side but then you will start
>> getting skips and resets if you miss any packets.
>>
>> This comes up all the time when people first try doing this.  Either you
>> need to just accept the delay since most people will not even know its
>> there or use some other method.
>>
>>
>>
>>
>> On Wed, Aug 14, 2013 at 5:12 PM, Stephen Dame <sdame at 207me.com> wrote:
>>
>> Did you restart the freeswitch server after changes to the config
>> files,   I?m not sure 100% if reloadxml resets the conference settings. *
>> ***
>> ** **
>> Also I use some flash based players, I think there where client settiings
>> for buffering.   Can you set buffer in HTML tag?****
>> ** **
>> Regards,****
>> Stephen****
>> ** **
>> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>> ** **
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jorge N??ez
>> *Sent:* Wednesday, August 14, 2013 4:18 PM
>>
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big delay*
>> ***
>>
>> ** **
>> Hi thanks, I set your configuration but nothing changed, I reduced the
>> burst size and it takes me just 11s and also I tried with 8k, 16k and 32k
>> and nothing change****
>> ** **
>>     <limits>****
>>         <clients>100</clients>****
>>         <sources>10</sources>****
>>         <threadpool>5</threadpool>****
>>         <queue-size>524288</queue-size>****
>>         <!-- <queue-size>524288</queue-size> -->****
>>         <client-timeout>30</client-timeout>****
>>         <header-timeout>15</header-timeout>****
>>         <source-timeout>10</source-timeout>****
>>         <!-- If enabled, this will provide a burst of data when a client
>> ****
>>              first connects, thereby significantly reducing the startup *
>> ***
>>              time for listeners that do substantial buffering. However,**
>> **
>>              it also significantly increases latency between the source**
>> **
>>              client and listening client.  For low-latency setups, you***
>> *
>>              might want to disable this. -->****
>>          <burst-on-connect>0</burst-on-connect>****
>>         <!-- same as burst-on-connect, but this allows for being more****
>>              specific on how much to burst. Most people won't need to****
>>              change from the default 64k. Applies to all mountpoints ****
>> ** **
>>  -->****
>>         <burst-size>4096</burst-size>****
>>      </limits>****
>> ** **
>> Regards****
>> ** **
>>  Jorge****
>> ** **
>> ** **
>> 2013/8/14 Stephen Dame <sdame at 207me.com>****
>>
>> Jorge, ****
>>  ****
>> Play around with the burst size and queue size?  here is my xml config
>> around 3-4 second delay from 16k freeswitch conference. ****
>> To small a buffer and the players disconnect? Im also running icecast on
>> same server.****
>>  ****
>> <icecast>****
>>     <limits>****
>>         <clients>100</clients>****
>>         <sources>10</sources>****
>>         <threadpool>5</threadpool>****
>>         <queue-size>524288</queue-size>****
>>         <client-timeout>30</client-timeout>****
>>         <header-timeout>15</header-timeout>****
>>          <source-timeout>10</source-timeout>****
>>         <burst-on-connect>1</burst-on-connect>****
>>         <!-- same as burst-on-connect, but this allows for being more****
>>               specific on how much to burst. Most people won't need to***
>> *
>>              change from the default 64k. Applies to all mountpoints  -->
>> ****
>>         <burst-size>65535</burst-size>****
>>     </limits>****
>>  ****
>> Regards,****
>> Stephen****
>>  ****
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Ken Rice
>> *Sent:* Wednesday, August 14, 2013 12:35 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Mod_shout using with icecast Big delay*
>> ***
>>  ****
>> You have to keep in mind that icecast itself has a fair bit of buffering
>> involved with it and theres not a lot you can do about that
>>
>>
>> On 8/14/13 11:30 AM, "Jorge N??ez" <jorgefren12 at gmail.com> wrote:****
>>
>> Hi I am using mod_shout to send a conference call to icecast and hear it
>> from a tag audio in html5 in realtime, but it has a big delay like 12
>> seconds. How can I reduce the latency of the audio sent from freeswitch or
>> what can I do to improve this.
>>
>> Regards
>>
>> Jorge****
>> ------------------------------
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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****
>>
>>
>> --
>> Ken
>> *http://www.FreeSWITCH.org <http://www.freeswitch.org/>
>> http://www.ClueCon.com <http://www.cluecon.com/>
>> http://www.OSTAG.org <http://www.ostag.org/>
>> *irc.freenode.net #freeswitch****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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****
>>
>> ** **
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>>  _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveayre at gmail.com  Mon Aug 19 20:45:00 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 19 Aug 2013 17:45:00 +0100
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <94920D4E-5645-4BCE-A6A9-C7780E0BDBC6@imtelecom.fr>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
	<CAFiqYumBFafVazhjXUf1CdXgVaYqvbU4QzrPEc2svCLJu0BaFw@mail.gmail.com>
	<CAHwzD9n+CfCbCYNvwPcSiwYQoQe9bved3uK6q1=okh+2ravZoA@mail.gmail.com>
	<A46F2C5A-8082-4155-A4E0-96CE65AF3431@insensate.co.uk>
	<CAHwzD9ncYxxJxaMYnu2qx78Q_SSLp3onz+5=kN=DSz+vRVQGKA@mail.gmail.com>
	<CAFiqYunDhXi+v1H4P=vQnrGW2WRd61w1=nnGwVPo+Pv9CWG3Dw@mail.gmail.com>
	<CAHwzD9=gpZjsuVxdj3O04PMPCh_mqoxj8SudvXak=ejWZJiohw@mail.gmail.com>
	<94920D4E-5645-4BCE-A6A9-C7780E0BDBC6@imtelecom.fr>
Message-ID: <CAFiqYunihE+DaE71kqX0TtUaSr03nLFT7cXPWHrLgGTVbp=CJg@mail.gmail.com>

AFAIK port zero is a method of putting the call on hold in a reinvite, and
shouldn't be in the initial invite.

Without a port you can only receive audio not send it (as you have nowhere
to send to).

Your SDP shows the phone is hiding its model - what is it? There's a 2011
mailing list  thread suggesting it may be an issue with a bad firmware on a
type of phone.




On Monday, August 19, 2013, Vallimamod ABDULLAH wrote:

> Hi,
>
> It looks like the remote SDP does not contain correct rtp port
> information. You have:
>
> m=audio 0 RTP/AVP 0 18 101
>
> The first '0' should be the rtp port normally. I am not sure but it does
> not looks correct and may be the cause of the incompatible destination
> error.
> Hope this helps.
>
> --
> Best Regards,
> Vallimamod
> .
>
>
>
> On Aug 19, 2013, at 2:59 PM, Royce Mitchell III <royce3 at gmail.com> wrote:
>
> Here is one example, thanks
>
>
>
> Royce Mitchell, IT Consultant
> ITAS Solutions
> royce3 at itas-solutions.com
>
>
> On Fri, Aug 16, 2013 at 4:12 PM, Steven Ayre <steveayre at gmail.com> wrote:
>
> Can you paste a debug log of the entire lifetime of the call?
>
>
> On 16 August 2013 21:57, Royce Mitchell III <royce3 at gmail.com> wrote:
>
> The only thing I'm doing in FS regarding codecs is I force PCMU in certain
> conditions ( ran into a transcoding bug in FreeSwitch between G722 HD and
> G711 with the Polycom phones )
>
>
>
> Royce Mitchell, IT Consultant
> ITAS Solutions
> royce3 at itas-solutions.com
>
>
> On Fri, Aug 16, 2013 at 3:35 PM, Lawrence Conroy <lconroy at insensate.co.uk>wrote:
>
> Hi there,
>  Are you sure about fS allowing PCMU?
> According to the remote SDP, your incall is proposing PCMU as its top
> choice; g729 is 2nd best (quite right too :).
> Forcing the adtran to offer only PCMU should not make ay difference to
> that -- it'll still propose PCMU so no change.
> Either your setup is somehow blocking PCMU on the b-leg (but I'd expect to
> see that on the log), OR is trying to transcode because the b-leg requires
> some (non-PCMU) codec and can't (again, I'd expect that to be logged), OR
> fS is not accepting PCMU.
> Assuming that PCMU is in the fS vars codec lists, does your dialplan do
> anything funky with the codec list for an incall?
> all the best,
>   Lawrence
>
> On 16 Aug 2013, at 21:06, Royce Mitchell III wrote:
> > My FreeSWITCH is configured to prefer PCMU, and the devices it is talking
> > to are Adtran 908e's. The Adtrans are configured for the default codec
> > group which is supposed to be PCMU, but I can reconfigure them to
> > explicitly allow only PCMU. I will try that and see if it makes a
> > difference.
> >
> >
> >
> > Royce Mitchell, IT Consultant
> > ITAS Solutions
> > royce3 at itas-solutions.com
> >
> >
> > On Fri, Aug 16, 2013 at 11:04 AM, Steven Ayre <steveayre at gmail.com>
> wrote:
> >
> >> INCOMPATIBLE_DESTINATION means a codec problem.
> >>
> >> The remote SDP sends they're offerring PCMU and G729.
> >>
> >> What codecs are you allowing, what codecs are you bridging with, and
> since
> >> G729 is on the list are you perhaps trying to transcode without using
> >> mod_com_g729+licenses?
> >>
> >>
> >> On 16 August 2013 16:07, Royce Mitchell III <
>
>
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From sravi123 at yahoo.com  Mon Aug 19 21:42:00 2013
From: sravi123 at yahoo.com (Ravi)
Date: Mon, 19 Aug 2013 23:12:00 +0530
Subject: [Freeswitch-users] Transfer / Forward calls - from an operator
 to an extension after the call is picked
In-Reply-To: <CAHKqMb5SK8V5h-CYf8uaivj_LkvMR-xjQ1k+gG7UA_0z_5g44w@mail.gmail.com>
References: <1376827283.11657.YahooMailNeo@web160501.mail.bf1.yahoo.com>
	<CAHKqMb5SK8V5h-CYf8uaivj_LkvMR-xjQ1k+gG7UA_0z_5g44w@mail.gmail.com>
Message-ID: <521258E8.90900@yahoo.com>

Thank you Nandy for the response. It helped, and I am able to route calls.

I am able to transfer the call, when it comes in, to the operator 
through Freeswitch. After the operator takes the call, the operator 
talks to the caller for a couple of minutes, and then has to transfer 
the call to one of the extensions say 1010. How is that done ? Or as a 
simple case, how can i transfer a call that I have received to another 
extension ?

Perhaps this is very easy, but I am not sure where to find this 
information.

Thanks.
Ravi

On 19/08/13 9:24 AM, Nandy Dagondon wrote:
> Hi,
>
> 1. Copy your working dialplan entry.
> 2. Add <condition field="caller_id_number" data="4302000"/>
> 3. Change the bridge/transfer application to connect to the operator
>
> Take note to place the above entry ahead of your working entry. I hope 
> this helps.
> /Nandy
>
>
> On Sun, Aug 18, 2013 at 8:01 PM, Ravi <sravi123 at yahoo.com 
> <mailto:sravi123 at yahoo.com>> wrote:
>
>     Hello Everyone !
>
>     I have successfully configured my PRI connection. I am able to
>     send and receive calls. Now I am trying to have an operator handle
>     one number - say 4302000. When any call is received the operator
>     will talk to the caller and need to transfer the call to an
>     extension. I tried looking at dialplan documentation, and I think
>     I find only instances where the transfer is done within the
>     program and not when some one picks up the call.
>
>     Can some one please point me to right place/ documents where I can
>     find information regarding this.
>
>     I did find a link with some similar request, but the information
>     is limited.
>     http://lists.freeswitch.org/pipermail/freeswitch-users/2012-October/088579.html
>
>     Thanks for your help.
>     Ravi
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 avi at avimarcus.net  Mon Aug 19 21:55:51 2013
From: avi at avimarcus.net (Avi Marcus)
Date: Mon, 19 Aug 2013 17:55:51 +0000
Subject: [Freeswitch-users] Congratulations and Wanted: Single Consultant
In-Reply-To: <521223ED.4030408@puzzled.xs4all.nl>
References: <521210F7.5040308@digitalmail.com>
	<521223ED.4030408@puzzled.xs4all.nl>
Message-ID: <0000014097b7f9cc-12c1b148-ca94-46af-9ad0-9ec0ebf430f1-000000@email.amazonses.com>

Thanks!

(I hadn't announced to the list because that seemed personal and off topic.
Gifts are of course welcomed!)

-Avi
On Aug 19, 2013 4:59 PM, "Patrick Lists" <freeswitch-list at puzzled.xs4all.nl>
wrote:

> On 08/19/2013 02:35 PM, Alex Lake wrote:
> > A little bird tells me that Avi Marcus is getting married (or
>
> Congratulations Avi!
>
> Regards,
> Patrick
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 royce3 at gmail.com  Mon Aug 19 21:59:35 2013
From: royce3 at gmail.com (Royce Mitchell III)
Date: Mon, 19 Aug 2013 12:59:35 -0500
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <CAFiqYunihE+DaE71kqX0TtUaSr03nLFT7cXPWHrLgGTVbp=CJg@mail.gmail.com>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
	<CAFiqYumBFafVazhjXUf1CdXgVaYqvbU4QzrPEc2svCLJu0BaFw@mail.gmail.com>
	<CAHwzD9n+CfCbCYNvwPcSiwYQoQe9bved3uK6q1=okh+2ravZoA@mail.gmail.com>
	<A46F2C5A-8082-4155-A4E0-96CE65AF3431@insensate.co.uk>
	<CAHwzD9ncYxxJxaMYnu2qx78Q_SSLp3onz+5=kN=DSz+vRVQGKA@mail.gmail.com>
	<CAFiqYunDhXi+v1H4P=vQnrGW2WRd61w1=nnGwVPo+Pv9CWG3Dw@mail.gmail.com>
	<CAHwzD9=gpZjsuVxdj3O04PMPCh_mqoxj8SudvXak=ejWZJiohw@mail.gmail.com>
	<94920D4E-5645-4BCE-A6A9-C7780E0BDBC6@imtelecom.fr>
	<CAFiqYunihE+DaE71kqX0TtUaSr03nLFT7cXPWHrLgGTVbp=CJg@mail.gmail.com>
Message-ID: <CAHwzD9=OuAJ-JDHcZZEM_VdxzGnT1v9En3BvSj36ey96ULwTuA@mail.gmail.com>

The other device is not a phone, but a pair of Adtran 908e 2nd Gen doing
basic PRI to VoIP conversion. All of a sudden all calls were rejecting like
this, but when I restarted FreeSWITCH everything started working fine
again. I did not reboot or otherwise do anything to the Adtrans. The
Adtrans are sending early media and have no reason to receive it, yet, so
that might explain why they are sending the 0.

On another note, I just searched a recent log file, and I don't find the
string "m=audio 0" anywhere in the file.



Royce Mitchell, IT Consultant
ITAS Solutions
royce3 at itas-solutions.com


On Mon, Aug 19, 2013 at 11:45 AM, Steven Ayre <steveayre at gmail.com> wrote:

> AFAIK port zero is a method of putting the call on hold in a reinvite, and
> shouldn't be in the initial invite.
>
> Without a port you can only receive audio not send it (as you have nowhere
> to send to).
>
> Your SDP shows the phone is hiding its model - what is it? There's a 2011
> mailing list  thread suggesting it may be an issue with a bad firmware on a
> type of phone.
>
>
>
>
> On Monday, August 19, 2013, Vallimamod ABDULLAH wrote:
>
>> Hi,
>>
>> It looks like the remote SDP does not contain correct rtp port
>> information. You have:
>>
>> m=audio 0 RTP/AVP 0 18 101
>>
>> The first '0' should be the rtp port normally. I am not sure but it does
>> not looks correct and may be the cause of the incompatible destination
>> error.
>> Hope this helps.
>>
>> --
>> Best Regards,
>> Vallimamod
>> .
>>
>>
>>
>> On Aug 19, 2013, at 2:59 PM, Royce Mitchell III <royce3 at gmail.com> wrote:
>>
>> Here is one example, thanks
>>
>>
>>
>> Royce Mitchell, IT Consultant
>> ITAS Solutions
>> royce3 at itas-solutions.com
>>
>>
>> On Fri, Aug 16, 2013 at 4:12 PM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Can you paste a debug log of the entire lifetime of the call?
>>
>>
>> On 16 August 2013 21:57, Royce Mitchell III <royce3 at gmail.com> wrote:
>>
>> The only thing I'm doing in FS regarding codecs is I force PCMU in
>> certain conditions ( ran into a transcoding bug in FreeSwitch between G722
>> HD and G711 with the Polycom phones )
>>
>>
>>
>> Royce Mitchell, IT Consultant
>> ITAS Solutions
>> royce3 at itas-solutions.com
>>
>>
>> On Fri, Aug 16, 2013 at 3:35 PM, Lawrence Conroy <lconroy at insensate.co.uk
>> > wrote:
>>
>> Hi there,
>>  Are you sure about fS allowing PCMU?
>> According to the remote SDP, your incall is proposing PCMU as its top
>> choice; g729 is 2nd best (quite right too :).
>> Forcing the adtran to offer only PCMU should not make ay difference to
>> that -- it'll still propose PCMU so no change.
>> Either your setup is somehow blocking PCMU on the b-leg (but I'd expect
>> to see that on the log), OR is trying to transcode because the b-leg
>> requires some (non-PCMU) codec and can't (again, I'd expect that to be
>> logged), OR fS is not accepting PCMU.
>> Assuming that PCMU is in the fS vars codec lists, does your dialplan do
>> anything funky with the codec list for an incall?
>> all the best,
>>   Lawrence
>>
>> On 16 Aug 2013, at 21:06, Royce Mitchell III wrote:
>> > My FreeSWITCH is configured to prefer PCMU, and the devices it is
>> talking
>> > to are Adtran 908e's. The Adtrans are configured for the default codec
>> > group which is supposed to be PCMU, but I can reconfigure them to
>> > explicitly allow only PCMU. I will try that and see if it makes a
>> > difference.
>> >
>> >
>> >
>> > Royce Mitchell, IT Consultant
>> > ITAS Solutions
>> > royce3 at itas-solutions.com
>> >
>> >
>> > On Fri, Aug 16, 2013 at 11:04 AM, Steven Ayre <steveayre at gmail.com>
>> wrote:
>> >
>> >> INCOMPATIBLE_DESTINATION means a codec problem.
>> >>
>> >> The remote SDP sends they're offerring PCMU and G729.
>> >>
>> >> What codecs are you allowing, what codecs are you bridging with, and
>> since
>> >> G729 is on the list are you perhaps trying to transcode without using
>> >> mod_com_g729+licenses?
>> >>
>> >>
>> >> On 16 August 2013 16:07, Royce Mitchell III <
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 karl at xtronics.com  Mon Aug 19 22:42:19 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Mon, 19 Aug 2013 13:42:19 -0500
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <CAHt+dNHYN85jK-BmZF-MvsqwWmp=nu07pM6XQM-c-xBXwqUE6A@mail.gmail.com>
References: <CE33869E.78FAE%krice@freeswitch.org>
	<57B0D003-A269-448E-A323-3E8F3B6DD8C0@insensate.co.uk>
	<CAHt+dNH3sFf2So6EszoAZp3E74c5y+cqARG-_eadbPw71e5foQ@mail.gmail.com>
	<CAHt+dNHYN85jK-BmZF-MvsqwWmp=nu07pM6XQM-c-xBXwqUE6A@mail.gmail.com>
Message-ID: <5212670B.6030208@xtronics.com>

Do most of these phones offer a 'transfer' button?

Do the Siemens/Gigaset Desk phones offer BLF that anyone has used?

What about the Cisco SPA50x ?

BLF is THE feature I need for the 2 phones.  ( I haven't seen BLF on any of the softphones I played 
with.)

I also have a page for soft-phones started- heavily Linux centered.
http://wiki.xtronics.com/index.php/Debian_Linux#Telephony_VoIP-_Video_chat

These links are mostly my learning notes - There may be bits people want to grab and stick in 
wiki.freeswitch.org




--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

When a bureaucrat calls for accountability,
it means he has found a way to game the system.
--------------------------------------------------------------------------------


From covici at ccs.covici.com  Mon Aug 19 22:51:41 2013
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 19 Aug 2013 14:51:41 -0400
Subject: [Freeswitch-users] Congratulations and Wanted: Single Consultant
In-Reply-To: <0000014097b7f9cc-12c1b148-ca94-46af-9ad0-9ec0ebf430f1-000000@email.amazonses.com>
References: <521210F7.5040308@digitalmail.com>
	<521223ED.4030408@puzzled.xs4all.nl>
	<0000014097b7f9cc-12c1b148-ca94-46af-9ad0-9ec0ebf430f1-000000@email.amazonses.com>
Message-ID: <14463.1376938301@ccs.covici.com>

Congrats to you!

Avi Marcus <avi at avimarcus.net> wrote:

> Thanks!
> 
> (I hadn't announced to the list because that seemed personal and off topic.
> Gifts are of course welcomed!)
> 
> -Avi
> On Aug 19, 2013 4:59 PM, "Patrick Lists" <freeswitch-list at puzzled.xs4all.nl>
> wrote:
> 
> > On 08/19/2013 02:35 PM, Alex Lake wrote:
> > > A little bird tells me that Avi Marcus is getting married (or
> >
> > Congratulations Avi!
> >
> > Regards,
> > Patrick
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
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 karl at xtronics.com  Mon Aug 19 23:32:19 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Mon, 19 Aug 2013 14:32:19 -0500
Subject: [Freeswitch-users] domain (IP),
	domain_name vs both sides of a fire wall
In-Reply-To: <CAHwzD9=OuAJ-JDHcZZEM_VdxzGnT1v9En3BvSj36ey96ULwTuA@mail.gmail.com>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
	<CAFiqYumBFafVazhjXUf1CdXgVaYqvbU4QzrPEc2svCLJu0BaFw@mail.gmail.com>
	<CAHwzD9n+CfCbCYNvwPcSiwYQoQe9bved3uK6q1=okh+2ravZoA@mail.gmail.com>
	<A46F2C5A-8082-4155-A4E0-96CE65AF3431@insensate.co.uk>
	<CAHwzD9ncYxxJxaMYnu2qx78Q_SSLp3onz+5=kN=DSz+vRVQGKA@mail.gmail.com>
	<CAFiqYunDhXi+v1H4P=vQnrGW2WRd61w1=nnGwVPo+Pv9CWG3Dw@mail.gmail.com>
	<CAHwzD9=gpZjsuVxdj3O04PMPCh_mqoxj8SudvXak=ejWZJiohw@mail.gmail.com>
	<94920D4E-5645-4BCE-A6A9-C7780E0BDBC6@imtelecom.fr>
	<CAFiqYunihE+DaE71kqX0TtUaSr03nLFT7cXPWHrLgGTVbp=CJg@mail.gmail.com>
	<CAHwzD9=OuAJ-JDHcZZEM_VdxzGnT1v9En3BvSj36ey96ULwTuA@mail.gmail.com>
Message-ID: <521272C3.8030505@xtronics.com>

There is one bit of setting this up that I'm having trouble finding good examples - and perhaps the 
weakest part of the book.

I have a real firewall (not a router with some hacked up Linux kernel) that I DNAT the appropriate 
ports to the freeswitch server.

I would think there would be some place in the default configs where one would/could specify four 
things:

Server host name as seen from LAN
Server IP address as seen from LAN

Public FQDN as seen from Internet
Public IP address as seen from Internet

There is something - in the book here there is a brief mention of :
ext-sip-ip and
ext-rtp-ip

and more at
https://wiki.freeswitch.org/wiki/Sofia.conf.xml#ext-rtp-ip

I'm assuming that the parameter sip-ip means internal-sip-ip and rtp-ip means internal-rtp-ip ?

If I directly define these will I mess up NAT transversal on a external phone that is behind NAT?

( At one time I actually understood how a stun router worked - the insight might have only lasted 
for a day, but I can understand this stuff).


These are seperately defined in:
  /etc/freeswitch/sip_profiles/external.xml and
/etc/freeswitch/sip_profiles/internal.xml

Which makes me think I don't understand the context of 'internal' and 'external' as used for sip 
profiles?  I was assuming that external was for calls from the public internet and internal were for 
calls from the local LAN - but I'm not sure anymore.


This combined with the use of <domain name="$${domain}">  at the top of directory/default.xml
-  which IP address should it be?
- Should there be two similar sections - one for the internal IP address and a second for the 
external IP address? What if I go on a trip and want to register my extension from outside?
- just what is domain used for in the directory?


and also the line in directory/default.xml

       <param name="dial-string" 
value="{^^:sip_invite_domain=${dialed_domain}:presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(*/${dialed_user}@${dialed_domain})}"/>
     </params>

I think it defines the sip invitation string? - should it be different for the internal/vs external 
IP addresses?


I have a feeling like there is something obvious that I don't get.


I think it is important to know what is going on here before I start working with the ALCs


--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

It is a miracle that curiosity survives formal education.
-Albert Einstein
--------------------------------------------------------------------------------


From covici at ccs.covici.com  Mon Aug 19 23:49:46 2013
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Mon, 19 Aug 2013 15:49:46 -0400
Subject: [Freeswitch-users] unable to subscrib to freeswitch-sec
Message-ID: <23878.1376941786@ccs.covici.com>

Hi.  I am unable to subscribe to the freeswitch-sec list.  When I get
the confirmation, the address it wants to  send the reply to is
tron.freeswitch.org which seems not to exist -- can't ping or anything.

Thanks.

-- 
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 nandy1925 at gmail.com  Tue Aug 20 00:23:30 2013
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Tue, 20 Aug 2013 04:23:30 +0800
Subject: [Freeswitch-users] Transfer / Forward calls - from an operator
 to an extension after the call is picked
In-Reply-To: <521258E8.90900@yahoo.com>
References: <1376827283.11657.YahooMailNeo@web160501.mail.bf1.yahoo.com>
	<CAHKqMb5SK8V5h-CYf8uaivj_LkvMR-xjQ1k+gG7UA_0z_5g44w@mail.gmail.com>
	<521258E8.90900@yahoo.com>
Message-ID: <CAHKqMb4Xzwswcr3PtbA5OT-AVGSNZ4+BiiW5v4B8e0=g1o1PUQ@mail.gmail.com>

Hi Ravi,

Nice to know it helped. What  you're describing is Supervised Call
Transfer. Locate your phone's HOLD and TRANSFER buttons.
1. Press HOLD to place the caller on-hold. She'll enjoy the music :-)
2. You'll hear a dialtone. Dial extension 1010. Talk to the 3rd party.
3. If she accepts it, press the TRANSFER button then hang-up.
4. If not, press the HOLD button to resume talking to the caller.

For analog phones, the vanilla default.xml dialplan has an example in the
Local extension section what keys are bound to call transfers.

/Nandy


On Tue, Aug 20, 2013 at 1:42 AM, Ravi <sravi123 at yahoo.com> wrote:

>  Thank you Nandy for the response. It helped, and I am able to route
> calls.
>
> I am able to transfer the call, when it comes in, to the operator through
> Freeswitch. After the operator takes the call, the operator talks to the
> caller for a couple of minutes, and then has to transfer the call to one of
> the extensions say 1010. How is that done ? Or as a simple case, how can i
> transfer a call that I have received to another extension ?
>
> Perhaps this is very easy, but I am not sure where to find this
> information.
>
> Thanks.
> Ravi
>
>
> On 19/08/13 9:24 AM, Nandy Dagondon wrote:
>
>  Hi,
>
> 1. Copy your working dialplan entry.
>  2. Add <condition field="caller_id_number" data="4302000"/>
>  3. Change the bridge/transfer application to connect to the operator
>
>  Take note to place the above entry ahead of your working entry. I hope
> this helps.
>  /Nandy
>
>
> On Sun, Aug 18, 2013 at 8:01 PM, Ravi <sravi123 at yahoo.com> wrote:
>
>>  Hello Everyone !
>>
>> I have successfully configured my PRI connection. I am able to send and
>> receive calls. Now I am trying to have an operator handle one number - say
>> 4302000. When any call is received the operator will talk to the caller and
>> need to transfer the call to an extension. I tried looking at dialplan
>> documentation, and I think I find only instances where the transfer is done
>> within the program and not when some one picks up the call.
>>
>> Can some one please point me to right place/ documents where I can find
>> information regarding this.
>>
>> I did find a link with some similar request, but the information is
>> limited.
>>
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2012-October/088579.html
>>
>> Thanks for your help.
>> Ravi
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 soapee01.fs at stubbornroses.com  Tue Aug 20 00:33:45 2013
From: soapee01.fs at stubbornroses.com (soapee01.fs at stubbornroses.com)
Date: Mon, 19 Aug 2013 15:33:45 -0500
Subject: [Freeswitch-users] Snom and valet_parking
In-Reply-To: <52127F44.3060103@stubbornroses.com>
References: <52127F44.3060103@stubbornroses.com>
Message-ID: <52128129.4050004@stubbornroses.com>

Hi list:

I have an issue with a new Snom 821.  I've configured valet_parking for 
auto park mode (eg: my_lot auto in).  After the attended transfer, I 
always receive a Temporarily Unavailable response on the Snom phone from 
FS.  This does not hold true on my yealink/cisco/linksys/etc phones.  I 
can do proper attended transfers from extension to extension on the Snom.

I've tried this on a separate machine with latest stable, and latest git 
head.

Dialplan:
<extension name="valet_auto_park_in" >
<condition field="destination_number" expression="^(7900)$" >
<action application="valet_park" data="my_lot@${domain_name} auto in 
7901 7999" />
</condition>
</extension>

<extension name="valet_auto_park_out" >
<condition field="destination_number" expression="^(79\d\d)$" >
<action application="answer" />
<action application="valet_park" data="my_lot@${domain_name} $1" />
</condition>
</extension>

     screenshot from wireshark: http://tinypic.com/r/jtp5hc/5
     Relevant Log: http://pastebin.freeswitch.org/21322

     Snom Firmware Version: snom821-SIP 8.7.3.19

The call flow analysis on wireshark doesn't include any of the sip 
messages that follow the 480, so it doesn't really provide much there.  
Any pointers would be greatly appreciated.  Duplication (for my own 
personal sanity) would also be appreciated.  Could be I just got a bad 
phone...

Regards,

James
PS: I know the timestamps on the wireshark don't match the log. They are 
from different tests, but the results are always the same.


From schoch+freeswitch.org at xwin32.com  Tue Aug 20 02:11:02 2013
From: schoch+freeswitch.org at xwin32.com (Steven Schoch)
Date: Mon, 19 Aug 2013 15:11:02 -0700
Subject: [Freeswitch-users] Transfer / Forward calls - from an operator
 to an extension after the call is picked
In-Reply-To: <CAHKqMb4Xzwswcr3PtbA5OT-AVGSNZ4+BiiW5v4B8e0=g1o1PUQ@mail.gmail.com>
References: <1376827283.11657.YahooMailNeo@web160501.mail.bf1.yahoo.com>
	<CAHKqMb5SK8V5h-CYf8uaivj_LkvMR-xjQ1k+gG7UA_0z_5g44w@mail.gmail.com>
	<521258E8.90900@yahoo.com>
	<CAHKqMb4Xzwswcr3PtbA5OT-AVGSNZ4+BiiW5v4B8e0=g1o1PUQ@mail.gmail.com>
Message-ID: <CAOkuKdeSKj6bNpwXYM+CFyoUBT8Hup-2n-DjNLFN=OhhT-U9kw@mail.gmail.com>

It seems that different phones behave differently regarding call transfer.
On our Polycom phones, you press Trans, which puts the caller on hold and
prompts for another number to dial. After you call the extension, and talk
to the 3rd party, you press Trans again. You can't transfer a call that has
been put on hold without taking it off hold first.

-- 
Steve


On Mon, Aug 19, 2013 at 1:23 PM, Nandy Dagondon <nandy1925 at gmail.com> wrote:

> Hi Ravi,
>
> Nice to know it helped. What  you're describing is Supervised Call
> Transfer. Locate your phone's HOLD and TRANSFER buttons.
> 1. Press HOLD to place the caller on-hold. She'll enjoy the music :-)
> 2. You'll hear a dialtone. Dial extension 1010. Talk to the 3rd party.
> 3. If she accepts it, press the TRANSFER button then hang-up.
> 4. If not, press the HOLD button to resume talking to the caller.
>
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From dannygershman at cinchcast.com  Tue Aug 20 02:03:17 2013
From: dannygershman at cinchcast.com (Danny Gershman)
Date: Mon, 19 Aug 2013 22:03:17 +0000
Subject: [Freeswitch-users] Freeswitch RTP endpoint linked to another RTP
	endpoint
Message-ID: <CE380E64.BD6B%dannygershman@cinchcast.com>

So I have an INVITE that comes in that doesn?t have SDP in it. I have the ?enable-3pcc? turned on so FreeSWITCH does respond properly. So then it responds with SDP allocating an endpoint. Then I take that SDP information and I turn around and send a new INVITE with the SDP in it. The hopes is that this endpoint is then bridges to a new endpoint. Essentially Freeswitch is connected to itself.

Here is the pastebin - http://pastebin.freeswitch.org/21327

Look at the CreatePlayConnectionForConference:1 in the User-Agent field.  And then look at the CreatePlayConnectionForConference:2.  Those two are supposed be linked together with RTP.

Danny Gershman
Principal Engineer
Cinchcast
475 Park Avenue South, 19th Floor
New York, NY 10016
646.807.0840 (o)
973.931.6239 (c)
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From eidevm5 at gmail.com  Tue Aug 20 04:47:29 2013
From: eidevm5 at gmail.com (Peter)
Date: Tue, 20 Aug 2013 10:47:29 +1000
Subject: [Freeswitch-users] Call not hanging up
In-Reply-To: <210DAC89-E7C0-4417-ACFE-2A77E424C934@jerris.com>
References: <CABkS=xoKBtF6zP4BzWb8tiNzpK2N-yLM6F1Jqhfz26fPcnO34Q@mail.gmail.com>
	<210DAC89-E7C0-4417-ACFE-2A77E424C934@jerris.com>
Message-ID: <CABkS=xpWv2jHm8twr5QV4dHTQYtDtjMGZC4sp1erqDWDhAJAnA@mail.gmail.com>

It appears the BYE is getting through on both ends (although I'm not 100%
familiar with the full SIP handshake).

Here's the SIP trace for the caller (ie: the end that isn't hanging up)

http://pastebin.freeswitch.org/21328

and here's the SIP trace for the callee (ie:  the end the call is hung up)

http://pastebin.freeswitch.org/21329



I'm not sure if having TLS and inbound-bypass-media set on FS1 are having
any affect on this problem or not.


On Tue, Aug 20, 2013 at 1:51 AM, Michael Jerris <mike at jerris.com> wrote:

> Check out sip trace.. sounds like a BYE is not getting somewhere.
>
> On Aug 19, 2013, at 2:02 AM, Peter <eidevm5 at gmail.com> wrote:
>
> Running FS 1.2.12 in the following environment:
>
> FS Ext  <--->  FS SBC <---> FS Int
>
> I can successfully make calls in both directions.
>
> If the caller hangs up, the other end correctly hangs up.
>
> However, if the callee hangs up, the caller session remains active until
> the RTP timeout kicks in.
>
> Here's a snippet from the FS log where this occurs (1004 is the caller and
> 2010 is the callee)
>
> [NOTICE] sofia.c:716 Hangup sofia/internal/2010 at 10.1.1.206:5061[CS_HIBERNATE] [NORMAL_CLEARING]
> [NOTICE] switch_ivr_bridge.c:1109 Hangup sofia/internal/1004 at 10.1.1.204[CS_HIBERNATE] [NORMAL_CLEARING]
> [NOTICE] switch_core_session.c:1560 Session 4 (sofia/internal/
> 2010 at 10.1.1.206:5061) Ended
> [NOTICE] switch_core_session.c:1564 Close Channel sofia/internal/
> 2010 at 10.1.1.206:5061 [CS_DESTROY]
> [NOTICE] switch_core_session.c:1560 Session 3 (sofia/internal/
> 1004 at 10.1.1.104) Ended
> [NOTICE] switch_core_session.c:1564 Close Channel sofia/internal/
> 1004 at 10.1.1.104 [CS_DESTROY]
>
> I've tried setting hangup_after_bridge=true in the dialplan on FS Ext.
> I wasn't 100% sure whether it should go on FS Ext or FS SBC, so I tried
> both and it made no difference.
>
> What else should I be checking?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Tue Aug 20 04:56:33 2013
From: eidevm5 at gmail.com (Peter)
Date: Tue, 20 Aug 2013 10:56:33 +1000
Subject: [Freeswitch-users] Call not hanging up
In-Reply-To: <CABkS=xpWv2jHm8twr5QV4dHTQYtDtjMGZC4sp1erqDWDhAJAnA@mail.gmail.com>
References: <CABkS=xoKBtF6zP4BzWb8tiNzpK2N-yLM6F1Jqhfz26fPcnO34Q@mail.gmail.com>
	<210DAC89-E7C0-4417-ACFE-2A77E424C934@jerris.com>
	<CABkS=xpWv2jHm8twr5QV4dHTQYtDtjMGZC4sp1erqDWDhAJAnA@mail.gmail.com>
Message-ID: <CABkS=xqSjCKmRUKBxGcBC7Qhr4qA=k_=uYtDTp3z_k4Ya9ouTQ@mail.gmail.com>

Also just noticed that I'm getting the message:

nua_server.c:155 nua_stack_process_request() nua(0x7fe54c054880): strange
ACK from <sip:1004 at 10.1.1.204>

Not sure what that means.


On Tue, Aug 20, 2013 at 10:47 AM, Peter <eidevm5 at gmail.com> wrote:

> It appears the BYE is getting through on both ends (although I'm not 100%
> familiar with the full SIP handshake).
>
> Here's the SIP trace for the caller (ie: the end that isn't hanging up)
>
> http://pastebin.freeswitch.org/21328
>
> and here's the SIP trace for the callee (ie:  the end the call is hung up)
>
> http://pastebin.freeswitch.org/21329
>
>
>
> I'm not sure if having TLS and inbound-bypass-media set on FS1 are having
> any affect on this problem or not.
>
>
> On Tue, Aug 20, 2013 at 1:51 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> Check out sip trace.. sounds like a BYE is not getting somewhere.
>>
>> On Aug 19, 2013, at 2:02 AM, Peter <eidevm5 at gmail.com> wrote:
>>
>> Running FS 1.2.12 in the following environment:
>>
>> FS Ext  <--->  FS SBC <---> FS Int
>>
>> I can successfully make calls in both directions.
>>
>> If the caller hangs up, the other end correctly hangs up.
>>
>> However, if the callee hangs up, the caller session remains active until
>> the RTP timeout kicks in.
>>
>> Here's a snippet from the FS log where this occurs (1004 is the caller
>> and 2010 is the callee)
>>
>> [NOTICE] sofia.c:716 Hangup sofia/internal/2010 at 10.1.1.206:5061[CS_HIBERNATE] [NORMAL_CLEARING]
>> [NOTICE] switch_ivr_bridge.c:1109 Hangup sofia/internal/1004 at 10.1.1.204[CS_HIBERNATE] [NORMAL_CLEARING]
>> [NOTICE] switch_core_session.c:1560 Session 4 (sofia/internal/
>> 2010 at 10.1.1.206:5061) Ended
>> [NOTICE] switch_core_session.c:1564 Close Channel sofia/internal/
>> 2010 at 10.1.1.206:5061 [CS_DESTROY]
>> [NOTICE] switch_core_session.c:1560 Session 3 (sofia/internal/
>> 1004 at 10.1.1.104) Ended
>> [NOTICE] switch_core_session.c:1564 Close Channel sofia/internal/
>> 1004 at 10.1.1.104 [CS_DESTROY]
>>
>> I've tried setting hangup_after_bridge=true in the dialplan on FS Ext.
>> I wasn't 100% sure whether it should go on FS Ext or FS SBC, so I tried
>> both and it made no difference.
>>
>> What else should I be checking?
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 steveayre at gmail.com  Tue Aug 20 07:49:48 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 20 Aug 2013 04:49:48 +0100
Subject: [Freeswitch-users] unable to subscrib to freeswitch-sec
In-Reply-To: <23878.1376941786@ccs.covici.com>
References: <23878.1376941786@ccs.covici.com>
Message-ID: <CAFiqYunUWJ3L8gPWhgathroaxXPw8o-NAJLw5NZY73Ey=GyFXA@mail.gmail.com>

I see 100% loss too, I'm sure one of the admins will look into it...

If you read the email though you'll see it also has a link where you can
also confirm your subscription. That should work.



On Monday, August 19, 2013, wrote:

> Hi.  I am unable to subscribe to the freeswitch-sec list.  When I get
> the confirmation, the address it wants to  send the reply to is
> tron.freeswitch.org which seems not to exist -- can't ping or anything.
>
> Thanks.
>
> --
> 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 <javascript:;>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <javascript:;>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> http://lists.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  Tue Aug 20 08:13:36 2013
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 19 Aug 2013 23:13:36 -0500
Subject: [Freeswitch-users] unable to subscrib to freeswitch-sec
In-Reply-To: <CAFiqYunUWJ3L8gPWhgathroaxXPw8o-NAJLw5NZY73Ey=GyFXA@mail.gmail.com>
Message-ID: <CE385720.7910D%krice@freeswitch.org>

I know the problem here... Someone forget to change a setting on it...

Just change the reply to from tron to lists.freeswitch.org... I?ve fixed it
on the list server also


On 8/19/13 10:49 PM, "Steven Ayre" <steveayre at gmail.com> wrote:

> I see 100% loss too, I'm sure one of the admins will look into it...
> 
> If you read the email though you'll see it also has a link?where you can also
> confirm your subscription. That should work.
> 
> 
> 
> On Monday, August 19, 2013,   wrote:
>> Hi. ?I am unable to subscribe to the freeswitch-sec list. ?When I get
>> the confirmation, the address it wants to ?send the reply to is
>> tron.freeswitch.org <http://tron.freeswitch.org>  which seems not to exist --
>> can't ping or anything.
>> 
>> Thanks.
>> 
>> --
>> 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 <javascript:;>
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <javascript:;>
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.com
>> 
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org <javascript:;>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From covici at ccs.covici.com  Tue Aug 20 08:51:24 2013
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 20 Aug 2013 00:51:24 -0400
Subject: [Freeswitch-users] unable to subscrib to freeswitch-sec
In-Reply-To: <CAFiqYunUWJ3L8gPWhgathroaxXPw8o-NAJLw5NZY73Ey=GyFXA@mail.gmail.com>
References: <23878.1376941786@ccs.covici.com>
	<CAFiqYunUWJ3L8gPWhgathroaxXPw8o-NAJLw5NZY73Ey=GyFXA@mail.gmail.com>
Message-ID: <18190.1376974284@ccs.covici.com>

What I did was change the address where it went to lists.freeswitch.org
and that worked, but even the welcome email says tron, so someone should
fix.

Steven Ayre <steveayre at gmail.com> wrote:

> I see 100% loss too, I'm sure one of the admins will look into it...
> 
> If you read the email though you'll see it also has a link where you can
> also confirm your subscription. That should work.
> 
> 
> 
> On Monday, August 19, 2013, wrote:
> 
> > Hi.  I am unable to subscribe to the freeswitch-sec list.  When I get
> > the confirmation, the address it wants to  send the reply to is
> > tron.freeswitch.org which seems not to exist -- can't ping or anything.
> >
> > Thanks.
> >
> > --
> > 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 <javascript:;>
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org <javascript:;>
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org <javascript:;>
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
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 eidevm5 at gmail.com  Tue Aug 20 10:11:43 2013
From: eidevm5 at gmail.com (Peter)
Date: Tue, 20 Aug 2013 16:11:43 +1000
Subject: [Freeswitch-users] Call not hanging up
In-Reply-To: <CABkS=xqSjCKmRUKBxGcBC7Qhr4qA=k_=uYtDTp3z_k4Ya9ouTQ@mail.gmail.com>
References: <CABkS=xoKBtF6zP4BzWb8tiNzpK2N-yLM6F1Jqhfz26fPcnO34Q@mail.gmail.com>
	<210DAC89-E7C0-4417-ACFE-2A77E424C934@jerris.com>
	<CABkS=xpWv2jHm8twr5QV4dHTQYtDtjMGZC4sp1erqDWDhAJAnA@mail.gmail.com>
	<CABkS=xqSjCKmRUKBxGcBC7Qhr4qA=k_=uYtDTp3z_k4Ya9ouTQ@mail.gmail.com>
Message-ID: <CABkS=xroda_i9tZLJh1KwSTVRRNfbm9cFcZ8Q4R+GvurrqX0KQ@mail.gmail.com>

Turns out that it looks like the version of Linphone I was using was not
handling the hangup correctly.

I compiled the latest Linphone version on another phone and the signalling
works fine.


On Tue, Aug 20, 2013 at 10:56 AM, Peter <eidevm5 at gmail.com> wrote:

> Also just noticed that I'm getting the message:
>
> nua_server.c:155 nua_stack_process_request() nua(0x7fe54c054880): strange
> ACK from <sip:1004 at 10.1.1.204>
>
> Not sure what that means.
>
>
> On Tue, Aug 20, 2013 at 10:47 AM, Peter <eidevm5 at gmail.com> wrote:
>
>> It appears the BYE is getting through on both ends (although I'm not 100%
>> familiar with the full SIP handshake).
>>
>> Here's the SIP trace for the caller (ie: the end that isn't hanging up)
>>
>> http://pastebin.freeswitch.org/21328
>>
>> and here's the SIP trace for the callee (ie:  the end the call is hung up)
>>
>> http://pastebin.freeswitch.org/21329
>>
>>
>>
>> I'm not sure if having TLS and inbound-bypass-media set on FS1 are having
>> any affect on this problem or not.
>>
>>
>> On Tue, Aug 20, 2013 at 1:51 AM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> Check out sip trace.. sounds like a BYE is not getting somewhere.
>>>
>>> On Aug 19, 2013, at 2:02 AM, Peter <eidevm5 at gmail.com> wrote:
>>>
>>> Running FS 1.2.12 in the following environment:
>>>
>>> FS Ext  <--->  FS SBC <---> FS Int
>>>
>>> I can successfully make calls in both directions.
>>>
>>> If the caller hangs up, the other end correctly hangs up.
>>>
>>> However, if the callee hangs up, the caller session remains active until
>>> the RTP timeout kicks in.
>>>
>>> Here's a snippet from the FS log where this occurs (1004 is the caller
>>> and 2010 is the callee)
>>>
>>> [NOTICE] sofia.c:716 Hangup sofia/internal/2010 at 10.1.1.206:5061[CS_HIBERNATE] [NORMAL_CLEARING]
>>> [NOTICE] switch_ivr_bridge.c:1109 Hangup sofia/internal/1004 at 10.1.1.204[CS_HIBERNATE] [NORMAL_CLEARING]
>>> [NOTICE] switch_core_session.c:1560 Session 4 (sofia/internal/
>>> 2010 at 10.1.1.206:5061) Ended
>>> [NOTICE] switch_core_session.c:1564 Close Channel sofia/internal/
>>> 2010 at 10.1.1.206:5061 [CS_DESTROY]
>>> [NOTICE] switch_core_session.c:1560 Session 3 (sofia/internal/
>>> 1004 at 10.1.1.104) Ended
>>> [NOTICE] switch_core_session.c:1564 Close Channel sofia/internal/
>>> 1004 at 10.1.1.104 [CS_DESTROY]
>>>
>>> I've tried setting hangup_after_bridge=true in the dialplan on FS Ext.
>>>    I wasn't 100% sure whether it should go on FS Ext or FS SBC, so I tried
>>> both and it made no difference.
>>>
>>> What else should I be checking?
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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 at cassidywebservices.co.uk  Tue Aug 20 12:29:04 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Tue, 20 Aug 2013 09:29:04 +0100
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <5212670B.6030208@xtronics.com>
References: <CE33869E.78FAE%krice@freeswitch.org>
	<57B0D003-A269-448E-A323-3E8F3B6DD8C0@insensate.co.uk>
	<CAHt+dNH3sFf2So6EszoAZp3E74c5y+cqARG-_eadbPw71e5foQ@mail.gmail.com>
	<CAHt+dNHYN85jK-BmZF-MvsqwWmp=nu07pM6XQM-c-xBXwqUE6A@mail.gmail.com>
	<5212670B.6030208@xtronics.com>
Message-ID: <CAMbW9oJ7Kxw8JgQw=H8jxod+eCPo-1-w4Mu37+WxWvUQCD=w5Q@mail.gmail.com>

SPA50x has both blind and attended transfer buttons and BLF can be
configured. Set server type on the Attendant Console tab to the RFC one,
and set up the line keys. The line keys are multipurpose and programmable
on these phones and can either be used to select sip accounts, as quick
dial/BLF/call-pickup, or to launch custom XML applications.

For the line key button settings you do:

func=sd+blf;sub=<user>@$PROXY

You can add in call pickup, too:

func=sd+blf+cp;sub=<user>@$PROXY

which when the monitored extension is ringing, pressing the button will
dial **<user> (the user prepended with 2 asterisks)


On 19 August 2013 19:42, Karl Schmidt <karl at xtronics.com> wrote:

> Do most of these phones offer a 'transfer' button?
>
> Do the Siemens/Gigaset Desk phones offer BLF that anyone has used?
>
> What about the Cisco SPA50x ?
>
> BLF is THE feature I need for the 2 phones.  ( I haven't seen BLF on any
> of the softphones I played
> with.)
>
> I also have a page for soft-phones started- heavily Linux centered.
> http://wiki.xtronics.com/index.php/Debian_Linux#Telephony_VoIP-_Video_chat
>
> These links are mostly my learning notes - There may be bits people want
> to grab and stick in
> wiki.freeswitch.org
>
>
>
>
>
> --------------------------------------------------------------------------------
> Karl Schmidt                                  EMail Karl at xtronics.com
> Transtronics, Inc.                              WEB
> http://secure.transtronics.com
> 3209 West 9th Street                             Ph (785) 841-3089
> Lawrence, KS 66049                              FAX (785) 841-0434
>
> When a bureaucrat calls for accountability,
> it means he has found a way to game the system.
>
> --------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From andrew at cassidywebservices.co.uk  Tue Aug 20 12:30:26 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Tue, 20 Aug 2013 09:30:26 +0100
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <CAMbW9oJ7Kxw8JgQw=H8jxod+eCPo-1-w4Mu37+WxWvUQCD=w5Q@mail.gmail.com>
References: <CE33869E.78FAE%krice@freeswitch.org>
	<57B0D003-A269-448E-A323-3E8F3B6DD8C0@insensate.co.uk>
	<CAHt+dNH3sFf2So6EszoAZp3E74c5y+cqARG-_eadbPw71e5foQ@mail.gmail.com>
	<CAHt+dNHYN85jK-BmZF-MvsqwWmp=nu07pM6XQM-c-xBXwqUE6A@mail.gmail.com>
	<5212670B.6030208@xtronics.com>
	<CAMbW9oJ7Kxw8JgQw=H8jxod+eCPo-1-w4Mu37+WxWvUQCD=w5Q@mail.gmail.com>
Message-ID: <CAMbW9oLvCLHzd_0HpZQVMmjCiQtM0woFn31Keyj0kZzh6E8-Qw@mail.gmail.com>

And yes, most phones have transfer buttons. (Sorry for double post)


On 20 August 2013 09:29, Andrew Cassidy <andrew at cassidywebservices.co.uk>wrote:

> SPA50x has both blind and attended transfer buttons and BLF can be
> configured. Set server type on the Attendant Console tab to the RFC one,
> and set up the line keys. The line keys are multipurpose and programmable
> on these phones and can either be used to select sip accounts, as quick
> dial/BLF/call-pickup, or to launch custom XML applications.
>
> For the line key button settings you do:
>
> func=sd+blf;sub=<user>@$PROXY
>
> You can add in call pickup, too:
>
> func=sd+blf+cp;sub=<user>@$PROXY
>
> which when the monitored extension is ringing, pressing the button will
> dial **<user> (the user prepended with 2 asterisks)
>
>
> On 19 August 2013 19:42, Karl Schmidt <karl at xtronics.com> wrote:
>
>> Do most of these phones offer a 'transfer' button?
>>
>> Do the Siemens/Gigaset Desk phones offer BLF that anyone has used?
>>
>> What about the Cisco SPA50x ?
>>
>> BLF is THE feature I need for the 2 phones.  ( I haven't seen BLF on any
>> of the softphones I played
>> with.)
>>
>> I also have a page for soft-phones started- heavily Linux centered.
>> http://wiki.xtronics.com/index.php/Debian_Linux#Telephony_VoIP-_Video_chat
>>
>> These links are mostly my learning notes - There may be bits people want
>> to grab and stick in
>> wiki.freeswitch.org
>>
>>
>>
>>
>>
>> --------------------------------------------------------------------------------
>> Karl Schmidt                                  EMail Karl at xtronics.com
>> Transtronics, Inc.                              WEB
>> http://secure.transtronics.com
>> 3209 West 9th Street                             Ph (785) 841-3089
>> Lawrence, KS 66049                              FAX (785) 841-0434
>>
>> When a bureaucrat calls for accountability,
>> it means he has found a way to game the system.
>>
>> --------------------------------------------------------------------------------
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From lists at kavun.ch  Tue Aug 20 13:24:46 2013
From: lists at kavun.ch (Emrah)
Date: Tue, 20 Aug 2013 11:24:46 +0200
Subject: [Freeswitch-users] Inexplicable and silent FreeSWITCH crash
Message-ID: <93D5FAB0-93F7-482B-BDDD-4E365193752A@kavun.ch>

Guys,
Very recently my FS instance has been crashing continuously and there is nothing in the logs to help? 
I am on log 9 and there is absolutely nothing out of the ordinary. FS just dies on me.
This has started happening ever since I started playing with video and proxying media. 
How do I get more verbose and get to know what crashes FS?

I tried both the Master branch and the V1.4.Beta.

Thanks for any help I'm going nuts. 

Emrah

From lists at kavun.ch  Tue Aug 20 13:34:20 2013
From: lists at kavun.ch (Emrah)
Date: Tue, 20 Aug 2013 11:34:20 +0200
Subject: [Freeswitch-users] Inexplicable and silent FreeSWITCH crash
In-Reply-To: <93D5FAB0-93F7-482B-BDDD-4E365193752A@kavun.ch>
References: <93D5FAB0-93F7-482B-BDDD-4E365193752A@kavun.ch>
Message-ID: <E449B194-6F41-40EC-8CBB-AB30630DAD58@kavun.ch>

OK I found this and will work on debugging.
https://wiki.freeswitch.org/wiki/Debugging_Freeswitch
On Aug 20, 2013, at 11:24 AM, Emrah <lists at kavun.ch> wrote:

> Guys,
> Very recently my FS instance has been crashing continuously and there is nothing in the logs to help? 
> I am on log 9 and there is absolutely nothing out of the ordinary. FS just dies on me.
> This has started happening ever since I started playing with video and proxying media. 
> How do I get more verbose and get to know what crashes FS?
> 
> I tried both the Master branch and the V1.4.Beta.
> 
> Thanks for any help I'm going nuts. 
> 
> Emrah



From bigx333 at gmail.com  Tue Aug 20 13:36:48 2013
From: bigx333 at gmail.com (Nelson Luiz Ferraz de Camargo Penteado)
Date: Tue, 20 Aug 2013 11:36:48 +0200
Subject: [Freeswitch-users] Inexplicable and silent FreeSWITCH crash
In-Reply-To: <93D5FAB0-93F7-482B-BDDD-4E365193752A@kavun.ch>
References: <93D5FAB0-93F7-482B-BDDD-4E365193752A@kavun.ch>
Message-ID: <CAPt4n6_fyRZLOCc=beCiYa5p6BKDT=R0jo56uWpAYGmJBOakOw@mail.gmail.com>

What's the last line in the logs when it crashes...


On Tue, Aug 20, 2013 at 11:24 AM, Emrah <lists at kavun.ch> wrote:

> Guys,
> Very recently my FS instance has been crashing continuously and there is
> nothing in the logs to help?
> I am on log 9 and there is absolutely nothing out of the ordinary. FS just
> dies on me.
> This has started happening ever since I started playing with video and
> proxying media.
> How do I get more verbose and get to know what crashes FS?
>
> I tried both the Master branch and the V1.4.Beta.
>
> Thanks for any help I'm going nuts.
>
> Emrah
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ssinyagin at yahoo.com  Tue Aug 20 15:35:37 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Tue, 20 Aug 2013 04:35:37 -0700 (PDT)
Subject: [Freeswitch-users] Inexplicable and silent FreeSWITCH crash
In-Reply-To: <93D5FAB0-93F7-482B-BDDD-4E365193752A@kavun.ch>
References: <93D5FAB0-93F7-482B-BDDD-4E365193752A@kavun.ch>
Message-ID: <1376998537.78726.YahooMailNeo@web126203.mail.ne1.yahoo.com>

it's probably too late, but it really helps to keep the whole FreeSWITCH configuration in a Git repository. Then it's easy to track what has changed and what caused the crash.







>________________________________
> From: Emrah <lists at kavun.ch>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Tuesday, August 20, 2013 11:24 AM
>Subject: [Freeswitch-users] Inexplicable and silent FreeSWITCH crash
> 
>
>Guys,
>Very recently my FS instance has been crashing continuously and there is nothing in the logs to help? 
>I am on log 9 and there is absolutely nothing out of the ordinary. FS just dies on me.
>This has started happening ever since I started playing with video and proxying media. 
>How do I get more verbose and get to know what crashes FS?
>
>I tried both the Master branch and the V1.4.Beta.
>
>Thanks for any help I'm going nuts. 
>
>Emrah
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 emamirazavi at gmail.com  Tue Aug 20 16:29:04 2013
From: emamirazavi at gmail.com (Sayyed Mohammad Emami Razavi)
Date: Tue, 20 Aug 2013 16:59:04 +0430
Subject: [Freeswitch-users] Play sound over local_stream (Mix sounds)
Message-ID: <CAJEstwPx5EQtt=2Xt1ZV9jcsFr2f_V5LMrNNT6O7xe2_NbvE5Q@mail.gmail.com>

I want play sound over local_stream without stopping it!
Now i use uuid_broadcast api with aleg argument but it stops MOH
any solution?!
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From GB at cm.nl  Tue Aug 20 16:53:11 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Tue, 20 Aug 2013 14:53:11 +0200
Subject: [Freeswitch-users] List of $$ variables
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD1995B@clubvirtual40.ClubMessage.local>

Hello,

Is there a list available for variables starting with $$? Like $${local_ip_v4}?

Does the following xml element support multiple values?
<X-PRE-PROCESS cmd="set" data="bind_server_ip=auto "/>

Something like:
<X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.1.1,10.0.0.1 "/> #Notice the comma separating the bind IP's.

Thanks,

Grant


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From mike at jerris.com  Tue Aug 20 17:39:26 2013
From: mike at jerris.com (Michael Jerris)
Date: Tue, 20 Aug 2013 09:39:26 -0400
Subject: [Freeswitch-users] Freeswitch RTP endpoint linked to another
	RTP endpoint
In-Reply-To: <CE380E64.BD6B%dannygershman@cinchcast.com>
References: <CE380E64.BD6B%dannygershman@cinchcast.com>
Message-ID: <C84C1EC5-3D87-4ADA-A6E5-92A82DF94D17@jerris.com>

The ports assigned in that sdp are for the session that sends it only? you can't just re-use them on a different invite.  Can you try to explain a bit more what you are trying to accomplish?  

Mike

On Aug 19, 2013, at 6:03 PM, Danny Gershman <dannygershman at cinchcast.com> wrote:

> So I have an INVITE that comes in that doesn?t have SDP in it. I have the ?enable-3pcc? turned on so FreeSWITCH does respond properly. So then it responds with SDP allocating an endpoint. Then I take that SDP information and I turn around and send a new INVITE with the SDP in it. The hopes is that this endpoint is then bridges to a new endpoint. Essentially Freeswitch is connected to itself.
> 
> Here is the pastebin - http://pastebin.freeswitch.org/21327
> 
> Look at the CreatePlayConnectionForConference:1 in the User-Agent field.  And then look at the CreatePlayConnectionForConference:2.  Those two are supposed be linked together with RTP.
> 
> Danny Gershman

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From lists at kavun.ch  Tue Aug 20 18:15:44 2013
From: lists at kavun.ch (Emrah)
Date: Tue, 20 Aug 2013 16:15:44 +0200
Subject: [Freeswitch-users] Inexplicable and silent FreeSWITCH crash
In-Reply-To: <CAPt4n6_fyRZLOCc=beCiYa5p6BKDT=R0jo56uWpAYGmJBOakOw@mail.gmail.com>
References: <93D5FAB0-93F7-482B-BDDD-4E365193752A@kavun.ch>
	<CAPt4n6_fyRZLOCc=beCiYa5p6BKDT=R0jo56uWpAYGmJBOakOw@mail.gmail.com>
Message-ID: <72324CEB-84FC-4DF1-BF66-393FA35CB7AF@kavun.ch>

This time I dumped the core and ran FS in the foreground.
Here are the 2 last lines.
freeswitch: src/switch_channel.c:1494: switch_channel_test_flag: Assertion `channel != ((void *)0)' failed.
Aborted (core dumped)

Thanks for any suggestion to correct this.

Emrah
On Aug 20, 2013, at 11:36 AM, Nelson Luiz Ferraz de Camargo Penteado <bigx333 at gmail.com> wrote:

> What's the last line in the logs when it crashes...
> 
> 
> On Tue, Aug 20, 2013 at 11:24 AM, Emrah <lists at kavun.ch> wrote:
> Guys,
> Very recently my FS instance has been crashing continuously and there is nothing in the logs to help?
> I am on log 9 and there is absolutely nothing out of the ordinary. FS just dies on me.
> This has started happening ever since I started playing with video and proxying media.
> How do I get more verbose and get to know what crashes FS?
> 
> I tried both the Master branch and the V1.4.Beta.
> 
> Thanks for any help I'm going nuts.
> 
> Emrah
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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.brabants at gmail.com  Tue Aug 20 18:32:11 2013
From: michel.brabants at gmail.com (Michel Brabants)
Date: Tue, 20 Aug 2013 16:32:11 +0200
Subject: [Freeswitch-users] SIP phones -
In-Reply-To: <5212670B.6030208@xtronics.com>
References: <CE33869E.78FAE%krice@freeswitch.org>
	<57B0D003-A269-448E-A323-3E8F3B6DD8C0@insensate.co.uk>
	<CAHt+dNH3sFf2So6EszoAZp3E74c5y+cqARG-_eadbPw71e5foQ@mail.gmail.com>
	<CAHt+dNHYN85jK-BmZF-MvsqwWmp=nu07pM6XQM-c-xBXwqUE6A@mail.gmail.com>
	<5212670B.6030208@xtronics.com>
Message-ID: <CAHt+dNENjTjWAe_jL2hMjD3sXZE6bYfLjNPmUx32zMtQE0rKvA@mail.gmail.com>

for BLF on softphones, check the Counterpath Bria-softphones.


On Mon, Aug 19, 2013 at 8:42 PM, Karl Schmidt <karl at xtronics.com> wrote:

> Do most of these phones offer a 'transfer' button?
>
> Do the Siemens/Gigaset Desk phones offer BLF that anyone has used?
>
> What about the Cisco SPA50x ?
>
> BLF is THE feature I need for the 2 phones.  ( I haven't seen BLF on any
> of the softphones I played
> with.)
>
> I also have a page for soft-phones started- heavily Linux centered.
> http://wiki.xtronics.com/index.php/Debian_Linux#Telephony_VoIP-_Video_chat
>
> These links are mostly my learning notes - There may be bits people want
> to grab and stick in
> wiki.freeswitch.org
>
>
>
>
>
> --------------------------------------------------------------------------------
> Karl Schmidt                                  EMail Karl at xtronics.com
> Transtronics, Inc.                              WEB
> http://secure.transtronics.com
> 3209 West 9th Street                             Ph (785) 841-3089
> Lawrence, KS 66049                              FAX (785) 841-0434
>
> When a bureaucrat calls for accountability,
> it means he has found a way to game the system.
>
> --------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Tue Aug 20 18:32:22 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 20 Aug 2013 09:32:22 -0500
Subject: [Freeswitch-users] Inexplicable and silent FreeSWITCH crash
In-Reply-To: <72324CEB-84FC-4DF1-BF66-393FA35CB7AF@kavun.ch>
References: <93D5FAB0-93F7-482B-BDDD-4E365193752A@kavun.ch>
	<CAPt4n6_fyRZLOCc=beCiYa5p6BKDT=R0jo56uWpAYGmJBOakOw@mail.gmail.com>
	<72324CEB-84FC-4DF1-BF66-393FA35CB7AF@kavun.ch>
Message-ID: <CAKbxfG9XE0u-xEP+7+8-_ZYNwJoiTYhLypHuBPvN4v4WPH8UTQ@mail.gmail.com>

You should always start by doing a fresh clean build.
If you still get a core you should get a backtrace and file a jira.



On Tue, Aug 20, 2013 at 9:15 AM, Emrah <lists at kavun.ch> wrote:

> This time I dumped the core and ran FS in the foreground.
> Here are the 2 last lines.
> freeswitch: src/switch_channel.c:1494: switch_channel_test_flag: Assertion
> `channel != ((void *)0)' failed.
> Aborted (core dumped)
>
> Thanks for any suggestion to correct this.
>
> Emrah
> On Aug 20, 2013, at 11:36 AM, Nelson Luiz Ferraz de Camargo Penteado <
> bigx333 at gmail.com> wrote:
>
> What's the last line in the logs when it crashes...
>
>
> On Tue, Aug 20, 2013 at 11:24 AM, Emrah <lists at kavun.ch> wrote:
>
>> Guys,
>> Very recently my FS instance has been crashing continuously and there is
>> nothing in the logs to help?
>> I am on log 9 and there is absolutely nothing out of the ordinary. FS
>> just dies on me.
>> This has started happening ever since I started playing with video and
>> proxying media.
>> How do I get more verbose and get to know what crashes FS?
>>
>> I tried both the Master branch and the V1.4.Beta.
>>
>> Thanks for any help I'm going nuts.
>>
>> Emrah
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From lists at kavun.ch  Tue Aug 20 18:41:51 2013
From: lists at kavun.ch (Emrah)
Date: Tue, 20 Aug 2013 16:41:51 +0200
Subject: [Freeswitch-users] Inexplicable and silent FreeSWITCH crash
In-Reply-To: <CAKbxfG9XE0u-xEP+7+8-_ZYNwJoiTYhLypHuBPvN4v4WPH8UTQ@mail.gmail.com>
References: <93D5FAB0-93F7-482B-BDDD-4E365193752A@kavun.ch>
	<CAPt4n6_fyRZLOCc=beCiYa5p6BKDT=R0jo56uWpAYGmJBOakOw@mail.gmail.com>
	<72324CEB-84FC-4DF1-BF66-393FA35CB7AF@kavun.ch>
	<CAKbxfG9XE0u-xEP+7+8-_ZYNwJoiTYhLypHuBPvN4v4WPH8UTQ@mail.gmail.com>
Message-ID: <EED1E20A-F83A-4D66-8B43-4E68E6598062@kavun.ch>

Yes that's where I was headed now. I am doing a fresh build on a different server and will let you know.
On Aug 20, 2013, at 4:32 PM, Anthony Minessale <anthony.minessale at gmail.com> wrote:

> You should always start by doing a fresh clean build.
> If you still get a core you should get a backtrace and file a jira.
>  
> 
> 
> On Tue, Aug 20, 2013 at 9:15 AM, Emrah <lists at kavun.ch> wrote:
> This time I dumped the core and ran FS in the foreground.
> Here are the 2 last lines.
> freeswitch: src/switch_channel.c:1494: switch_channel_test_flag: Assertion `channel != ((void *)0)' failed.
> Aborted (core dumped)
> 
> Thanks for any suggestion to correct this.
> 
> Emrah
> On Aug 20, 2013, at 11:36 AM, Nelson Luiz Ferraz de Camargo Penteado <bigx333 at gmail.com> wrote:
> 
>> What's the last line in the logs when it crashes...
>> 
>> 
>> On Tue, Aug 20, 2013 at 11:24 AM, Emrah <lists at kavun.ch> wrote:
>> Guys,
>> Very recently my FS instance has been crashing continuously and there is nothing in the logs to help?
>> I am on log 9 and there is absolutely nothing out of the ordinary. FS just dies on me.
>> This has started happening ever since I started playing with video and proxying media.
>> How do I get more verbose and get to know what crashes FS?
>> 
>> I tried both the Master branch and the V1.4.Beta.
>> 
>> Thanks for any help I'm going nuts.
>> 
>> Emrah
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ldobrodziej at gmail.com  Tue Aug 20 19:25:11 2013
From: ldobrodziej at gmail.com (=?ISO-8859-2?Q?=A3ukasz_Dobrodziej?=)
Date: Tue, 20 Aug 2013 17:25:11 +0200
Subject: [Freeswitch-users] different RTP port ranges per sip profile
Message-ID: <CA+KDsywJUPeNP-qqgc5qMpPGXXop6X2-7WWJpG7Ji=9b+avxeQ@mail.gmail.com>

Hello

Is it possible to define different RTP port ranges for different sip
profiles? For example the Internal sip profile would use ports 60100-60200
for RTP traffic and the External sip profile would use ports 1100-1200. I
can set RTP port range in switch.conf.xml file using rtp-start-port and
rtp-end-port params but it is global configuration. Moreover configuration
from switch.conf.xml forces a continuity of RTP port range (one start and
end). Is it possible to define disjunctive ranges?


-- 
Lucas
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From royce3 at gmail.com  Tue Aug 20 20:01:51 2013
From: royce3 at gmail.com (Royce Mitchell III)
Date: Tue, 20 Aug 2013 11:01:51 -0500
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <CAHwzD9=OuAJ-JDHcZZEM_VdxzGnT1v9En3BvSj36ey96ULwTuA@mail.gmail.com>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
	<CAFiqYumBFafVazhjXUf1CdXgVaYqvbU4QzrPEc2svCLJu0BaFw@mail.gmail.com>
	<CAHwzD9n+CfCbCYNvwPcSiwYQoQe9bved3uK6q1=okh+2ravZoA@mail.gmail.com>
	<A46F2C5A-8082-4155-A4E0-96CE65AF3431@insensate.co.uk>
	<CAHwzD9ncYxxJxaMYnu2qx78Q_SSLp3onz+5=kN=DSz+vRVQGKA@mail.gmail.com>
	<CAFiqYunDhXi+v1H4P=vQnrGW2WRd61w1=nnGwVPo+Pv9CWG3Dw@mail.gmail.com>
	<CAHwzD9=gpZjsuVxdj3O04PMPCh_mqoxj8SudvXak=ejWZJiohw@mail.gmail.com>
	<94920D4E-5645-4BCE-A6A9-C7780E0BDBC6@imtelecom.fr>
	<CAFiqYunihE+DaE71kqX0TtUaSr03nLFT7cXPWHrLgGTVbp=CJg@mail.gmail.com>
	<CAHwzD9=OuAJ-JDHcZZEM_VdxzGnT1v9En3BvSj36ey96ULwTuA@mail.gmail.com>
Message-ID: <CAHwzD9nuO+2hcNi_qG0+2-fnS1dcAGMAg8nO5nQhNKfyT=SZMw@mail.gmail.com>

It happened again this morning. I enabled console loglevel debug and placed
a test call. Then, to try something different, I rebooted the primary
adtran instead of FreeSWITCH. Just like when I restarted FreeSWITCH the
first time, rebooting the Adtran fixed the problem, as calls started
rolling in normally once it came back up.

Given that the adtran is sending these "0" port packets, I think I need to
open a ticket with them. Can I get some advice on what to tell them as to
why sending a one-way audio packet during early media is a problem ( or am
I even describing the problem correctly )?



Royce Mitchell, IT Consultant
ITAS Solutions
royce3 at itas-solutions.com


On Mon, Aug 19, 2013 at 12:59 PM, Royce Mitchell III <royce3 at gmail.com>wrote:

> The other device is not a phone, but a pair of Adtran 908e 2nd Gen doing
> basic PRI to VoIP conversion. All of a sudden all calls were rejecting like
> this, but when I restarted FreeSWITCH everything started working fine
> again. I did not reboot or otherwise do anything to the Adtrans. The
> Adtrans are sending early media and have no reason to receive it, yet, so
> that might explain why they are sending the 0.
>
> On another note, I just searched a recent log file, and I don't find the
> string "m=audio 0" anywhere in the file.
>
>
>
> Royce Mitchell, IT Consultant
> ITAS Solutions
> royce3 at itas-solutions.com
>
>
> On Mon, Aug 19, 2013 at 11:45 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> AFAIK port zero is a method of putting the call on hold in a reinvite,
>> and shouldn't be in the initial invite.
>>
>> Without a port you can only receive audio not send it (as you have
>> nowhere to send to).
>>
>> Your SDP shows the phone is hiding its model - what is it? There's a 2011
>> mailing list  thread suggesting it may be an issue with a bad firmware on a
>> type of phone.
>>
>>
>>
>>
>> On Monday, August 19, 2013, Vallimamod ABDULLAH wrote:
>>
>>> Hi,
>>>
>>> It looks like the remote SDP does not contain correct rtp port
>>> information. You have:
>>>
>>> m=audio 0 RTP/AVP 0 18 101
>>>
>>> The first '0' should be the rtp port normally. I am not sure but it does
>>> not looks correct and may be the cause of the incompatible destination
>>> error.
>>> Hope this helps.
>>>
>>> --
>>> Best Regards,
>>> Vallimamod
>>> .
>>>
>>>
>>>
>>> On Aug 19, 2013, at 2:59 PM, Royce Mitchell III <royce3 at gmail.com>
>>> wrote:
>>>
>>> Here is one example, thanks
>>>
>>>
>>>
>>> Royce Mitchell, IT Consultant
>>> ITAS Solutions
>>> royce3 at itas-solutions.com
>>>
>>>
>>> On Fri, Aug 16, 2013 at 4:12 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>> Can you paste a debug log of the entire lifetime of the call?
>>>
>>>
>>> On 16 August 2013 21:57, Royce Mitchell III <royce3 at gmail.com> wrote:
>>>
>>> The only thing I'm doing in FS regarding codecs is I force PCMU in
>>> certain conditions ( ran into a transcoding bug in FreeSwitch between G722
>>> HD and G711 with the Polycom phones )
>>>
>>>
>>>
>>> Royce Mitchell, IT Consultant
>>> ITAS Solutions
>>> royce3 at itas-solutions.com
>>>
>>>
>>> On Fri, Aug 16, 2013 at 3:35 PM, Lawrence Conroy <
>>> lconroy at insensate.co.uk> wrote:
>>>
>>> Hi there,
>>>  Are you sure about fS allowing PCMU?
>>> According to the remote SDP, your incall is proposing PCMU as its top
>>> choice; g729 is 2nd best (quite right too :).
>>> Forcing the adtran to offer only PCMU should not make ay difference to
>>> that -- it'll still propose PCMU so no change.
>>> Either your setup is somehow blocking PCMU on the b-leg (but I'd expect
>>> to see that on the log), OR is trying to transcode because the b-leg
>>> requires some (non-PCMU) codec and can't (again, I'd expect that to be
>>> logged), OR fS is not accepting PCMU.
>>> Assuming that PCMU is in the fS vars codec lists, does your dialplan do
>>> anything funky with the codec list for an incall?
>>> all the best,
>>>   Lawrence
>>>
>>> On 16 Aug 2013, at 21:06, Royce Mitchell III wrote:
>>> > My FreeSWITCH is configured to prefer PCMU, and the devices it is
>>> talking
>>> > to are Adtran 908e's. The Adtrans are configured for the default codec
>>> > group which is supposed to be PCMU, but I can reconfigure them to
>>> > explicitly allow only PCMU. I will try that and see if it makes a
>>> > difference.
>>> >
>>> >
>>> >
>>> > Royce Mitchell, IT Consultant
>>> > ITAS Solutions
>>> > royce3 at itas-solutions.com
>>> >
>>> >
>>> > On Fri, Aug 16, 2013 at 11:04 AM, Steven Ayre <steveayre at gmail.com>
>>> wrote:
>>> >
>>> >> INCOMPATIBLE_DESTINATION means a codec problem.
>>> >>
>>> >> The remote SDP sends they're offerring PCMU and G729.
>>> >>
>>> >> What codecs are you allowing, what codecs are you bridging with, and
>>> since
>>> >> G729 is on the list are you perhaps trying to transcode without using
>>> >> mod_com_g729+licenses?
>>> >>
>>> >>
>>> >> On 16 August 2013 16:07, Royce Mitchell III <
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 bob.mccarthy at experient.com  Tue Aug 20 20:15:28 2013
From: bob.mccarthy at experient.com (Bob McCarthy)
Date: Tue, 20 Aug 2013 10:15:28 -0600
Subject: [Freeswitch-users] mod_sms endpoints
In-Reply-To: <521238B2.2090400@quentustech.com>
References: <06df01ce9ad4$1018d450$304a7cf0$@experient.com>	<CAMeZp63hV6_QQQJ52ooc7+0dprYRyXcX4ZUa4yM8WtAx3p0ykw@mail.gmail.com>	<071601ce9b6a$e7219720$b564c560$@experient.com>	<521020CB.3000101@quentustech.com>	<075501ce9c2b$07edb9f0$17c92dd0$@experient.com>
	<521238B2.2090400@quentustech.com>
Message-ID: <00bd01ce9dc0$7d7d5310$7877f930$@experient.com>

I think as long as I don't "send" and just use the lua script it should
work.

 or if I change the header to a registered user I know it works

Thanks

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William
King
Sent: Monday, August 19, 2013 9:25 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_sms endpoints

If you have a lua script in your chat plan that returns successful, that
counts as an endpoint delivery.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 08/18/2013 08:53 AM, Bob McCarthy wrote:
> This is the error->
> 
> 2013-08-18 09:51:46.317229 [ERR] sofia_presence.c:265 Chat proto [sip] 
> from ["Win 7" <sip:3038183060 at 192.168.1.8:5060>;tag=f402884a]
> to [911 at 192.168.1.212]
> <font face="Arial, sans-serif"
> size="2">hello</font><L><http://lis.ddti.net:56074/?LbyR=39b93693-a22a
> -45bd-
> ab42-7c91c3ad2f6e><l><B>C0-12-34-56-78-90<b>
> Nobody to send to: Profile internal
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of 
> William King
> Sent: Saturday, August 17, 2013 7:18 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_sms endpoints
> 
> What's the error message you are getting?
> 
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
> 
> On 08/17/2013 09:57 AM, Bob McCarthy wrote:
>> What I end up doing is deleteing the to header and replacing it with 
>> a registered user.  But I would rather not.
>>
>>  
>>
>> *From:*freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
>> *Daniel Ivanov
>> *Sent:* Saturday, August 17, 2013 1:07 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] mod_sms endpoints
>>
>>  
>>
>> I haven't seen fs complain about a message not been sent in the 
>> chatplan. Just process it via a lua handler and do whatever you want 
>> with it.
>>
>> On Aug 17, 2013 2:03 AM, "Bob McCarthy" <bob.mccarthy at experient.com 
>> <mailto:bob.mccarthy at experient.com>> wrote:
>>
>>     Is it possible to create a fictious endpoint to route sms messages
>>     to ? using mod_sms to send messages,  I am intercepting the messages
>>     and disseminating them via the event socket. By sending them to
>>     nowhere I get a nuisance error message.
>>
>>      
>>
>>     Bob McCarthy
>>
>>
>>
> ______________________________________________________________________
> ___
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.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-use
> rs
>>     http://www.freeswitch.org
>>
>>
>>
>> _____________________________________________________________________
>> _ ___ Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>>  
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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-us
>> e
>> rs
>> http://www.freeswitch.org
>>
> 
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
>  
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-use
> rs
> http://www.freeswitch.org
> 
> 
> 
> 
> ______________________________________________________________________
> ___ Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
>  
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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-use
> rs
> http://www.freeswitch.org
> 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 sravi123 at yahoo.com  Tue Aug 20 21:11:50 2013
From: sravi123 at yahoo.com (Ravi)
Date: Tue, 20 Aug 2013 22:41:50 +0530
Subject: [Freeswitch-users] Transfer / Forward calls - from an operator
 to an extension after the call is picked
In-Reply-To: <CAHKqMb4Xzwswcr3PtbA5OT-AVGSNZ4+BiiW5v4B8e0=g1o1PUQ@mail.gmail.com>
References: <1376827283.11657.YahooMailNeo@web160501.mail.bf1.yahoo.com>
	<CAHKqMb5SK8V5h-CYf8uaivj_LkvMR-xjQ1k+gG7UA_0z_5g44w@mail.gmail.com>
	<521258E8.90900@yahoo.com>
	<CAHKqMb4Xzwswcr3PtbA5OT-AVGSNZ4+BiiW5v4B8e0=g1o1PUQ@mail.gmail.com>
Message-ID: <5213A356.4030805@yahoo.com>

Thanks again, Nandy. The TRANSFER and HOLD buttons have been there for 
long, but went unnoticed :) For some reason, I was thinking that there 
should be some provision in freeswitch to transfer calls. But it has 
turned out to be much simpler. I have Grandstream phones, and I am able 
to transfer calls.
Since this solution worked, I have not checked the default.xml for call 
transfers. I shall check them soon.

Thanks.
Ravi


On 20/08/13 1:53 AM, Nandy Dagondon wrote:
> Hi Ravi,
>
> Nice to know it helped. What  you're describing is Supervised Call 
> Transfer. Locate your phone's HOLD and TRANSFER buttons.
> 1. Press HOLD to place the caller on-hold. She'll enjoy the music :-)
> 2. You'll hear a dialtone. Dial extension 1010. Talk to the 3rd party.
> 3. If she accepts it, press the TRANSFER button then hang-up.
> 4. If not, press the HOLD button to resume talking to the caller.
>
> For analog phones, the vanilla default.xml dialplan has an example in 
> the Local extension section what keys are bound to call transfers.
>
> /Nandy
>
>
> On Tue, Aug 20, 2013 at 1:42 AM, Ravi <sravi123 at yahoo.com 
> <mailto:sravi123 at yahoo.com>> wrote:
>
>     Thank you Nandy for the response. It helped, and I am able to
>     route calls.
>
>     I am able to transfer the call, when it comes in, to the operator
>     through Freeswitch. After the operator takes the call, the
>     operator talks to the caller for a couple of minutes, and then has
>     to transfer the call to one of the extensions say 1010. How is
>     that done ? Or as a simple case, how can i transfer a call that I
>     have received to another extension ?
>
>     Perhaps this is very easy, but I am not sure where to find this
>     information.
>
>     Thanks.
>     Ravi
>
>
>     On 19/08/13 9:24 AM, Nandy Dagondon wrote:
>>     Hi,
>>
>>     1. Copy your working dialplan entry.
>>     2. Add <condition field="caller_id_number" data="4302000"/>
>>     3. Change the bridge/transfer application to connect to the operator
>>
>>     Take note to place the above entry ahead of your working entry. I
>>     hope this helps.
>>     /Nandy
>>
>>
>>     On Sun, Aug 18, 2013 at 8:01 PM, Ravi <sravi123 at yahoo.com
>>     <mailto:sravi123 at yahoo.com>> wrote:
>>
>>         Hello Everyone !
>>
>>         I have successfully configured my PRI connection. I am able
>>         to send and receive calls. Now I am trying to have an
>>         operator handle one number - say 4302000. When any call is
>>         received the operator will talk to the caller and need to
>>         transfer the call to an extension. I tried looking at
>>         dialplan documentation, and I think I find only instances
>>         where the transfer is done within the program and not when
>>         some one picks up the call.
>>
>>         Can some one please point me to right place/ documents where
>>         I can find information regarding this.
>>
>>         I did find a link with some similar request, but the
>>         information is limited.
>>         http://lists.freeswitch.org/pipermail/freeswitch-users/2012-October/088579.html
>>
>>         Thanks for your help.
>>         Ravi
>>
>>
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.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
>>
>>
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org  <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.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
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jpyle at fidelityvoice.com  Tue Aug 20 21:12:55 2013
From: jpyle at fidelityvoice.com (Jeff Pyle)
Date: Tue, 20 Aug 2013 13:12:55 -0400
Subject: [Freeswitch-users] in/outbound-use-uuid-as-callid as channel
	variables
Message-ID: <CACYJG3J93Ge1Qca+=jacNfP57eWPTKu7kFaovVgkSSbgcbEcEA@mail.gmail.com>

Hello,

Is it possible to set the equivalent of inbound-use-uuid-as-callid
or outbound-use-uuid-as-callid per channel from within the dialplan?


- Jeff
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From sravi123 at yahoo.com  Tue Aug 20 21:13:51 2013
From: sravi123 at yahoo.com (Ravi)
Date: Tue, 20 Aug 2013 22:43:51 +0530
Subject: [Freeswitch-users] Transfer / Forward calls - from an operator
 to an extension after the call is picked
In-Reply-To: <CAOkuKdeSKj6bNpwXYM+CFyoUBT8Hup-2n-DjNLFN=OhhT-U9kw@mail.gmail.com>
References: <1376827283.11657.YahooMailNeo@web160501.mail.bf1.yahoo.com>
	<CAHKqMb5SK8V5h-CYf8uaivj_LkvMR-xjQ1k+gG7UA_0z_5g44w@mail.gmail.com>
	<521258E8.90900@yahoo.com>
	<CAHKqMb4Xzwswcr3PtbA5OT-AVGSNZ4+BiiW5v4B8e0=g1o1PUQ@mail.gmail.com>
	<CAOkuKdeSKj6bNpwXYM+CFyoUBT8Hup-2n-DjNLFN=OhhT-U9kw@mail.gmail.com>
Message-ID: <5213A3CF.5080600@yahoo.com>

Thanks Steve. I have a couple of Grandstream phones. The Grandstream 
phones work the same way as you have outlined in your mail. I do need 
some more phones for my office, and since you have mentioned about 
Polycom, I could try them this time.

Thanks again for the response.
Ravi


On 20/08/13 3:41 AM, Steven Schoch wrote:
> It seems that different phones behave differently regarding call 
> transfer. On our Polycom phones, you press Trans, which puts the 
> caller on hold and prompts for another number to dial. After you call 
> the extension, and talk to the 3rd party, you press Trans again. You 
> can't transfer a call that has been put on hold without taking it off 
> hold first.
>
> -- 
> Steve
>
>
> On Mon, Aug 19, 2013 at 1:23 PM, Nandy Dagondon <nandy1925 at gmail.com 
> <mailto:nandy1925 at gmail.com>> wrote:
>
>     Hi Ravi,
>
>     Nice to know it helped. What  you're describing is Supervised Call
>     Transfer. Locate your phone's HOLD and TRANSFER buttons.
>     1. Press HOLD to place the caller on-hold. She'll enjoy the music :-)
>     2. You'll hear a dialtone. Dial extension 1010. Talk to the 3rd party.
>     3. If she accepts it, press the TRANSFER button then hang-up.
>     4. If not, press the HOLD button to resume talking to the caller.
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Tue Aug 20 22:27:19 2013
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Aug 2013 11:27:19 -0700
Subject: [Freeswitch-users] List of $$ variables
In-Reply-To: <FB7D97A214987F458242ACBDF87614073B1BD1995B@clubvirtual40.ClubMessage.local>
References: <FB7D97A214987F458242ACBDF87614073B1BD1995B@clubvirtual40.ClubMessage.local>
Message-ID: <CAKzWOxWS73OZjjRpSU+wbKStFd541wTMz8kfC31O2OCAjh+7sQ@mail.gmail.com>

There isn't a list per se, since technically any variable could be used
with $$. For this item it's best to search the wiki for "bind_server_ip"
which yields this page:

https://wiki.freeswitch.org/wiki/Getting_Started_Guide#Preprocessor_Variables

-MC


On Tue, Aug 20, 2013 at 5:53 AM, Grant Bagdasarian <GB at cm.nl> wrote:

> Hello,****
>
> ** **
>
> Is there a list available for variables starting with $$? Like
> $${local_ip_v4}?****
>
> ** **
>
> Does the following xml element support multiple values?****
>
> <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto "/>****
>
> ** **
>
> Something like:****
>
> <X-PRE-PROCESS cmd="set" data="bind_server_ip=192.168.1.1,10.0.0.1 "/>
> #Notice the comma separating the bind IP?s.****
>
> ** **
>
> Thanks,****
>
> ** **
>
> Grant****
>
> ** **
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From karl at xtronics.com  Tue Aug 20 23:54:35 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Tue, 20 Aug 2013 14:54:35 -0500
Subject: [Freeswitch-users] nate-mode -nonat apply-nat-acl documentation
Message-ID: <5213C97B.2060108@xtronics.com>

OK -nonat turns off NAT detection, but I don't know if this applies to external SIP clients behind 
NAT?

===============================================================================================
There are clearly 2 separate cases where NAT detection may be needed and some way to enable each one 
independent of the other.
===============================================================================================

The following explores my confusion:

apply-nat-acl

If value is rfc1918 enable 'nat-mode' if IP matches - OK but what exactly is nat-mode?

And what is value="nat.auto" ?  How does it relate to auto-nat?

Does any of this do anything if I'm running with -nonat ?

Is running with -nonat the opposite of nat-mode?

nat-mode could mean several things and I can't find it defined anywhere.

There is also -nonatmap which allows NAT detection but avoids UPnP/NAT-PMP


In my situation, I have FS situated behind a firewall and have the proper ports DNATed.  I'm running 
with -nonat and this works, and should not create any problems.  BUT there could be situations where 
I am out of town and need to connect to the server from behind some NAT that I have no control over. 
Thus there are two separate situations that might want NAT detection - FS behind NAT and a SIP 
client behind NAT.

ext-rtp-ip, ext-sip-ip, rtp-ip, and sip-ip  Should tell FS what it needs to know about the server 
end - but I'm not finding clarity on what a sane setup would be for the possibility of an external 
SIP client behind NAT.

,.,.

I think the best way to document this is to come up with a few scenarios and show a sane set up for 
each one.  We can assume in all cases that phones on both sides of NAT need to register and if the 
server is on the LAN side of a NAT there could be need for a double transit of NAT.

1 - FS server on a LAN with ports DNATed by firewall - static IP - Phones on LAN and phones 
registering from Internet

2 - FS on public IP  by phones on the other side of a NAT firewall need to register as well as 
phones on bare internet

3 - FS server on LAN with firewall on dynamic IP

4 -


Just so we are on the same page:
DNAT (Destination network address translation) DNAT is a technique for transparently changing the 
destination IP address of an en route packet and performing the inverse function for any replies. 
This use of DNAT is also called port forwarding.

I'm trying to document this for myself and finding it similar to walking in a swamp each step is 
getting me bogged down.



--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

Inflation is taxation without legislation.
Milton Friedman
--------------------------------------------------------------------------------


From karl at xtronics.com  Wed Aug 21 00:02:04 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Tue, 20 Aug 2013 15:02:04 -0500
Subject: [Freeswitch-users] List of $$ variables
In-Reply-To: <CAKzWOxWS73OZjjRpSU+wbKStFd541wTMz8kfC31O2OCAjh+7sQ@mail.gmail.com>
References: <FB7D97A214987F458242ACBDF87614073B1BD1995B@clubvirtual40.ClubMessage.local>
	<CAKzWOxWS73OZjjRpSU+wbKStFd541wTMz8kfC31O2OCAjh+7sQ@mail.gmail.com>
Message-ID: <5213CB3C.7020109@xtronics.com>

On 08/20/2013 01:27 PM, Michael Collins wrote:
> There isn't a list per se, since technically any variable could be used with $$. For this item it's
> best to search the wiki for "bind_server_ip" which yields this page:
>
> https://wiki.freeswitch.org/wiki/Getting_Started_Guide#Preprocessor_Variables

I think his question is a list of ones that are not defined in the configs - but already defined.

$${local_ip_v4} is set before vars.xml is scanned.

A list of all the available predefined variables could be useful.



--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

Why are so many spending time watching dark movies about
hopelessness, the macabre, and perversion; why are they reading
books about unfaithfulness and self destruction?  Why is nothing
uplifting, also considered 'cool' or entertaining? -kps
--------------------------------------------------------------------------------


From mike at jerris.com  Wed Aug 21 00:10:43 2013
From: mike at jerris.com (Michael Jerris)
Date: Tue, 20 Aug 2013 16:10:43 -0400
Subject: [Freeswitch-users] List of $$ variables
In-Reply-To: <5213CB3C.7020109@xtronics.com>
References: <FB7D97A214987F458242ACBDF87614073B1BD1995B@clubvirtual40.ClubMessage.local>
	<CAKzWOxWS73OZjjRpSU+wbKStFd541wTMz8kfC31O2OCAjh+7sQ@mail.gmail.com>
	<5213CB3C.7020109@xtronics.com>
Message-ID: <FBABE432-943A-4461-841F-3C9F4396383C@jerris.com>

mike at mike [Tue Aug 20 04:08 PM]  ~/src/freeswitch/src
<5>:git grep -n switch_core_set_variable *.c

?.

hostname
local_ip_v4
local_mask_v4
local_ip_v6
switch_serial
base_dir
recordings_dir
sound_prefix
sounds_dir
core_uuid
zrtp_enabled
nat_public_addr
nat_private_addr
nat_type


On Aug 20, 2013, at 4:02 PM, Karl Schmidt <karl at xtronics.com> wrote:

> On 08/20/2013 01:27 PM, Michael Collins wrote:
>> There isn't a list per se, since technically any variable could be used with $$. For this item it's
>> best to search the wiki for "bind_server_ip" which yields this page:
>> 
>> https://wiki.freeswitch.org/wiki/Getting_Started_Guide#Preprocessor_Variables
> 
> I think his question is a list of ones that are not defined in the configs - but already defined.
> 
> $${local_ip_v4} is set before vars.xml is scanned.
> 
> A list of all the available predefined variables could be useful.
> 



From karl at xtronics.com  Wed Aug 21 00:32:19 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Tue, 20 Aug 2013 15:32:19 -0500
Subject: [Freeswitch-users] List of $$ variables
In-Reply-To: <FBABE432-943A-4461-841F-3C9F4396383C@jerris.com>
References: <FB7D97A214987F458242ACBDF87614073B1BD1995B@clubvirtual40.ClubMessage.local>
	<CAKzWOxWS73OZjjRpSU+wbKStFd541wTMz8kfC31O2OCAjh+7sQ@mail.gmail.com>
	<5213CB3C.7020109@xtronics.com>
	<FBABE432-943A-4461-841F-3C9F4396383C@jerris.com>
Message-ID: <5213D253.7030906@xtronics.com>

Thanks - this list probably should exist at the top of vars.xml as a comment

On 08/20/2013 03:10 PM, Michael Jerris wrote:
> hostname
> local_ip_v4
> local_mask_v4
> local_ip_v6
> switch_serial
> base_dir
> recordings_dir
> sound_prefix
> sounds_dir
> core_uuid
> zrtp_enabled
> nat_public_addr
> nat_private_addr
> nat_type


--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

Opportunity is missed by most people because it is dressed in overalls and looks like work.
Thomas Alva Edison

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


From vallimamod.abdullah at imtelecom.fr  Wed Aug 21 00:52:11 2013
From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH)
Date: Tue, 20 Aug 2013 22:52:11 +0200
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <CAHwzD9nuO+2hcNi_qG0+2-fnS1dcAGMAg8nO5nQhNKfyT=SZMw@mail.gmail.com>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
	<CAFiqYumBFafVazhjXUf1CdXgVaYqvbU4QzrPEc2svCLJu0BaFw@mail.gmail.com>
	<CAHwzD9n+CfCbCYNvwPcSiwYQoQe9bved3uK6q1=okh+2ravZoA@mail.gmail.com>
	<A46F2C5A-8082-4155-A4E0-96CE65AF3431@insensate.co.uk>
	<CAHwzD9ncYxxJxaMYnu2qx78Q_SSLp3onz+5=kN=DSz+vRVQGKA@mail.gmail.com>
	<CAFiqYunDhXi+v1H4P=vQnrGW2WRd61w1=nnGwVPo+Pv9CWG3Dw@mail.gmail.com>
	<CAHwzD9=gpZjsuVxdj3O04PMPCh_mqoxj8SudvXak=ejWZJiohw@mail.gmail.com>
	<94920D4E-5645-4BCE-A6A9-C7780E0BDBC6@imtelecom.fr>
	<CAFiqYunihE+DaE71kqX0TtUaSr03nLFT7cXPWHrLgGTVbp=CJg@mail.gmail.com>
	<CAHwzD9=OuAJ-JDHcZZEM_VdxzGnT1v9En3BvSj36ey96ULwTuA@mail.gmail.com>
	<CAHwzD9nuO+2hcNi_qG0+2-fnS1dcAGMAg8nO5nQhNKfyT=SZMw@mail.gmail.com>
Message-ID: <FA8C09A8-2FDB-4BCF-AEFE-66D7581E26CF@imtelecom.fr>

Hi,

What is really strange is that your problem does not happen consistently.

Have you checked for any sip enabled firewall or rtp proxy on the call path?
You can make a sip trace with ngrep on the interface connected to the pri gateway and check if you get the 0 port sdp everytime from the adtran or only when you get the error on freeswitch (in wich case, you can point that to their support.)
Make also ngrep trace on the freeswith interface and on any router in the middle to check if your invite is modified along the path.

-- 
Cheers,
Vallimamod
.

On Aug 20, 2013, at 6:01 PM, Royce Mitchell III <royce3 at gmail.com> wrote:

> It happened again this morning. I enabled console loglevel debug and placed a test call. Then, to try something different, I rebooted the primary adtran instead of FreeSWITCH. Just like when I restarted FreeSWITCH the first time, rebooting the Adtran fixed the problem, as calls started rolling in normally once it came back up.
> 
> Given that the adtran is sending these "0" port packets, I think I need to open a ticket with them. Can I get some advice on what to tell them as to why sending a one-way audio packet during early media is a problem ( or am I even describing the problem correctly )?
> 
> 
> 
> Royce Mitchell, IT Consultant
> ITAS Solutions
> royce3 at itas-solutions.com
> 
> 
> On Mon, Aug 19, 2013 at 12:59 PM, Royce Mitchell III <royce3 at gmail.com> wrote:
> The other device is not a phone, but a pair of Adtran 908e 2nd Gen doing basic PRI to VoIP conversion. All of a sudden all calls were rejecting like this, but when I restarted FreeSWITCH everything started working fine again. I did not reboot or otherwise do anything to the Adtrans. The Adtrans are sending early media and have no reason to receive it, yet, so that might explain why they are sending the 0.
> 
> On another note, I just searched a recent log file, and I don't find the string "m=audio 0" anywhere in the file.
> 
> 
> 
> Royce Mitchell, IT Consultant
> ITAS Solutions
> royce3 at itas-solutions.com
> 
> 
> On Mon, Aug 19, 2013 at 11:45 AM, Steven Ayre <steveayre at gmail.com> wrote:
> AFAIK port zero is a method of putting the call on hold in a reinvite, and shouldn't be in the initial invite.
> 
> Without a port you can only receive audio not send it (as you have nowhere to send to).
> 
> Your SDP shows the phone is hiding its model - what is it? There's a 2011 mailing list  thread suggesting it may be an issue with a bad firmware on a type of phone.
> 
> 
> 
> 
> On Monday, August 19, 2013, Vallimamod ABDULLAH wrote:
> Hi,
> 
> It looks like the remote SDP does not contain correct rtp port information. You have:
> 
> m=audio 0 RTP/AVP 0 18 101
> 
> The first '0' should be the rtp port normally. I am not sure but it does not looks correct and may be the cause of the incompatible destination error.
> Hope this helps.
> 
> -- 
> Best Regards,
> Vallimamod
> .
> 
> 
> 
> On Aug 19, 2013, at 2:59 PM, Royce Mitchell III <royce3 at gmail.com> wrote:
> 
>> Here is one example, thanks
>> 
>> 
>> 
>> Royce Mitchell, IT Consultant
>> ITAS Solutions
>> royce3 at itas-solutions.com
>> 
>> 
>> On Fri, Aug 16, 2013 at 4:12 PM, Steven Ayre <steveayre at gmail.com> wrote:
>> Can you paste a debug log of the entire lifetime of the call?
>> 
>> 
>> On 16 August 2013 21:57, Royce Mitchell III <royce3 at gmail.com> wrote:
>> The only thing I'm doing in FS regarding codecs is I force PCMU in certain conditions ( ran into a transcoding bug in FreeSwitch between G722 HD and G711 with the Polycom phones )
>> 
>> 
>> 
>> Royce Mitchell, IT Consultant
>> ITAS Solutions
>> royce3 at itas-solutions.com
>> 
>> 
>> On Fri, Aug 16, 2013 at 3:35 PM, Lawrence Conroy <lconroy at insensate.co.uk> wrote:
>> Hi there,
>>  Are you sure about fS allowing PCMU?
>> According to the remote SDP, your incall is proposing PCMU as its top choice; g729 is 2nd best (quite right too :).
>> Forcing the adtran to offer only PCMU should not make ay difference to that -- it'll still propose PCMU so no change.
>> Either your setup is somehow blocking PCMU on the b-leg (but I'd expect to see that on the log), OR is trying to transcode because the b-leg requires some (non-PCMU) codec and can't (again, I'd expect that to be logged), OR fS is not accepting PCMU.
>> Assuming that PCMU is in the fS vars codec lists, does your dialplan do anything funky with the codec list for an incall?
>> all the best,
>>   Lawrence
>> 
>> On 16 Aug 2013, at 21:06, Royce Mitchell III wrote:
>> > My FreeSWITCH is configured to prefer PCMU, and the devices it is talking
>> > to are Adtran 908e's. The Adtrans are configured for the default codec
>> > group which is supposed to be PCMU, but I can reconfigure them to
>> > explicitly allow only PCMU. I will try that and see if it makes a
>> > difference.
>> >
>> >
>> >
>> > Royce Mitchell, IT Consultant
>> > ITAS Solutions
>> > royce3 at itas-solutions.com
>> >
>> >
>> > On Fri, Aug 16, 2013 at 11:04 AM, Steven Ayre <steveayre at gmail.com> wrote:
>> >
>> >> INCOMPATIBLE_DESTINATION means a codec problem.
>> >>
>> >> The remote SDP sends they're offerring PCMU and G729.
>> >>
>> >> What codecs are you allowing, what codecs are you bridging with, and since
>> >> G729 is on the list are you perhaps trying to transcode without using
>> >> mod_com_g729+licenses?
>> >>
>> >>
>> >> On 16 August 2013 16:07, Royce Mitchell III <
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 royce3 at gmail.com  Wed Aug 21 01:18:00 2013
From: royce3 at gmail.com (Royce Mitchell III)
Date: Tue, 20 Aug 2013 16:18:00 -0500
Subject: [Freeswitch-users] INCOMPATIBLE_DESTINATION
In-Reply-To: <FA8C09A8-2FDB-4BCF-AEFE-66D7581E26CF@imtelecom.fr>
References: <CAHwzD9k47-dwT1CPvKn06z1PxsXUxJXs4qEpMupGfcY+Kq00oA@mail.gmail.com>
	<CAFiqYumBFafVazhjXUf1CdXgVaYqvbU4QzrPEc2svCLJu0BaFw@mail.gmail.com>
	<CAHwzD9n+CfCbCYNvwPcSiwYQoQe9bved3uK6q1=okh+2ravZoA@mail.gmail.com>
	<A46F2C5A-8082-4155-A4E0-96CE65AF3431@insensate.co.uk>
	<CAHwzD9ncYxxJxaMYnu2qx78Q_SSLp3onz+5=kN=DSz+vRVQGKA@mail.gmail.com>
	<CAFiqYunDhXi+v1H4P=vQnrGW2WRd61w1=nnGwVPo+Pv9CWG3Dw@mail.gmail.com>
	<CAHwzD9=gpZjsuVxdj3O04PMPCh_mqoxj8SudvXak=ejWZJiohw@mail.gmail.com>
	<94920D4E-5645-4BCE-A6A9-C7780E0BDBC6@imtelecom.fr>
	<CAFiqYunihE+DaE71kqX0TtUaSr03nLFT7cXPWHrLgGTVbp=CJg@mail.gmail.com>
	<CAHwzD9=OuAJ-JDHcZZEM_VdxzGnT1v9En3BvSj36ey96ULwTuA@mail.gmail.com>
	<CAHwzD9nuO+2hcNi_qG0+2-fnS1dcAGMAg8nO5nQhNKfyT=SZMw@mail.gmail.com>
	<FA8C09A8-2FDB-4BCF-AEFE-66D7581E26CF@imtelecom.fr>
Message-ID: <CAHwzD9=c8z4MNxdE3zVTDDq7Fxih+ZSijJG_FG3+oUoJuoj_XQ@mail.gmail.com>

I agree, the inconsistency is really baffling. I am intimately familiar
with this environment and can assure there is nothing but basic network
switches between the Adtrans and the FreeSWITCH instances. They are all in
the same rack and plugged into Dell PowerEdge switches.



Royce Mitchell, IT Consultant
ITAS Solutions
royce3 at itas-solutions.com


On Tue, Aug 20, 2013 at 3:52 PM, Vallimamod ABDULLAH <
vallimamod.abdullah at imtelecom.fr> wrote:

> Hi,
>
> What is really strange is that your problem does not happen consistently.
>
> Have you checked for any sip enabled firewall or rtp proxy on the call
> path?
> You can make a sip trace with ngrep on the interface connected to the pri
> gateway and check if you get the 0 port sdp everytime from the adtran or
> only when you get the error on freeswitch (in wich case, you can point that
> to their support.)
> Make also ngrep trace on the freeswith interface and on any router in the
> middle to check if your invite is modified along the path.
>
> --
> Cheers,
> Vallimamod
> .
>
> On Aug 20, 2013, at 6:01 PM, Royce Mitchell III <royce3 at gmail.com> wrote:
>
> It happened again this morning. I enabled console loglevel debug and
> placed a test call. Then, to try something different, I rebooted the
> primary adtran instead of FreeSWITCH. Just like when I restarted FreeSWITCH
> the first time, rebooting the Adtran fixed the problem, as calls started
> rolling in normally once it came back up.
>
> Given that the adtran is sending these "0" port packets, I think I need to
> open a ticket with them. Can I get some advice on what to tell them as to
> why sending a one-way audio packet during early media is a problem ( or am
> I even describing the problem correctly )?
>
>
>
> Royce Mitchell, IT Consultant
> ITAS Solutions
> royce3 at itas-solutions.com
>
>
> On Mon, Aug 19, 2013 at 12:59 PM, Royce Mitchell III <royce3 at gmail.com>wrote:
>
>> The other device is not a phone, but a pair of Adtran 908e 2nd Gen doing
>> basic PRI to VoIP conversion. All of a sudden all calls were rejecting like
>> this, but when I restarted FreeSWITCH everything started working fine
>> again. I did not reboot or otherwise do anything to the Adtrans. The
>> Adtrans are sending early media and have no reason to receive it, yet, so
>> that might explain why they are sending the 0.
>>
>> On another note, I just searched a recent log file, and I don't find the
>> string "m=audio 0" anywhere in the file.
>>
>>
>>
>> Royce Mitchell, IT Consultant
>> ITAS Solutions
>> royce3 at itas-solutions.com
>>
>>
>> On Mon, Aug 19, 2013 at 11:45 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>
>>> AFAIK port zero is a method of putting the call on hold in a reinvite,
>>> and shouldn't be in the initial invite.
>>>
>>> Without a port you can only receive audio not send it (as you have
>>> nowhere to send to).
>>>
>>> Your SDP shows the phone is hiding its model - what is it? There's a
>>> 2011 mailing list  thread suggesting it may be an issue with a bad firmware
>>> on a type of phone.
>>>
>>>
>>>
>>>
>>> On Monday, August 19, 2013, Vallimamod ABDULLAH wrote:
>>>
>>>> Hi,
>>>>
>>>> It looks like the remote SDP does not contain correct rtp port
>>>> information. You have:
>>>>
>>>> m=audio 0 RTP/AVP 0 18 101
>>>>
>>>> The first '0' should be the rtp port normally. I am not sure but it
>>>> does not looks correct and may be the cause of the incompatible destination
>>>> error.
>>>> Hope this helps.
>>>>
>>>> --
>>>> Best Regards,
>>>> Vallimamod
>>>> .
>>>>
>>>>
>>>>
>>>> On Aug 19, 2013, at 2:59 PM, Royce Mitchell III <royce3 at gmail.com>
>>>> wrote:
>>>>
>>>> Here is one example, thanks
>>>>
>>>>
>>>>
>>>> Royce Mitchell, IT Consultant
>>>> ITAS Solutions
>>>> royce3 at itas-solutions.com
>>>>
>>>>
>>>> On Fri, Aug 16, 2013 at 4:12 PM, Steven Ayre <steveayre at gmail.com>wrote:
>>>>
>>>> Can you paste a debug log of the entire lifetime of the call?
>>>>
>>>>
>>>> On 16 August 2013 21:57, Royce Mitchell III <royce3 at gmail.com> wrote:
>>>>
>>>> The only thing I'm doing in FS regarding codecs is I force PCMU in
>>>> certain conditions ( ran into a transcoding bug in FreeSwitch between G722
>>>> HD and G711 with the Polycom phones )
>>>>
>>>>
>>>>
>>>> Royce Mitchell, IT Consultant
>>>> ITAS Solutions
>>>> royce3 at itas-solutions.com
>>>>
>>>>
>>>> On Fri, Aug 16, 2013 at 3:35 PM, Lawrence Conroy <
>>>> lconroy at insensate.co.uk> wrote:
>>>>
>>>> Hi there,
>>>>  Are you sure about fS allowing PCMU?
>>>> According to the remote SDP, your incall is proposing PCMU as its top
>>>> choice; g729 is 2nd best (quite right too :).
>>>> Forcing the adtran to offer only PCMU should not make ay difference to
>>>> that -- it'll still propose PCMU so no change.
>>>> Either your setup is somehow blocking PCMU on the b-leg (but I'd expect
>>>> to see that on the log), OR is trying to transcode because the b-leg
>>>> requires some (non-PCMU) codec and can't (again, I'd expect that to be
>>>> logged), OR fS is not accepting PCMU.
>>>> Assuming that PCMU is in the fS vars codec lists, does your dialplan do
>>>> anything funky with the codec list for an incall?
>>>> all the best,
>>>>   Lawrence
>>>>
>>>> On 16 Aug 2013, at 21:06, Royce Mitchell III wrote:
>>>> > My FreeSWITCH is configured to prefer PCMU, and the devices it is
>>>> talking
>>>> > to are Adtran 908e's. The Adtrans are configured for the default codec
>>>> > group which is supposed to be PCMU, but I can reconfigure them to
>>>> > explicitly allow only PCMU. I will try that and see if it makes a
>>>> > difference.
>>>> >
>>>> >
>>>> >
>>>> > Royce Mitchell, IT Consultant
>>>> > ITAS Solutions
>>>> > royce3 at itas-solutions.com
>>>> >
>>>> >
>>>> > On Fri, Aug 16, 2013 at 11:04 AM, Steven Ayre <steveayre at gmail.com>
>>>> wrote:
>>>> >
>>>> >> INCOMPATIBLE_DESTINATION means a codec problem.
>>>> >>
>>>> >> The remote SDP sends they're offerring PCMU and G729.
>>>> >>
>>>> >> What codecs are you allowing, what codecs are you bridging with, and
>>>> since
>>>> >> G729 is on the list are you perhaps trying to transcode without using
>>>> >> mod_com_g729+licenses?
>>>> >>
>>>> >>
>>>> >> On 16 August 2013 16:07, Royce Mitchell III <
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Wed Aug 21 01:34:12 2013
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 20 Aug 2013 14:34:12 -0700
Subject: [Freeswitch-users] List of $$ variables
In-Reply-To: <5213D253.7030906@xtronics.com>
References: <FB7D97A214987F458242ACBDF87614073B1BD1995B@clubvirtual40.ClubMessage.local>
	<CAKzWOxWS73OZjjRpSU+wbKStFd541wTMz8kfC31O2OCAjh+7sQ@mail.gmail.com>
	<5213CB3C.7020109@xtronics.com>
	<FBABE432-943A-4461-841F-3C9F4396383C@jerris.com>
	<5213D253.7030906@xtronics.com>
Message-ID: <CAKzWOxWZGStzHyOVWPOao7YwdAc2GWDC0cxjEFuB=Ek76r56Lg@mail.gmail.com>

If you add those to your vars.xml, make a diff, and then throw it on a Jira
then one of the committers will gladly add it.
-MC


On Tue, Aug 20, 2013 at 1:32 PM, Karl Schmidt <karl at xtronics.com> wrote:

> Thanks - this list probably should exist at the top of vars.xml as a
> comment
>
> On 08/20/2013 03:10 PM, Michael Jerris wrote:
> > hostname
> > local_ip_v4
> > local_mask_v4
> > local_ip_v6
> > switch_serial
> > base_dir
> > recordings_dir
> > sound_prefix
> > sounds_dir
> > core_uuid
> > zrtp_enabled
> > nat_public_addr
> > nat_private_addr
> > nat_type
>
>
>
> --------------------------------------------------------------------------------
> Karl Schmidt                                  EMail Karl at xtronics.com
> Transtronics, Inc.                              WEB
> http://secure.transtronics.com
> 3209 West 9th Street                             Ph (785) 841-3089
> Lawrence, KS 66049                              FAX (785) 841-0434
>
> Opportunity is missed by most people because it is dressed in overalls and
> looks like work.
> Thomas Alva Edison
>
>
> --------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From fs.user at fordior.net  Wed Aug 21 03:30:09 2013
From: fs.user at fordior.net (EL)
Date: Wed, 21 Aug 2013 01:30:09 +0200
Subject: [Freeswitch-users] nate-mode -nonat apply-nat-acl documentation
In-Reply-To: <5213C97B.2060108@xtronics.com>
References: <5213C97B.2060108@xtronics.com>
Message-ID: <20130820233009.GA1746@0rdior.com>

Karl,

I agree with you on the documentation part. Unfortunately, I'm not
able to give you the answers. I had difficulties getting my own setup
working when I was changing my setup (included a vpn). In the end I
figured it out myself with wireshark, but the documentation is (at
least to me) not that clear on the subject.

Maybe because of two reasons: 
[1] The average FS user has already a fair amount of knowledge on the subject; 
[2] The average FS user has the same kind of setup, which is documented enough.

-- 
EL


From karl at xtronics.com  Wed Aug 21 05:03:39 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Tue, 20 Aug 2013 20:03:39 -0500
Subject: [Freeswitch-users] nate-mode -nonat apply-nat-acl documentation
In-Reply-To: <20130820233009.GA1746@0rdior.com>
References: <5213C97B.2060108@xtronics.com> <20130820233009.GA1746@0rdior.com>
Message-ID: <521411EB.4070108@xtronics.com>

On 08/20/2013 06:30 PM, EL wrote:
> Karl,
>
> I agree with you on the documentation part. Unfortunately, I'm not
> able to give you the answers. I had difficulties getting my own setup
> working when I was changing my setup (included a vpn). In the end I
> figured it out myself with wireshark, but the documentation is (at
> least to me) not that clear on the subject.
>
> Maybe because of two reasons:
> [1] The average FS user has already a fair amount of knowledge on the subject;
> [2] The average FS user has the same kind of setup, which is documented enough.

I'm not complaining - This is some complex code and it could be very hard for those closest to it to 
document it. (Hard to see the forest for all the trees in the way).


The bit that I can't figure out is what controls the two cases of NAT helpers (server end vs client 
behind NAT ).

I could be that I should turn all the NAT stuff off - most of the SIP clients can do stun or other 
tricks to get past NAT. If I could just get some of the FS terms defined and know if using the nat 
parameters turns any of this on or off, I could even write it up.

I've been searching the mailing list with:

http://www.google.com/search?q=site:lists.freeswitch.org%2Fpipermail%2Ffreeswitch-users+searchterm

but I'm not making much progress.



--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

Truth is mighty and will prevail.
There is nothing wrong with this,
except that it ain't so.
--Mark Twain
--------------------------------------------------------------------------------


From babak.freeswitch at gmail.com  Wed Aug 21 09:28:49 2013
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Wed, 21 Aug 2013 09:58:49 +0430
Subject: [Freeswitch-users] Play sound over local_stream (Mix sounds)
In-Reply-To: <CAJEstwPx5EQtt=2Xt1ZV9jcsFr2f_V5LMrNNT6O7xe2_NbvE5Q@mail.gmail.com>
References: <CAJEstwPx5EQtt=2Xt1ZV9jcsFr2f_V5LMrNNT6O7xe2_NbvE5Q@mail.gmail.com>
Message-ID: <CAEVfcabueb+Vd8UQzni_wZkT6Z0GQCoOOuErEA+-FzxG_+wA=g@mail.gmail.com>

 you can use uuid_displace.
http://wiki.freeswitch.org/wiki/Mod_commands#uuid_displace


On Tue, Aug 20, 2013 at 4:59 PM, Sayyed Mohammad Emami Razavi <
emamirazavi at gmail.com> wrote:

> I want play sound over local_stream without stopping it!
> Now i use uuid_broadcast api with aleg argument but it stops MOH
> any solution?!
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Wed Aug 21 09:31:58 2013
From: eidevm5 at gmail.com (Peter)
Date: Wed, 21 Aug 2013 15:31:58 +1000
Subject: [Freeswitch-users] Video via a FreeSWITCH SBS
Message-ID: <CABkS=xpmK3G9gRd1Nce5tVeW1fteJUXeUAv3PueGVXsxxCoo7w@mail.gmail.com>

Is is possible to get video going through a FreeSWITCH SBS where one side
is using SRTP and the other RTP?

I suspect the answer is no, as it requires proxy_media=true to be set,
although I could be wrong.

When I try with proxy_media=true, I always get a:

415 Unsupported Media Type

Note that I have all the appropriate codec enabled on the FS servers and
the SIP clients.

Thanks

Peter
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From GB at cm.nl  Wed Aug 21 10:52:01 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Wed, 21 Aug 2013 08:52:01 +0200
Subject: [Freeswitch-users] List of $$ variables
In-Reply-To: <CAKzWOxWZGStzHyOVWPOao7YwdAc2GWDC0cxjEFuB=Ek76r56Lg@mail.gmail.com>
References: <FB7D97A214987F458242ACBDF87614073B1BD1995B@clubvirtual40.ClubMessage.local>
	<CAKzWOxWS73OZjjRpSU+wbKStFd541wTMz8kfC31O2OCAjh+7sQ@mail.gmail.com>
	<5213CB3C.7020109@xtronics.com>
	<FBABE432-943A-4461-841F-3C9F4396383C@jerris.com>
	<5213D253.7030906@xtronics.com>
	<CAKzWOxWZGStzHyOVWPOao7YwdAc2GWDC0cxjEFuB=Ek76r56Lg@mail.gmail.com>
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD199E1@clubvirtual40.ClubMessage.local>

Thanks everyone!

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, August 20, 2013 11:34 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] List of $$ variables

If you add those to your vars.xml, make a diff, and then throw it on a Jira then one of the committers will gladly add it.
-MC

On Tue, Aug 20, 2013 at 1:32 PM, Karl Schmidt <karl at xtronics.com<mailto:karl at xtronics.com>> wrote:
Thanks - this list probably should exist at the top of vars.xml as a comment

On 08/20/2013 03:10 PM, Michael Jerris wrote:
> hostname
> local_ip_v4
> local_mask_v4
> local_ip_v6
> switch_serial
> base_dir
> recordings_dir
> sound_prefix
> sounds_dir
> core_uuid
> zrtp_enabled
> nat_public_addr
> nat_private_addr
> nat_type

--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com<mailto:Karl at xtronics.com>
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089<tel:%28785%29%20841-3089>
Lawrence, KS 66049                              FAX (785) 841-0434<tel:%28785%29%20841-0434>
Opportunity is missed by most people because it is dressed in overalls and looks like work.
Thomas Alva Edison

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

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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



--
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From GB at cm.nl  Wed Aug 21 11:27:45 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Wed, 21 Aug 2013 09:27:45 +0200
Subject: [Freeswitch-users] Validating configuration freeswitch.xml.fsxml
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD199FB@clubvirtual40.ClubMessage.local>

Hello,

I've started to configure FS from scratch by making my own configuration files, based on the files in the vanilla directory.
I don't know if this is normal behavior but whenever I start FreeSwitch and check the freeswith.xml.fsxml file, the XML elements in the freeswitch/conf/customsbc/freeswitch.xml and vars.xml are not shown. Only the commentary is shown in this .fsxml file. Is this normal?

Also when running the netstat -unlp command, I see freeswitch isn't binding to the address defined in the bind_server_ip variable. Is this because the configuration isn't properly loaded?
Htop shows FS is running.
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From avi at avimarcus.net  Wed Aug 21 11:40:55 2013
From: avi at avimarcus.net (Avi Marcus)
Date: Wed, 21 Aug 2013 07:40:55 +0000
Subject: [Freeswitch-users] Validating configuration freeswitch.xml.fsxml
In-Reply-To: <FB7D97A214987F458242ACBDF87614073B1BD199FB@clubvirtual40.ClubMessage.local>
References: <FB7D97A214987F458242ACBDF87614073B1BD199FB@clubvirtual40.ClubMessage.local>
Message-ID: <000001409fd1b4ec-151071ba-8781-43ed-9b4a-6d24f306df44-000000@email.amazonses.com>

You need to make sure you include each file/directory you want. Only one
file is loaded by default and that specifies which other files to load. If
you take a look at the standard config files, you should see what I'm
talking about.
-Avi

On Wed, Aug 21, 2013 at 10:27 AM, Grant Bagdasarian <GB at cm.nl> wrote:

> Hello,****
>
> ** **
>
> I?ve started to configure FS from scratch by making my own configuration
> files, based on the files in the vanilla directory. ****
>
> I don?t know if this is normal behavior but whenever I start FreeSwitch
> and check the freeswith.xml.fsxml file, the XML elements in the
> freeswitch/conf/customsbc/freeswitch.xml and vars.xml are not shown. Only
> the commentary is shown in this .fsxml file. Is this normal?****
>
> ** **
>
> Also when running the netstat ?unlp command, I see freeswitch isn?t
> binding to the address defined in the bind_server_ip variable. Is this
> because the configuration isn?t properly loaded? ****
>
> Htop shows FS is running.
>
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From GB at cm.nl  Wed Aug 21 12:08:46 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Wed, 21 Aug 2013 10:08:46 +0200
Subject: [Freeswitch-users] Validating configuration freeswitch.xml.fsxml
In-Reply-To: <000001409fd1b4ec-151071ba-8781-43ed-9b4a-6d24f306df44-000000@email.amazonses.com>
References: <FB7D97A214987F458242ACBDF87614073B1BD199FB@clubvirtual40.ClubMessage.local>
	<000001409fd1b4ec-151071ba-8781-43ed-9b4a-6d24f306df44-000000@email.amazonses.com>
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD19A1D@clubvirtual40.ClubMessage.local>

Yes, I did that.

If I'm correct, then first /freeswitch/conf/freeswitch.xml is loaded, which point to the directory/file where the configuration files are located.
Which is: <X-PRE-PROCESS cmd="include" data="carriersbc/freeswitch.xml"/>

The carriersbc/freeswitch.xml looks like this:
<?xml version="1.0"?>
<document type="freeswitch/xml">
  <!--Load Global FreeSWITCH variables-->
  <X-PRE-PROCESS cmd="include" data="vars.xml" />

  <!--Load various configurations-->
  <section name="configuration" description="Various Configuration">
    <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml"/>
  </section>

  <!--Load dialplans for incoming and outgoing traffic-->
  <section name="dialplan" description="Regex/XML Dialplan">
    <X-PRE-PROCESS cmd="include" data="dialplan/*.xml"/>
  </section>
</document>

My vars.xml looks like this:

<include>
  <!--Preprocessor Variables-->

  <X-PRE-PROCESS cmd="set" data="domain=company.local"/>

  <!--Binding to a private and public IP-->
  <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>

  <!--We only need PCMA for now, which is G.711 Alaw-->
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA"/>

  <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.1"/>
  <X-PRE-PROCESS cmd="set" data="public_ip_v4=10.0.0.1"/>

  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=anonymous"/>
  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=anonymous"/>

  <!--Various debug and defaults-->
  <X-PRE-PROCESS cmd="set" data="call_debug=true"/>
  <X-PRE-PROCESS cmd="set" data="console_loglevel=debug"/>

  <!-- Internal SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=false"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_ip=192.168.1.1"/>
  <X-PRE-PROCESS cmd="set" data="internal_rtp_ip=192.168.1.1"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>

  <!--
  External SIP Profile - Carrier BT
  -->
  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=10.0.0.1"/>
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=10.0.0.1"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
</include>

I did load the vanilla/freeswitch.xml and checked the freeswitch.xml.fsxml file. The elements from freeswitch.xml and vars.xml are also not shown in this file, so I'm assuming this is normal behavior?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus
Sent: Wednesday, August 21, 2013 9:41 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Validating configuration freeswitch.xml.fsxml

You need to make sure you include each file/directory you want. Only one file is loaded by default and that specifies which other files to load. If you take a look at the standard config files, you should see what I'm talking about.
-Avi

On Wed, Aug 21, 2013 at 10:27 AM, Grant Bagdasarian <GB at cm.nl<mailto:GB at cm.nl>> wrote:
Hello,

I've started to configure FS from scratch by making my own configuration files, based on the files in the vanilla directory.
I don't know if this is normal behavior but whenever I start FreeSwitch and check the freeswith.xml.fsxml file, the XML elements in the freeswitch/conf/customsbc/freeswitch.xml and vars.xml are not shown. Only the commentary is shown in this .fsxml file. Is this normal?

Also when running the netstat -unlp command, I see freeswitch isn't binding to the address defined in the bind_server_ip variable. Is this because the configuration isn't properly loaded?
Htop shows FS is running.
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From GB at cm.nl  Wed Aug 21 12:32:24 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Wed, 21 Aug 2013 10:32:24 +0200
Subject: [Freeswitch-users] Validating configuration freeswitch.xml.fsxml
In-Reply-To: <FB7D97A214987F458242ACBDF87614073B1BD19A1D@clubvirtual40.ClubMessage.local>
References: <FB7D97A214987F458242ACBDF87614073B1BD199FB@clubvirtual40.ClubMessage.local>
	<000001409fd1b4ec-151071ba-8781-43ed-9b4a-6d24f306df44-000000@email.amazonses.com>
	<FB7D97A214987F458242ACBDF87614073B1BD19A1D@clubvirtual40.ClubMessage.local>
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD19A36@clubvirtual40.ClubMessage.local>

Problem solved. My internal and external sip profiles were binding to the wrong sip_ip variables in vars.xml.

Everything is working now.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian
Sent: Wednesday, August 21, 2013 10:09 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Validating configuration freeswitch.xml.fsxml

Yes, I did that.

If I'm correct, then first /freeswitch/conf/freeswitch.xml is loaded, which point to the directory/file where the configuration files are located.
Which is: <X-PRE-PROCESS cmd="include" data="carriersbc/freeswitch.xml"/>

The carriersbc/freeswitch.xml looks like this:
<?xml version="1.0"?>
<document type="freeswitch/xml">
  <!--Load Global FreeSWITCH variables-->
  <X-PRE-PROCESS cmd="include" data="vars.xml" />

  <!--Load various configurations-->
  <section name="configuration" description="Various Configuration">
    <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml"/>
  </section>

  <!--Load dialplans for incoming and outgoing traffic-->
  <section name="dialplan" description="Regex/XML Dialplan">
    <X-PRE-PROCESS cmd="include" data="dialplan/*.xml"/>
  </section>
</document>

My vars.xml looks like this:

<include>
  <!--Preprocessor Variables-->

  <X-PRE-PROCESS cmd="set" data="domain=company.local"/>

  <!--Binding to a private and public IP-->
  <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>

  <!--We only need PCMA for now, which is G.711 Alaw-->
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMA"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMA"/>

  <X-PRE-PROCESS cmd="set" data="local_ip_v4=192.168.1.1"/>
  <X-PRE-PROCESS cmd="set" data="public_ip_v4=10.0.0.1"/>

  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=anonymous"/>
  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=anonymous"/>

  <!--Various debug and defaults-->
  <X-PRE-PROCESS cmd="set" data="call_debug=true"/>
  <X-PRE-PROCESS cmd="set" data="console_loglevel=debug"/>

  <!-- Internal SIP Profile -->
  <X-PRE-PROCESS cmd="set" data="internal_auth_calls=false"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_ip=192.168.1.1"/>
  <X-PRE-PROCESS cmd="set" data="internal_rtp_ip=192.168.1.1"/>
  <X-PRE-PROCESS cmd="set" data="internal_sip_port=5060"/>

  <!--
  External SIP Profile - Carrier BT
  -->
  <X-PRE-PROCESS cmd="set" data="external_auth_calls=false"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_ip=10.0.0.1"/>
  <X-PRE-PROCESS cmd="set" data="external_rtp_ip=10.0.0.1"/>
  <X-PRE-PROCESS cmd="set" data="external_sip_port=5060"/>
</include>

I did load the vanilla/freeswitch.xml and checked the freeswitch.xml.fsxml file. The elements from freeswitch.xml and vars.xml are also not shown in this file, so I'm assuming this is normal behavior?

From: 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 Avi Marcus
Sent: Wednesday, August 21, 2013 9:41 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Validating configuration freeswitch.xml.fsxml

You need to make sure you include each file/directory you want. Only one file is loaded by default and that specifies which other files to load. If you take a look at the standard config files, you should see what I'm talking about.
-Avi

On Wed, Aug 21, 2013 at 10:27 AM, Grant Bagdasarian <GB at cm.nl<mailto:GB at cm.nl>> wrote:
Hello,

I've started to configure FS from scratch by making my own configuration files, based on the files in the vanilla directory.
I don't know if this is normal behavior but whenever I start FreeSwitch and check the freeswith.xml.fsxml file, the XML elements in the freeswitch/conf/customsbc/freeswitch.xml and vars.xml are not shown. Only the commentary is shown in this .fsxml file. Is this normal?

Also when running the netstat -unlp command, I see freeswitch isn't binding to the address defined in the bind_server_ip variable. Is this because the configuration isn't properly loaded?
Htop shows FS is running.
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From miha at softnet.si  Wed Aug 21 12:34:48 2013
From: miha at softnet.si (Miha)
Date: Wed, 21 Aug 2013 10:34:48 +0200
Subject: [Freeswitch-users] user_exists returns TRUE for non existing user
Message-ID: <52147BA8.2010208@softnet.si>

Hi,

I need a help regarding user_exists which return true for user, that is 
not in directory.

Is it possible to remove it from FS mem or where could be holding it?

Thx!
Miha


From andrew at cassidywebservices.co.uk  Wed Aug 21 13:08:13 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 21 Aug 2013 10:08:13 +0100
Subject: [Freeswitch-users] ACL Behaviour
Message-ID: <CAMbW9oJur+kuLM-2Q5hpkeDQ=eJOboA6EJGi2=Makfb1xbE41w@mail.gmail.com>

Hi all, I've just noticed a slightly unexpected (in my view) behavior when
checking ACLs.

If you have an empty ACL with default="allow" it will always return false.
Of course the workaround is to add a nonsense node into the list such as
0.0.0.0/32 (either allow or deny) then the ACL works as expected.

Now, before people shout FILE A JIRA at me, the reason I haven't is because
this behavior may be intended to prevent users accidentally leaving their
FreeSWITCH installations open.

So my question is, is this the intended behaviour? If so, I'll add it to
the wiki.

Thanks,

-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From miha at softnet.si  Wed Aug 21 13:33:42 2013
From: miha at softnet.si (Miha)
Date: Wed, 21 Aug 2013 11:33:42 +0200
Subject: [Freeswitch-users] user_exists returns TRUE for non existing
 user
In-Reply-To: <52147BA8.2010208@softnet.si>
References: <52147BA8.2010208@softnet.si>
Message-ID: <52148976.7020602@softnet.si>

I located the problem:)

I am doing user_exits id, it is true that user was not presented in 
directory, but group was created for this user if this id, so that is 
why it returns true:)

miha

Dne 8/21/2013 10:34 AM, pi?e Miha:
> Hi,
>
> I need a help regarding user_exists which return true for user, that is
> not in directory.
>
> Is it possible to remove it from FS mem or where could be holding it?
>
> Thx!
> Miha
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveayre at gmail.com  Wed Aug 21 14:08:17 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 21 Aug 2013 11:08:17 +0100
Subject: [Freeswitch-users] ACL Behaviour
In-Reply-To: <CAMbW9oJur+kuLM-2Q5hpkeDQ=eJOboA6EJGi2=Makfb1xbE41w@mail.gmail.com>
References: <CAMbW9oJur+kuLM-2Q5hpkeDQ=eJOboA6EJGi2=Makfb1xbE41w@mail.gmail.com>
Message-ID: <CAFiqYu=NUrp1A8hm+2QG7rT4HtQJXPGhuDayLw-gPr4O31Rm_A@mail.gmail.com>

Without shouting file a jira... if that is intended behaviour a jira ticket
could be commented and closed as such, and refered to from any duplicate
tickets later. Or corrected.



On Wednesday, August 21, 2013, Andrew Cassidy wrote:

> Hi all, I've just noticed a slightly unexpected (in my view) behavior when
> checking ACLs.
>
> If you have an empty ACL with default="allow" it will always return false.
> Of course the workaround is to add a nonsense node into the list such as
> 0.0.0.0/32 (either allow or deny) then the ACL works as expected.
>
> Now, before people shout FILE A JIRA at me, the reason I haven't is
> because this behavior may be intended to prevent users accidentally leaving
> their FreeSWITCH installations open.
>
> So my question is, is this the intended behaviour? If so, I'll add it to
> the wiki.
>
> Thanks,
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <javascript:_e({}, 'cvml', 'info at cassidywebservices.co.uk');> *03300
> 100 960  *F <javascript:_e({}, 'cvml', 'info at cassidywebservices.co.uk');>
> *03300 100 961
> *E <javascript:_e({}, 'cvml', 'info at cassidywebservices.co.uk');> *
> andrew at cassidywebservices.co.uk <javascript:_e({}, 'cvml',
> 'andrew at cassidywebservices.co.uk');>
> *W <javascript:_e({}, 'cvml', 'info at cassidywebservices.co.uk');> *
> www.cassidywebservices.co.uk
>
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From steveayre at gmail.com  Wed Aug 21 14:17:34 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 21 Aug 2013 11:17:34 +0100
Subject: [Freeswitch-users] in/outbound-use-uuid-as-callid as channel
	variables
In-Reply-To: <CACYJG3J93Ge1Qca+=jacNfP57eWPTKu7kFaovVgkSSbgcbEcEA@mail.gmail.com>
References: <CACYJG3J93Ge1Qca+=jacNfP57eWPTKu7kFaovVgkSSbgcbEcEA@mail.gmail.com>
Message-ID: <CAFiqYum+93poAaGpvqO3MfzkYgf49tityWqQd2hcEUkNQF4o0w@mail.gmail.com>

It would be too late for inbound in the dialplan as the channel already
exists, therefore already has a uuid. Though you can change the uuid, you'd
still have issues whenever a call hung up so quickly it never reached the
dialplan (yes, I've seen that happen).

Outbound you may have a chance with - but I don't know if such a variable
exists.

Is there a reason you can't just set it on the profile?



On Tuesday, August 20, 2013, Jeff Pyle wrote:

> Hello,
>
> Is it possible to set the equivalent of inbound-use-uuid-as-callid
> or outbound-use-uuid-as-callid per channel from within the dialplan?
>
>
> - Jeff
>
>
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From andrew at cassidywebservices.co.uk  Wed Aug 21 15:08:59 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 21 Aug 2013 12:08:59 +0100
Subject: [Freeswitch-users] ACL Behaviour
In-Reply-To: <CAFiqYu=NUrp1A8hm+2QG7rT4HtQJXPGhuDayLw-gPr4O31Rm_A@mail.gmail.com>
References: <CAMbW9oJur+kuLM-2Q5hpkeDQ=eJOboA6EJGi2=Makfb1xbE41w@mail.gmail.com>
	<CAFiqYu=NUrp1A8hm+2QG7rT4HtQJXPGhuDayLw-gPr4O31Rm_A@mail.gmail.com>
Message-ID: <CAMbW9oJTtT1aOOBbSGwC9_Zzd1xk8=FWrre8cAqBv=bHg0GRsQ@mail.gmail.com>

Good call.


On 21 August 2013 11:08, Steven Ayre <steveayre at gmail.com> wrote:

> Without shouting file a jira... if that is intended behaviour a jira
> ticket could be commented and closed as such, and refered to from any
> duplicate tickets later. Or corrected.
>
>
>
> On Wednesday, August 21, 2013, Andrew Cassidy wrote:
>
>> Hi all, I've just noticed a slightly unexpected (in my view) behavior
>> when checking ACLs.
>>
>> If you have an empty ACL with default="allow" it will always return
>> false. Of course the workaround is to add a nonsense node into the list
>> such as 0.0.0.0/32 (either allow or deny) then the ACL works as expected.
>>
>> Now, before people shout FILE A JIRA at me, the reason I haven't is
>> because this behavior may be intended to prevent users accidentally leaving
>> their FreeSWITCH installations open.
>>
>> So my question is, is this the intended behaviour? If so, I'll add it to
>> the wiki.
>>
>> Thanks,
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>>
>> *T *03300 100 960  *F *03300 100 961
>> *E *andrew at cassidywebservices.co.uk
>> *W *www.cassidywebservices.co.uk
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From GB at cm.nl  Wed Aug 21 15:49:31 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Wed, 21 Aug 2013 13:49:31 +0200
Subject: [Freeswitch-users] Matching context based on sip_req_host
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD19AE0@clubvirtual40.ClubMessage.local>

Hello,

I want to match a context based on the value of sip_req_host, but the context isn't matched.

    <extension name="internal_to_external">
      <condition field="sip_req_host" expression="^10\.0\.0\.1$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="sofia/external/${sip_req_uri}"/>
      </condition>
    </extension>

Debug output:
Dialplan: sofia/internal/31765727000 at 192.168.18.6 Regex (FAIL) [internal_to_external] sip_req_host() =~ /^62\.180\.237\.73$/ break=on-false

I checked with an online regex validator and it seems that the first and last '/' character is causing it to not match. I'm not a regex expert, so I would appreciate it if someone could help me out with this.

Thanks!
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From GB at cm.nl  Wed Aug 21 16:01:55 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Wed, 21 Aug 2013 14:01:55 +0200
Subject: [Freeswitch-users] Matching context based on sip_req_host
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD19AEC@clubvirtual40.ClubMessage.local>

The following line: <condition field="sip_req_host" expression="^10\.0\.0\.1$">
Should be  <condition field="sip_req_host" expression="^62\.180\.237\.73$">

Wrong sample.

From: Grant Bagdasarian
Sent: Wednesday, August 21, 2013 1:50 PM
To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)
Subject: Matching context based on sip_req_host

Hello,

I want to match a context based on the value of sip_req_host, but the context isn't matched.

    <extension name="internal_to_external">
      <condition field="sip_req_host" expression="^10\.0\.0\.1$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="sofia/external/${sip_req_uri}"/>
      </condition>
    </extension>

Debug output:
Dialplan: sofia/internal/31765727000 at 192.168.18.6<mailto:sofia/internal/31765727000 at 192.168.18.6> Regex (FAIL) [internal_to_external] sip_req_host() =~ /^62\.180\.237\.73$/ break=on-false

I checked with an online regex validator and it seems that the first and last '/' character is causing it to not match. I'm not a regex expert, so I would appreciate it if someone could help me out with this.

Thanks!
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From jpyle at fidelityvoice.com  Wed Aug 21 16:21:27 2013
From: jpyle at fidelityvoice.com (Jeff Pyle)
Date: Wed, 21 Aug 2013 08:21:27 -0400
Subject: [Freeswitch-users] in/outbound-use-uuid-as-callid as channel
	variables
In-Reply-To: <CAFiqYum+93poAaGpvqO3MfzkYgf49tityWqQd2hcEUkNQF4o0w@mail.gmail.com>
References: <CACYJG3J93Ge1Qca+=jacNfP57eWPTKu7kFaovVgkSSbgcbEcEA@mail.gmail.com>
	<CAFiqYum+93poAaGpvqO3MfzkYgf49tityWqQd2hcEUkNQF4o0w@mail.gmail.com>
Message-ID: <CACYJG3LeF8s9cPTDDbGzEAf3jnfi_s=W5Ns=KOsV4uP1bt=uqA@mail.gmail.com>

I was afraid of that on the inbound side.  It makes sense.

The short answer is sometimes I want full topology-hiding, other times I
want the same SIP callid all the way through.  Managing it per-profile
could work.  It has to do with correlating voice-quality monitoring on the
private side with other measurements from the public side.  In cases where
this happens, the same callid on both sides makes things much, much easier.

Yesterday I tried enabling both inbound-use-callid-as-uuid and
outbound-use-uuid-as-callid on both the public and private profiles.  They
didn't seem to have any effect - I still saw a new SIP callid on the b-leg
INVITE.  I wasn't sure how to debug it from there.


- Jeff


On Wed, Aug 21, 2013 at 6:17 AM, Steven Ayre <steveayre at gmail.com> wrote:

> It would be too late for inbound in the dialplan as the channel already
> exists, therefore already has a uuid. Though you can change the uuid, you'd
> still have issues whenever a call hung up so quickly it never reached the
> dialplan (yes, I've seen that happen).
>
> Outbound you may have a chance with - but I don't know if such a variable
> exists.
>
> Is there a reason you can't just set it on the profile?
>
>
>
> On Tuesday, August 20, 2013, Jeff Pyle wrote:
>
>> Hello,
>>
>> Is it possible to set the equivalent of inbound-use-uuid-as-callid
>> or outbound-use-uuid-as-callid per channel from within the dialplan?
>>
>>
>> - Jeff
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 peter at olssononline.se  Wed Aug 21 16:25:44 2013
From: peter at olssononline.se (Peter Olsson)
Date: Wed, 21 Aug 2013 14:25:44 +0200
Subject: [Freeswitch-users] Matching context based on sip_req_host
In-Reply-To: <FB7D97A214987F458242ACBDF87614073B1BD19AEC@clubvirtual40.ClubMessage.local>
References: <FB7D97A214987F458242ACBDF87614073B1BD19AEC@clubvirtual40.ClubMessage.local>
Message-ID: <CAPEdYTgNTQ5=4etmRKa1=oaaTTM2h1MyQ8jRVg1zVsGHkBS6QQ@mail.gmail.com>

That should probably be:  <condition field="${sip_req_host}"
expression="^62\.180\.237\.73$">

/Peter


2013/8/21 Grant Bagdasarian <GB at cm.nl>

> The following line: <condition field="sip_req_host"
> expression="^10\.0\.0\.1$">****
>
> Should be  <condition field="sip_req_host" expression="^62\.180\.237\.73$">
> ****
>
> ** **
>
> Wrong sample.****
>
> ** **
>
> *From:* Grant Bagdasarian
> *Sent:* Wednesday, August 21, 2013 1:50 PM
> *To:* FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)
> *Subject:* Matching context based on sip_req_host****
>
> ** **
>
> Hello,****
>
> ** **
>
> I want to match a context based on the value of sip_req_host, but the
> context isn?t matched.****
>
> ** **
>
>     <extension name="internal_to_external">****
>
>       <condition field="sip_req_host" expression="^10\.0\.0\.1$">****
>
>         <action application="set" data="hangup_after_bridge=false"/>****
>
>         <action application="set" data="continue_on_fail=true"/>****
>
>         <action application="bridge" data="sofia/external/${sip_req_uri}"/>
> ****
>
>       </condition>****
>
>     </extension>****
>
> ** **
>
> Debug output:****
>
> Dialplan: sofia/internal/31765727000 at 192.168.18.6 Regex (FAIL)
> [internal_to_external] sip_req_host() =~ /^62\.180\.237\.73$/ break=on-false
> ****
>
> ** **
>
> I checked with an online regex validator and it seems that the first and
> last ?/? character is causing it to not match. I?m not a regex expert, so I
> would appreciate it if someone could help me out with this.****
>
> ** **
>
> Thanks!****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Wed Aug 21 16:40:04 2013
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 21 Aug 2013 07:40:04 -0500
Subject: [Freeswitch-users] Matching context based on sip_req_host
In-Reply-To: <CAPEdYTgNTQ5=4etmRKa1=oaaTTM2h1MyQ8jRVg1zVsGHkBS6QQ@mail.gmail.com>
Message-ID: <CE3A1F54.791D9%krice@freeswitch.org>

If you are trying to match on the IP of the client, you probably really want
network_addr which reports from the ip stack which ip the invite came from,
using any other IP unless it is coming from a 100% trusted source leaves you
vulnerable to people spoofing the IPs inside the sip packets.

Also, the context on unauthenticated users goes with what is defined on the
sip profile, so you may need a sorting context


On 8/21/13 7:25 AM, "Peter Olsson" <peter at olssononline.se> wrote:

> That should probably be:??<condition field="${sip_req_host}"
> expression="^62\.180\.237\.73$">
> 
> /Peter
> 
> 
> 2013/8/21 Grant Bagdasarian <GB at cm.nl>
>> The following line: <condition field="sip_req_host"
>> expression="^10\.0\.0\.1$">
>> Should be ?<condition field="sip_req_host" expression="^62\.180\.237\.73$">
>> ?
>> Wrong sample.
>> ?
>> 
>> From: Grant Bagdasarian
>> Sent: Wednesday, August 21, 2013 1:50 PM
>> To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)
>> Subject: Matching context based on sip_req_host
>> ?
>> Hello,
>> ?
>> I want to match a context based on the value of sip_req_host, but the context
>> isn?t matched.
>> ?
>> ??? <extension name="internal_to_external">
>> ????? <condition field="sip_req_host" expression="^10\.0\.0\.1$">
>> ??????? <action application="set" data="hangup_after_bridge=false"/>
>> ??????? <action application="set" data="continue_on_fail=true"/>
>> ??????? <action application="bridge" data="sofia/external/${sip_req_uri}"/>
>> ????? </condition>
>> ??? </extension>
>> ?
>> Debug output:
>> Dialplan: sofia/internal/31765727000 at 192.168.18.6 Regex (FAIL)
>> [internal_to_external] sip_req_host() =~ /^62\.180\.237\.73$/ break=on-false
>> ?
>> I checked with an online regex validator and it seems that the first and last
>> ?/? character is causing it to not match. I?m not a regex expert, so I would
>> appreciate it if someone could help me out with this.
>> ?
>> Thanks!
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>> 
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From GB at cm.nl  Wed Aug 21 17:09:19 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Wed, 21 Aug 2013 15:09:19 +0200
Subject: [Freeswitch-users] Matching context based on sip_req_host
In-Reply-To: <CE3A1F54.791D9%krice@freeswitch.org>
References: <CAPEdYTgNTQ5=4etmRKa1=oaaTTM2h1MyQ8jRVg1zVsGHkBS6QQ@mail.gmail.com>
	<CE3A1F54.791D9%krice@freeswitch.org>
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD19B0B@clubvirtual40.ClubMessage.local>

Right, I forgot the ${...}. That did the trick. Thanks Peter!

I'll keep that in mind Ken. I'm not matching the IP of the client. The R-URI is set by our SIP Proxy, and this FS instance functions as our SBC which is connected to multiple carriers.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice
Sent: Wednesday, August 21, 2013 2:40 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Matching context based on sip_req_host

If you are trying to match on the IP of the client, you probably really want network_addr which reports from the ip stack which ip the invite came from, using any other IP unless it is coming from a 100% trusted source leaves you vulnerable to people spoofing the IPs inside the sip packets.

Also, the context on unauthenticated users goes with what is defined on the sip profile, so you may need a sorting context


On 8/21/13 7:25 AM, "Peter Olsson" <peter at olssononline.se> wrote:
That should probably be:  <condition field="${sip_req_host}" expression="^62\.180\.237\.73$">

/Peter


2013/8/21 Grant Bagdasarian <GB at cm.nl>
The following line: <condition field="sip_req_host" expression="^10\.0\.0\.1$">
Should be  <condition field="sip_req_host" expression="^62\.180\.237\.73$">

Wrong sample.


From: Grant Bagdasarian
Sent: Wednesday, August 21, 2013 1:50 PM
To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)
Subject: Matching context based on sip_req_host

Hello,

I want to match a context based on the value of sip_req_host, but the context isn't matched.

    <extension name="internal_to_external">
      <condition field="sip_req_host" expression="^10\.0\.0\.1$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="sofia/external/${sip_req_uri}"/>
      </condition>
    </extension>

Debug output:
Dialplan: sofia/internal/31765727000 at 192.168.18.6 Regex (FAIL) [internal_to_external] sip_req_host() =~ /^62\.180\.237\.73$/ break=on-false

I checked with an online regex validator and it seems that the first and last '/' character is causing it to not match. I'm not a regex expert, so I would appreciate it if someone could help me out with this.

Thanks!

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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

________________________________
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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

--
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch
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From fdelawarde at wirelessmundi.com  Wed Aug 21 17:29:12 2013
From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?=)
Date: Wed, 21 Aug 2013 15:29:12 +0200
Subject: [Freeswitch-users] Matching context based on sip_req_host
In-Reply-To: <FB7D97A214987F458242ACBDF87614073B1BD19AEC@clubvirtual40.ClubMessage.local>
References: <FB7D97A214987F458242ACBDF87614073B1BD19AEC@clubvirtual40.ClubMessage.local>
Message-ID: <1377091752.26874.3.camel@luna.madrid.commsmundi.com>

Try with ${sip_reg_host} in field.

Fran?ois.

On Wed, 2013-08-21 at 14:01 +0200, Grant Bagdasarian wrote: 
> The following line: <condition field="sip_req_host" expression="^10\.0
> \.0\.1$">
> 
> Should be  <condition field="sip_req_host" expression="^62\.180\.237
> \.73$">
> 
>  
> 
> Wrong sample.
> 
>  
> 
> From: Grant Bagdasarian 
> Sent: Wednesday, August 21, 2013 1:50 PM
> To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)
> Subject: Matching context based on sip_req_host
> 
> 
>  
> 
> Hello,
> 
>  
> 
> I want to match a context based on the value of sip_req_host, but the
> context isn?t matched.
> 
>  
> 
>     <extension name="internal_to_external">
> 
>       <condition field="sip_req_host" expression="^10\.0\.0\.1$">
> 
>         <action application="set" data="hangup_after_bridge=false"/>
> 
>         <action application="set" data="continue_on_fail=true"/>
> 
>         <action application="bridge"
> data="sofia/external/${sip_req_uri}"/>
> 
>       </condition>
> 
>     </extension>
> 
>  
> 
> Debug output:
> 
> Dialplan: sofia/internal/31765727000 at 192.168.18.6 Regex (FAIL)
> [internal_to_external] sip_req_host() =~ /^62\.180\.237\.73$/
> break=on-false
> 
>  
> 
> I checked with an online regex validator and it seems that the first
> and last ?/? character is causing it to not match. I?m not a regex
> expert, so I would appreciate it if someone could help me out with
> this.
> 
>  
> 
> Thanks!
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 covici at ccs.covici.com  Wed Aug 21 17:35:29 2013
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Wed, 21 Aug 2013 09:35:29 -0400
Subject: [Freeswitch-users] Matching context based on sip_req_host
In-Reply-To: <FB7D97A214987F458242ACBDF87614073B1BD19AE0@clubvirtual40.ClubMessage.local>
References: <FB7D97A214987F458242ACBDF87614073B1BD19AE0@clubvirtual40.ClubMessage.local>
Message-ID: <29905.1377092129@ccs.covici.com>

I am not sure about the /, but you might try enclosing the field name in
${} in addition to the quotes.

Grant Bagdasarian <GB at cm.nl> wrote:

> Hello,
> 
> I want to match a context based on the value of sip_req_host, but the context isn't matched.
> 
>     <extension name="internal_to_external">
>       <condition field="sip_req_host" expression="^10\.0\.0\.1$">
>         <action application="set" data="hangup_after_bridge=false"/>
>         <action application="set" data="continue_on_fail=true"/>
>         <action application="bridge" data="sofia/external/${sip_req_uri}"/>
>       </condition>
>     </extension>
> 
> Debug output:
> Dialplan: sofia/internal/31765727000 at 192.168.18.6 Regex (FAIL) [internal_to_external] sip_req_host() =~ /^62\.180\.237\.73$/ break=on-false
> 
> I checked with an online regex validator and it seems that the first and last '/' character is causing it to not match. I'm not a regex expert, so I would appreciate it if someone could help me out with this.
> 
> Thanks!
> 
> ----------------------------------------------------
> Alternatives:
> 
> ----------------------------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
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 mike at jerris.com  Wed Aug 21 17:37:23 2013
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Aug 2013 09:37:23 -0400
Subject: [Freeswitch-users] Video via a FreeSWITCH SBS
In-Reply-To: <CABkS=xpmK3G9gRd1Nce5tVeW1fteJUXeUAv3PueGVXsxxCoo7w@mail.gmail.com>
References: <CABkS=xpmK3G9gRd1Nce5tVeW1fteJUXeUAv3PueGVXsxxCoo7w@mail.gmail.com>
Message-ID: <70DC1387-6599-40EA-A576-127237ECC02D@jerris.com>

It certainly won't work with proxy_media=true, but why would you need to use proxy_media=true?

On Aug 21, 2013, at 1:31 AM, Peter <eidevm5 at gmail.com> wrote:

> Is is possible to get video going through a FreeSWITCH SBS where one side is using SRTP and the other RTP?
> 
> I suspect the answer is no, as it requires proxy_media=true to be set, although I could be wrong.
> 
> When I try with proxy_media=true, I always get a:
> 
> 415 Unsupported Media Type
> 
> Note that I have all the appropriate codec enabled on the FS servers and the SIP clients.



From pankajanand18 at gmail.com  Wed Aug 21 11:04:11 2013
From: pankajanand18 at gmail.com (pankajanand18 at gmail.com)
Date: Wed, 21 Aug 2013 12:34:11 +0530
Subject: [Freeswitch-users] can't load codec
Message-ID: <CAJhzRKcrQ6u2Pgv+4R3A3pfpNPfVH0qLBQm-s+W=Td2cj9w6iQ@mail.gmail.com>

2013-08-20 19:14:58.820110 [WARNING] switch_core_codec.c:699 Codec PROXY
Exists
but not at the desired implementation. 90000hz 0ms
2013-08-20 19:14:58.820110 [ERR] switch_core_media.c:1837 Can't load codec?

Getting this error with following settings:

1. media proxy mode enabled
2. late negotiation enabled
3. <X-PRE-PROCESS cmd="set"
data="global_codec_prefs=G722,PCMU,PCMA,GSM,H264"/>


Both sip clients are are h264 capable. The same configuration work fine on
freeswitch running over ubuntu 12.04 but generates the above log and error
on window 7 x64.

I found that the freeswitch version I was using on linux was 1.2 and on
windows it was 1.5 with git 39ad799
On Linux I checked out same revision and built it again. I could reproduce
the issue on linux too. but in the latest revision the problem seems to be
fixed. So now I am going to build the the current revision code for windows.

Any comment on this or have anybody faced this issue ?

Cheers,
Pankaj
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From mike at jerris.com  Wed Aug 21 17:41:23 2013
From: mike at jerris.com (Michael Jerris)
Date: Wed, 21 Aug 2013 09:41:23 -0400
Subject: [Freeswitch-users] ACL Behaviour
In-Reply-To: <CAFiqYu=NUrp1A8hm+2QG7rT4HtQJXPGhuDayLw-gPr4O31Rm_A@mail.gmail.com>
References: <CAMbW9oJur+kuLM-2Q5hpkeDQ=eJOboA6EJGi2=Makfb1xbE41w@mail.gmail.com>
	<CAFiqYu=NUrp1A8hm+2QG7rT4HtQJXPGhuDayLw-gPr4O31Rm_A@mail.gmail.com>
Message-ID: <BF98DFD5-4EDB-4208-8A27-134FCD55661F@jerris.com>

NO NEED TO SHOUT OVER IT!!!

On Aug 21, 2013, at 6:08 AM, Steven Ayre <steveayre at gmail.com> wrote:

> Without shouting file a jira... if that is intended behaviour a jira ticket could be commented and closed as such, and refered to from any duplicate tickets later. Or corrected.
> 
> 
> 
> On Wednesday, August 21, 2013, Andrew Cassidy wrote:
> Hi all, I've just noticed a slightly unexpected (in my view) behavior when checking ACLs.
> 
> If you have an empty ACL with default="allow" it will always return false. Of course the workaround is to add a nonsense node into the list such as 0.0.0.0/32 (either allow or deny) then the ACL works as expected.
> 
> Now, before people shout FILE A JIRA at me, the reason I haven't is because this behavior may be intended to prevent users accidentally leaving their FreeSWITCH installations open.
> 
> So my question is, is this the intended behaviour? If so, I'll add it to the wiki.
> 
> Thanks,
> 
> -- 
> Andrew Cassidy BSc (Hons) MBCS SSCA
> Managing Director
> 

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From steveayre at gmail.com  Wed Aug 21 17:53:10 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 21 Aug 2013 14:53:10 +0100
Subject: [Freeswitch-users] in/outbound-use-uuid-as-callid as channel
	variables
In-Reply-To: <CACYJG3LeF8s9cPTDDbGzEAf3jnfi_s=W5Ns=KOsV4uP1bt=uqA@mail.gmail.com>
References: <CACYJG3J93Ge1Qca+=jacNfP57eWPTKu7kFaovVgkSSbgcbEcEA@mail.gmail.com>
	<CAFiqYum+93poAaGpvqO3MfzkYgf49tityWqQd2hcEUkNQF4o0w@mail.gmail.com>
	<CACYJG3LeF8s9cPTDDbGzEAf3jnfi_s=W5Ns=KOsV4uP1bt=uqA@mail.gmail.com>
Message-ID: <CAFiqYumfRbnwe6EEBL8m_VwBw1U-=GUZO-Cx13yV6QBb2Nyaqw@mail.gmail.com>

>
> They didn't seem to have any effect - I still saw a new SIP callid on the
> b-leg INVITE


Perhaps you're misunderstanding the purpose of them.

Every channel in FreeSWITCH has a UUID. On a bridge you have two different
channels (inbound & outbound). Their UUIDs -must- be different, to tell
them apart.

The setting you found sets the UUID from the Call-ID on the inbound
channel, and visa versa for the outbound channel. That mostly makes it
easier to match up packet traces to calls, and your CDRs to a customers
(since they know the Call-ID but not your internal UUID).

If you set both the UUID of the incoming channel will be the incoming
Call-ID, the outgoing UUID will be generated by FreeSWITCH and the outgoing
Call-ID will match the outgoing UUID. You cannot set the same UUID for both.

Have you tried exporting the sip_call_id variable from the aleg to the
bleg? I don't know if that variable is read-only though. If sofia will read
from that should be sufficient, you won't need to tweak the UUIDs of the
channels.

If that doesn't work you could also try setting origination_uuid to
sip_call_id in the bridge dialstring, along with the outgoing parameter you
found.

-Steve



On 21 August 2013 13:21, Jeff Pyle <jpyle at fidelityvoice.com> wrote:

> I was afraid of that on the inbound side.  It makes sense.
>
> The short answer is sometimes I want full topology-hiding, other times I
> want the same SIP callid all the way through.  Managing it per-profile
> could work.  It has to do with correlating voice-quality monitoring on the
> private side with other measurements from the public side.  In cases where
> this happens, the same callid on both sides makes things much, much easier.
>
> Yesterday I tried enabling both inbound-use-callid-as-uuid and
> outbound-use-uuid-as-callid on both the public and private profiles.  They
> didn't seem to have any effect - I still saw a new SIP callid on the b-leg
> INVITE.  I wasn't sure how to debug it from there.
>
>
> - Jeff
>
>
> On Wed, Aug 21, 2013 at 6:17 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> It would be too late for inbound in the dialplan as the channel already
>> exists, therefore already has a uuid. Though you can change the uuid, you'd
>> still have issues whenever a call hung up so quickly it never reached the
>> dialplan (yes, I've seen that happen).
>>
>> Outbound you may have a chance with - but I don't know if such a variable
>> exists.
>>
>> Is there a reason you can't just set it on the profile?
>>
>>
>>
>> On Tuesday, August 20, 2013, Jeff Pyle wrote:
>>
>>>  Hello,
>>>
>>> Is it possible to set the equivalent of inbound-use-uuid-as-callid
>>> or outbound-use-uuid-as-callid per channel from within the dialplan?
>>>
>>>
>>> - Jeff
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 at cassidywebservices.co.uk  Wed Aug 21 18:00:12 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 21 Aug 2013 15:00:12 +0100
Subject: [Freeswitch-users] ACL Behaviour
In-Reply-To: <BF98DFD5-4EDB-4208-8A27-134FCD55661F@jerris.com>
References: <CAMbW9oJur+kuLM-2Q5hpkeDQ=eJOboA6EJGi2=Makfb1xbE41w@mail.gmail.com>
	<CAFiqYu=NUrp1A8hm+2QG7rT4HtQJXPGhuDayLw-gPr4O31Rm_A@mail.gmail.com>
	<BF98DFD5-4EDB-4208-8A27-134FCD55661F@jerris.com>
Message-ID: <CAMbW9oJeoPTzbiWoqtCUL5eX37EbkfycfTxyJBBZ8FktFHXU0w@mail.gmail.com>

No, but I remember about a week long period where the reply to nearly every
emails was 'FILE A JIRA' and it was starting to get tedious. I think there
were even meme images created :)


On 21 August 2013 14:41, Michael Jerris <mike at jerris.com> wrote:

> NO NEED TO SHOUT OVER IT!!!
>
> On Aug 21, 2013, at 6:08 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
> Without shouting file a jira... if that is intended behaviour a jira
> ticket could be commented and closed as such, and refered to from any
> duplicate tickets later. Or corrected.
>
>
>
> On Wednesday, August 21, 2013, Andrew Cassidy wrote:
>
>> Hi all, I've just noticed a slightly unexpected (in my view) behavior
>> when checking ACLs.
>>
>> If you have an empty ACL with default="allow" it will always return
>> false. Of course the workaround is to add a nonsense node into the list
>> such as 0.0.0.0/32 (either allow or deny) then the ACL works as expected.
>>
>> Now, before people shout FILE A JIRA at me, the reason I haven't is
>> because this behavior may be intended to prevent users accidentally leaving
>> their FreeSWITCH installations open.
>>
>> So my question is, is this the intended behaviour? If so, I'll add it to
>> the wiki.
>>
>> Thanks,
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From andrew at cassidywebservices.co.uk  Wed Aug 21 18:00:55 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 21 Aug 2013 15:00:55 +0100
Subject: [Freeswitch-users] ACL Behaviour
In-Reply-To: <CAMbW9oJeoPTzbiWoqtCUL5eX37EbkfycfTxyJBBZ8FktFHXU0w@mail.gmail.com>
References: <CAMbW9oJur+kuLM-2Q5hpkeDQ=eJOboA6EJGi2=Makfb1xbE41w@mail.gmail.com>
	<CAFiqYu=NUrp1A8hm+2QG7rT4HtQJXPGhuDayLw-gPr4O31Rm_A@mail.gmail.com>
	<BF98DFD5-4EDB-4208-8A27-134FCD55661F@jerris.com>
	<CAMbW9oJeoPTzbiWoqtCUL5eX37EbkfycfTxyJBBZ8FktFHXU0w@mail.gmail.com>
Message-ID: <CAMbW9oJK87i7fEBYYJgBT6=PydUu5xc_A_uBG37sOOfq6yMhMw@mail.gmail.com>

This one, in fact: http://wiki.freeswitch.org/wiki/File:Yunfj.jpeg


On 21 August 2013 15:00, Andrew Cassidy <andrew at cassidywebservices.co.uk>wrote:

> No, but I remember about a week long period where the reply to nearly
> every emails was 'FILE A JIRA' and it was starting to get tedious. I think
> there were even meme images created :)
>
>
> On 21 August 2013 14:41, Michael Jerris <mike at jerris.com> wrote:
>
>> NO NEED TO SHOUT OVER IT!!!
>>
>> On Aug 21, 2013, at 6:08 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Without shouting file a jira... if that is intended behaviour a jira
>> ticket could be commented and closed as such, and refered to from any
>> duplicate tickets later. Or corrected.
>>
>>
>>
>> On Wednesday, August 21, 2013, Andrew Cassidy wrote:
>>
>>> Hi all, I've just noticed a slightly unexpected (in my view) behavior
>>> when checking ACLs.
>>>
>>> If you have an empty ACL with default="allow" it will always return
>>> false. Of course the workaround is to add a nonsense node into the list
>>> such as 0.0.0.0/32 (either allow or deny) then the ACL works as
>>> expected.
>>>
>>> Now, before people shout FILE A JIRA at me, the reason I haven't is
>>> because this behavior may be intended to prevent users accidentally leaving
>>> their FreeSWITCH installations open.
>>>
>>> So my question is, is this the intended behaviour? If so, I'll add it to
>>> the wiki.
>>>
>>> Thanks,
>>>
>>> --
>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>> Managing Director
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From lists at kavun.ch  Wed Aug 21 18:12:54 2013
From: lists at kavun.ch (Emrah)
Date: Wed, 21 Aug 2013 16:12:54 +0200
Subject: [Freeswitch-users] Cannot load mod_spandsp.so: undefined symbol:
	lzma_stream_encoder
Message-ID: <8E465A5B-2B1E-42B1-A6C5-A445AAB2EA8A@kavun.ch>

Hi guys,

I am having an issue getting mod_spandsp to work on a fresh install.
Running Debian 7, error message as follows:

ERR [module load file routine returned an error]
2013-08-21 14:11:20.773383 [CRIT] switch_loadable_module.c:1383 Error Loading module /usr/local/freeswitch/mod/mod_spandsp.so
**/usr/local/freeswitch/mod/mod_spandsp.so: undefined symbol: lzma_stream_encoder**

It seems like a similar issue was reported here:
http://jira.freeswitch.org/browse/FS-4679?page=com.atlassian.streams.streams-jira-plugin:activity-stream-issue-tab

Any help would be greatly appreciated.

Emrah

From fs.user at fordior.net  Wed Aug 21 18:37:38 2013
From: fs.user at fordior.net (EL)
Date: Wed, 21 Aug 2013 16:37:38 +0200
Subject: [Freeswitch-users] ACL Behaviour
In-Reply-To: <CAMbW9oJK87i7fEBYYJgBT6=PydUu5xc_A_uBG37sOOfq6yMhMw@mail.gmail.com>
References: <CAMbW9oJur+kuLM-2Q5hpkeDQ=eJOboA6EJGi2=Makfb1xbE41w@mail.gmail.com>
	<CAFiqYu=NUrp1A8hm+2QG7rT4HtQJXPGhuDayLw-gPr4O31Rm_A@mail.gmail.com>
	<BF98DFD5-4EDB-4208-8A27-134FCD55661F@jerris.com>
	<CAMbW9oJeoPTzbiWoqtCUL5eX37EbkfycfTxyJBBZ8FktFHXU0w@mail.gmail.com>
	<CAMbW9oJK87i7fEBYYJgBT6=PydUu5xc_A_uBG37sOOfq6yMhMw@mail.gmail.com>
Message-ID: <20130821143738.GD1746@0rdior.com>


> This one, in fact: http://wiki.freeswitch.org/wiki/File:Yunfj.jpeg

[offtopic mode]

May be coincidental, but the eyes and face impression look a bit like:
bit.ly/19xkxzY

-- 
EL


From steveu at coppice.org  Wed Aug 21 18:39:14 2013
From: steveu at coppice.org (Steve Underwood)
Date: Wed, 21 Aug 2013 22:39:14 +0800
Subject: [Freeswitch-users] Cannot load mod_spandsp.so: undefined
 symbol: lzma_stream_encoder
In-Reply-To: <8E465A5B-2B1E-42B1-A6C5-A445AAB2EA8A@kavun.ch>
References: <8E465A5B-2B1E-42B1-A6C5-A445AAB2EA8A@kavun.ch>
Message-ID: <5214D112.9070607@coppice.org>

On 08/21/2013 10:12 PM, Emrah wrote:
> Hi guys,
>
> I am having an issue getting mod_spandsp to work on a fresh install.
> Running Debian 7, error message as follows:
>
> ERR [module load file routine returned an error]
> 2013-08-21 14:11:20.773383 [CRIT] switch_loadable_module.c:1383 Error Loading module /usr/local/freeswitch/mod/mod_spandsp.so
> **/usr/local/freeswitch/mod/mod_spandsp.so: undefined symbol: lzma_stream_encoder**
>
> It seems like a similar issue was reported here:
> http://jira.freeswitch.org/browse/FS-4679?page=com.atlassian.streams.streams-jira-plugin:activity-stream-issue-tab
>
> Any help would be greatly appreciated.
>
> Emrah
>
This means liblzma cannot be found at run time, although FS should not 
have linked if this was missing at compile time. Do you have liblzma in 
a directory that the linker searches but the loader does not?


From anthony.minessale at gmail.com  Wed Aug 21 18:39:48 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 21 Aug 2013 09:39:48 -0500
Subject: [Freeswitch-users] ACL Behaviour
In-Reply-To: <CAMbW9oJeoPTzbiWoqtCUL5eX37EbkfycfTxyJBBZ8FktFHXU0w@mail.gmail.com>
References: <CAMbW9oJur+kuLM-2Q5hpkeDQ=eJOboA6EJGi2=Makfb1xbE41w@mail.gmail.com>
	<CAFiqYu=NUrp1A8hm+2QG7rT4HtQJXPGhuDayLw-gPr4O31Rm_A@mail.gmail.com>
	<BF98DFD5-4EDB-4208-8A27-134FCD55661F@jerris.com>
	<CAMbW9oJeoPTzbiWoqtCUL5eX37EbkfycfTxyJBBZ8FktFHXU0w@mail.gmail.com>
Message-ID: <CAKbxfG8UZVzahUjusnV3k=6Ts7tjgQjdFarj4GGMfQgeoijV-A@mail.gmail.com>

That may be your perception but the reason is simple.  Discussions of
software issues are hard to track when you have to read all of thousands of
emails on the list and cross reference them in your mind.

Individuals on the list only care about their own thread so its not nearly
as hard.

Jiras are a forum and a systematic way to track issues feature discussions
and bugs and annotate code changes with tickets.

What is more tedious is that we continue to offer as much help as we can
and people complain about our policies designed to simplify this process.

So your original email was fine minus the snark about jira because that is
exactly what you should do and let the bug marshals process it.
On Aug 21, 2013 9:24 AM, "Andrew Cassidy" <andrew at cassidywebservices.co.uk>
wrote:

> No, but I remember about a week long period where the reply to nearly
> every emails was 'FILE A JIRA' and it was starting to get tedious. I think
> there were even meme images created :)
>
>
> On 21 August 2013 14:41, Michael Jerris <mike at jerris.com> wrote:
>
>> NO NEED TO SHOUT OVER IT!!!
>>
>> On Aug 21, 2013, at 6:08 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>> Without shouting file a jira... if that is intended behaviour a jira
>> ticket could be commented and closed as such, and refered to from any
>> duplicate tickets later. Or corrected.
>>
>>
>>
>> On Wednesday, August 21, 2013, Andrew Cassidy wrote:
>>
>>> Hi all, I've just noticed a slightly unexpected (in my view) behavior
>>> when checking ACLs.
>>>
>>> If you have an empty ACL with default="allow" it will always return
>>> false. Of course the workaround is to add a nonsense node into the list
>>> such as 0.0.0.0/32 (either allow or deny) then the ACL works as
>>> expected.
>>>
>>> Now, before people shout FILE A JIRA at me, the reason I haven't is
>>> because this behavior may be intended to prevent users accidentally leaving
>>> their FreeSWITCH installations open.
>>>
>>> So my question is, is this the intended behaviour? If so, I'll add it to
>>> the wiki.
>>>
>>> Thanks,
>>>
>>> --
>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>> Managing Director
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 hunterj91 at hotmail.com  Wed Aug 21 19:00:11 2013
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Wed, 21 Aug 2013 15:00:11 +0000
Subject: [Freeswitch-users] RFC4579- Adding a Participant by Focus- Invite
	with contact ; isfocus
Message-ID: <DUB121-W49C018631A3F6B4749DB92BD4C0@phx.gbl>

Hi Guys,
I previously posted questions about getting RFC4575/4579 working with Freeswitch, and in terms of SUBSCRIBE/NOTIFY this is working fine so thanks very much for the help.
 I just wondered if it had been tested, in particular around RFC4579, and section 5.2;
 5.2.  INVITE: Adding a Participant by the Focus - Dial-Out
   To directly add a participant to a conference, a focus SHOULD send an   INVITE to the participant containing a Contact header field with the   conference URI and the 'isfocus' feature parameter.

I have tested this, I cant see ;isfocus on the contact header field, when getting freeswitch to add a participant using originate, be it manually or via api.
Is there again any particular syntax required for this to work, or has it been tested, as I just see the standard contact header of <sip:mod_sofia at xx.xx.xx.xx:5080> and no ;isfocus.
Many thanks
Jon 		 	   		  
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From andrew at cassidywebservices.co.uk  Wed Aug 21 19:10:32 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 21 Aug 2013 16:10:32 +0100
Subject: [Freeswitch-users] ACL Behaviour
In-Reply-To: <CAKbxfG8UZVzahUjusnV3k=6Ts7tjgQjdFarj4GGMfQgeoijV-A@mail.gmail.com>
References: <CAMbW9oJur+kuLM-2Q5hpkeDQ=eJOboA6EJGi2=Makfb1xbE41w@mail.gmail.com>
	<CAFiqYu=NUrp1A8hm+2QG7rT4HtQJXPGhuDayLw-gPr4O31Rm_A@mail.gmail.com>
	<BF98DFD5-4EDB-4208-8A27-134FCD55661F@jerris.com>
	<CAMbW9oJeoPTzbiWoqtCUL5eX37EbkfycfTxyJBBZ8FktFHXU0w@mail.gmail.com>
	<CAKbxfG8UZVzahUjusnV3k=6Ts7tjgQjdFarj4GGMfQgeoijV-A@mail.gmail.com>
Message-ID: <CAMbW9oKWoZwbGeAqzCb63wgh7YjQLKdC0NHJazrq_TvK6PF9pw@mail.gmail.com>

I didn't mean it was tedious for me, more you guys. And in all seriousness
I can sympathise with the plight of managing tasks. Only today a customer
sent me a single email with no less than 21 individual items that I have to
wade through and track.

Of course what I didn't realise is even though for me it was a simple
yes/no question, filing a Jira would be so much more useful to the
community as a whole than just replying yes or no to me.


On 21 August 2013 15:39, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> That may be your perception but the reason is simple.  Discussions of
> software issues are hard to track when you have to read all of thousands of
> emails on the list and cross reference them in your mind.
>
> Individuals on the list only care about their own thread so its not nearly
> as hard.
>
> Jiras are a forum and a systematic way to track issues feature discussions
> and bugs and annotate code changes with tickets.
>
> What is more tedious is that we continue to offer as much help as we can
> and people complain about our policies designed to simplify this process.
>
> So your original email was fine minus the snark about jira because that is
> exactly what you should do and let the bug marshals process it.
> On Aug 21, 2013 9:24 AM, "Andrew Cassidy" <andrew at cassidywebservices.co.uk>
> wrote:
>
>> No, but I remember about a week long period where the reply to nearly
>> every emails was 'FILE A JIRA' and it was starting to get tedious. I think
>> there were even meme images created :)
>>
>>
>> On 21 August 2013 14:41, Michael Jerris <mike at jerris.com> wrote:
>>
>>> NO NEED TO SHOUT OVER IT!!!
>>>
>>> On Aug 21, 2013, at 6:08 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>>
>>> Without shouting file a jira... if that is intended behaviour a jira
>>> ticket could be commented and closed as such, and refered to from any
>>> duplicate tickets later. Or corrected.
>>>
>>>
>>>
>>> On Wednesday, August 21, 2013, Andrew Cassidy wrote:
>>>
>>>> Hi all, I've just noticed a slightly unexpected (in my view) behavior
>>>> when checking ACLs.
>>>>
>>>> If you have an empty ACL with default="allow" it will always return
>>>> false. Of course the workaround is to add a nonsense node into the list
>>>> such as 0.0.0.0/32 (either allow or deny) then the ACL works as
>>>> expected.
>>>>
>>>> Now, before people shout FILE A JIRA at me, the reason I haven't is
>>>> because this behavior may be intended to prevent users accidentally leaving
>>>> their FreeSWITCH installations open.
>>>>
>>>> So my question is, is this the intended behaviour? If so, I'll add it
>>>> to the wiki.
>>>>
>>>> Thanks,
>>>>
>>>> --
>>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>>> Managing Director
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>>
>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>  *03300 100 961
>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From krice at freeswitch.org  Wed Aug 21 19:22:57 2013
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 21 Aug 2013 10:22:57 -0500
Subject: [Freeswitch-users] Weekly News and Notes (Ken Style)
Message-ID: <CAPcNkdeKT-o7ehge=muLyxMGyFU0jAyd6XjGRAWjA3M1Tr6NLA@mail.gmail.com>

Hey Guys,
So I hope everyone that was able to make it to ClueCon was able to rest up
as now its time to get back to work!

If you saw Seven Du at ClueCon, then you dont want to miss todays
conference call, Seven will be joining us to talk FreeSWITCH, Video, and
other bits he's been working on. this will be covering some of the same
info from his presentation and expanding on it. So Come out and Join us,
remember thats at Noon Central, 1P Eastern time on the FreeSWITCH
Conference Bridge.

See http://wiki.freeswitch.org/wiki/Weekly_Conference_Call for more info on
how to join the call!

Be there!
--K
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From lists at kavun.ch  Wed Aug 21 19:53:15 2013
From: lists at kavun.ch (Emrah)
Date: Wed, 21 Aug 2013 17:53:15 +0200
Subject: [Freeswitch-users] Cannot load mod_spandsp.so: undefined
	symbol: lzma_stream_encoder
In-Reply-To: <5214D112.9070607@coppice.org>
References: <8E465A5B-2B1E-42B1-A6C5-A445AAB2EA8A@kavun.ch>
	<5214D112.9070607@coppice.org>
Message-ID: <414A04D3-B216-4509-81DA-08D0AC3F2E45@kavun.ch>

Good point. 
Where can I see the search paths for the loader?

Thanks!
On Aug 21, 2013, at 4:39 PM, Steve Underwood <steveu at coppice.org> wrote:

> On 08/21/2013 10:12 PM, Emrah wrote:
>> Hi guys,
>> 
>> I am having an issue getting mod_spandsp to work on a fresh install.
>> Running Debian 7, error message as follows:
>> 
>> ERR [module load file routine returned an error]
>> 2013-08-21 14:11:20.773383 [CRIT] switch_loadable_module.c:1383 Error Loading module /usr/local/freeswitch/mod/mod_spandsp.so
>> **/usr/local/freeswitch/mod/mod_spandsp.so: undefined symbol: lzma_stream_encoder**
>> 
>> It seems like a similar issue was reported here:
>> http://jira.freeswitch.org/browse/FS-4679?page=com.atlassian.streams.streams-jira-plugin:activity-stream-issue-tab
>> 
>> Any help would be greatly appreciated.
>> 
>> Emrah
>> 
> This means liblzma cannot be found at run time, although FS should not 
> have linked if this was missing at compile time. Do you have liblzma in 
> a directory that the linker searches but the loader does not?
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mishehu at freeswitch.org  Wed Aug 21 20:05:04 2013
From: mishehu at freeswitch.org (I put the Who? in Mishehu)
Date: Wed, 21 Aug 2013 11:05:04 -0500
Subject: [Freeswitch-users] Cannot load mod_spandsp.so: undefined
 symbol: lzma_stream_encoder
In-Reply-To: <414A04D3-B216-4509-81DA-08D0AC3F2E45@kavun.ch>
References: <8E465A5B-2B1E-42B1-A6C5-A445AAB2EA8A@kavun.ch>
	<5214D112.9070607@coppice.org>
	<414A04D3-B216-4509-81DA-08D0AC3F2E45@kavun.ch>
Message-ID: <5214E530.9010306@freeswitch.org>

Sounds like you'll have to do a bit of `find` on your system to locate 
the liblzma, and then set LD_LIBRARY_PATH shell variable before you 
launch FreeSWITCH so that it can locate the path to the liblzma.  (for 
example, LD_LIBRARY_PATH=$LD_LIBRARY_PATH:/directory/liblzma/is/in 
/usr/local/freeswitch/bin/freeswitch ...)

-Yossi

On 08/21/2013 10:53 AM, Emrah wrote:
> Good point.
> Where can I see the search paths for the loader?
>
> Thanks!
> On Aug 21, 2013, at 4:39 PM, Steve Underwood <steveu at coppice.org> wrote:
>
>> On 08/21/2013 10:12 PM, Emrah wrote:
>>> Hi guys,
>>>
>>> I am having an issue getting mod_spandsp to work on a fresh install.
>>> Running Debian 7, error message as follows:
>>>
>>> ERR [module load file routine returned an error]
>>> 2013-08-21 14:11:20.773383 [CRIT] switch_loadable_module.c:1383 Error Loading module /usr/local/freeswitch/mod/mod_spandsp.so
>>> **/usr/local/freeswitch/mod/mod_spandsp.so: undefined symbol: lzma_stream_encoder**
>>>
>>> It seems like a similar issue was reported here:
>>> http://jira.freeswitch.org/browse/FS-4679?page=com.atlassian.streams.streams-jira-plugin:activity-stream-issue-tab
>>>
>>> Any help would be greatly appreciated.
>>>
>>> Emrah
>>>
>> This means liblzma cannot be found at run time, although FS should not
>> have linked if this was missing at compile time. Do you have liblzma in
>> a directory that the linker searches but the loader does not?
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Wed Aug 21 20:16:13 2013
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Aug 2013 11:16:13 -0500
Subject: [Freeswitch-users] TLS/Freeswitch self signed certs
In-Reply-To: <CABkS=xpqry45O7XX7s4R5ib-uwD-wyW7TqAZtTVRujr0XpzaBg@mail.gmail.com>
References: <CABkS=xockpPROOhnbSaYJxt0uw2bUtKbyYus_CWqEsWz=sjB6Q@mail.gmail.com>
	<CAC0j6rfkNbJ7KMKMo7VVfGRG0uox3xZjqaw8heX2=HE09TcrEg@mail.gmail.com>
	<CABkS=xqg_Nw=X97szavNccZOT+cHA5PZ2sQ0-Tg=7DW+cqZTrg@mail.gmail.com>
	<CABkS=xqE=3D_PuSVPLWXM8gFzf_7+RPYjj4PTNAUUTU=F1iyog@mail.gmail.com>
	<FC8B9B69-D238-428C-9799-C3EFBA64DDB5@jerris.com>
	<CABkS=xpqry45O7XX7s4R5ib-uwD-wyW7TqAZtTVRujr0XpzaBg@mail.gmail.com>
Message-ID: <47059AC9-3C1A-4FE0-B82E-275EC3500818@freeswitch.org>

Your diff file on your jira is backwards, can you please patch the .in file itself then git diff and attach that.

On Aug 12, 2013, at 7:18 PM, Peter <eidevm5 at gmail.com> wrote:

> Yes, I'll open a jira ticket.
> 
> Also, just wanted to correct something I wrote in my last email.
> 
> Where I said:
> 
> "If these are present, then Android will treat the cert as a standard user cert."
> 
> I meant
> 
> "If these are NOT present, then Android will treat the cert as a standard user cert."
> 
> 
> On Tue, Aug 13, 2013 at 12:50 AM, Michael Jerris <mike at jerris.com> wrote:
> This sounds like it should be in the script for everyone.  Can you open a bug on jira.freeswitch.org for this issue.
> 
> Thanks
> Mike
> 
> On Aug 6, 2013, at 2:16 AM, Peter <eidevm5 at gmail.com> wrote:
> 
>> Finally figured out the issue was related to the gentls_cert script was generating an openssl template that didn't have the required x509v3 extensions set.
>> 
>> I modified the script where it generates config.tpl to add
>> 
>>                         x509_extensions = v3_ca
>> 
>> to the [req] section, then I added the section:
>> 
>>                         [ v3_ca ]
>>                         subjectKeyIdentifier=hash
>>                         authorityKeyIdentifier=keyid:always,issuer
>>                         basicConstraints=CA:TRUE
>> 
>> Now when you issue:
>> 
>> openssl x509 -noout -inform pem -text -in cafile.pem
>> 
>> you'll see the following section:
>> 
>>         X509v3 extensions:
>>             X509v3 Subject Key Identifier:
>>                 02:0A:A8:D0:5C:23:7C:8B:C4:EF:79:11:C7:0C:A8:86:71:15:59:D5
>>             X509v3 Authority Key Identifier:
>>                 keyid:02:0A:A8:D0:5C:23:7C:8B:C4:EF:79:11:C7:0C:A8:86:71:15:59:D5
>> 
>>             X509v3 Basic Constraints:
>>                 CA:TRUE
>> 
>> If these are present, then Android will treat the cert as a standard user cert.
>> 
>> Then it was a simple matter of copying cafile.pem to cafile.crt on the sdcard on the Android device and using the "install from device storage" option.
>> 
>> When the cert installer dialog comes up, it will now detect cafile.crt as a CA cert and not user cert.
>> 
>> Hope this helps other people, as cert management on Android is a right pain in the $#%^.
>> 
>> Peter
>> 
>> 
>> 
>> On Tue, Aug 6, 2013 at 2:31 PM, Peter <eidevm5 at gmail.com> wrote:
>> The reason I put it on a webserver is mostly for convenience to make it easier to install.
>> 
>> I tried copying cafile.pem to /sdcard on a Galaxy Note II, but when I try the "Install from device storage" option, it just comes back with:
>> 
>> "No certificate file found on SD card"
>> 
>> 
>> 
>> On Mon, Aug 5, 2013 at 5:51 PM, Mehroz Ashraf <mehroz.ashraf85 at gmail.com> wrote:
>> Why do you want to place the cert on webserver and point android browser? If you are doing this to download cert into android then that is probably not the right approach.
>> 
>> I used cafile.pem (without converting it into .der format) and placed the file in  SD card or phone memory, and point out linphone to get the CA from the path. You may search in libraries where it need to tell the path. 
>> 
>> 
>> On Mon, Aug 5, 2013 at 12:15 PM, Peter <eidevm5 at gmail.com> wrote:
>> Has anyone managed to get TLS working between Android Linphone and Freeswitch?
>> 
>> I've done the basic TLS setup as per https://wiki.freeswitch.org/wiki/Tls
>> 
>> I then convert the CA cert from PEM to DER format with:
>> 
>> openssl x509  -inform PEM -outform der -in cafile.pem -out fs.crt
>> 
>> I place fs.crt on a webserver and point my Android browser to it.
>> 
>> When I click on fs.crt, I get the default Android Certificate installer popup, but it always says:
>> 
>> "Package contains: one user certificate"
>> 
>> ie: it thinks it is a user cert rather than a CA cert.
>> 
>> Android appears to be a real pain to add a CA to its trusted credential store.
>> 
>> Really interested if anyone has managed to get Android to import the CA cert.
>> 
>> Thanks
>> 
>> Peter
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 brian at freeswitch.org  Wed Aug 21 20:30:34 2013
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Aug 2013 11:30:34 -0500
Subject: [Freeswitch-users] TLS setup failed (after changing key type)
In-Reply-To: <1371825449535-7592067.post@n2.nabble.com>
References: <1371825449535-7592067.post@n2.nabble.com>
Message-ID: <3093BAC3-571B-438F-96D7-8CE14091399A@freeswitch.org>

I've extracted what you were meaning to do but please next time please provide a diff to gentls_cert.in as that file is generated.

/b

On Jun 21, 2013, at 9:37 AM, mehroz <mehroz.ashraf85 at gmail.com> wrote:

> Hi all, 
> 
> My SIP over TLS is working perfect as long as i used gentls_cert script.
> 
> I have changed script to create ECC keys and changed that script replacing
> with :
> openssl ecparam -name secp160r2 -out CA_CURVE.pem
> 
> openssl req -out "${CONFDIR}/CA/cacert.pem" -new -x509 -keyout
> "${CONFDIR}/CA/cakey.pem" -newkey ec:CA_CURVE.pem -config "${TMPFILE}.cfg"
> -nodes -days ${DAYS} -sha1 >/dev/null 
> 
> openssl req -new -out "${TMPFILE}.req" -newkey ec:CA_CURVE.pem -keyout
> "${TMPFILE}.key" -config "${TMPFILE}.cfg" -nodes -sha1
> 
> And since then, i see "TLS setup failed"... 
> any one?
> 
> 
> 
> --
> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/TLS-setup-failed-after-changing-key-type-tp7592067.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Wed Aug 21 20:43:43 2013
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 21 Aug 2013 11:43:43 -0500
Subject: [Freeswitch-users] Patchs Jira and more....
Message-ID: <CE3A586F.7920E%krice@freeswitch.org>

Hey Guys, there seems to be a rash of people uploading patchs to Jira lately
that just do not work...

So lets hit this one really quick...

1. if you are patching a generated File like the Makefile or configure
script make sure you are patching the files they are generated from... ie:
configure.in for configure...
2. don?t try to diff 2 separate trees... ?git diff >
/tmp/my.fancy.patch.txt? from the root of your fs source checkout will give
you a plain text file of the patch suitable for uploading to jira.
3. Please include all information you have on the jira and what
problem/feature the patch addresses. Don?t just say ?here?s a patch? and
expect someone to look at it without knowing what it is.
4. If a commiter comments on your jira that you need to freshen the patch
please do so in a timely fashion or your patch will go back to the bottom of
the queue as they will move on to something else... This isnt ignoring your
efforts, but developers tend to get distracted easily with new shiny objects
so getting back to them quickly helps tremendously.


These simple things will help keep the devs from going crazy!


Now for you guys that use git on a regular basis, before you blow a gasket,
yes this is very basic, and is intended to be so... Yes there are other (and
possibly arguable better ways to generate a patch, but this way is the
simplest for people not familiar with git)


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From brian at freeswitch.org  Wed Aug 21 20:56:44 2013
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Aug 2013 11:56:44 -0500
Subject: [Freeswitch-users] Cannot load mod_spandsp.so: undefined
	symbol: lzma_stream_encoder
In-Reply-To: <414A04D3-B216-4509-81DA-08D0AC3F2E45@kavun.ch>
References: <8E465A5B-2B1E-42B1-A6C5-A445AAB2EA8A@kavun.ch>
	<5214D112.9070607@coppice.org>
	<414A04D3-B216-4509-81DA-08D0AC3F2E45@kavun.ch>
Message-ID: <9FDF70D3-8552-49A8-A94A-446F2901852F@freeswitch.org>


Have you tried to 'make spandsp-reconf' to make sure your spandsp source is in order?

/b



From brian at freeswitch.org  Wed Aug 21 20:57:18 2013
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Aug 2013 11:57:18 -0500
Subject: [Freeswitch-users] can't load codec
In-Reply-To: <CAJhzRKcrQ6u2Pgv+4R3A3pfpNPfVH0qLBQm-s+W=Td2cj9w6iQ@mail.gmail.com>
References: <CAJhzRKcrQ6u2Pgv+4R3A3pfpNPfVH0qLBQm-s+W=Td2cj9w6iQ@mail.gmail.com>
Message-ID: <D25AA888-60D1-4D8E-8170-68919AE763CD@freeswitch.org>

Have you filed a jira?  This smells like a bug possibly.

On Aug 21, 2013, at 2:04 AM, pankajanand18 at gmail.com wrote:

> 2013-08-20 19:14:58.820110 [WARNING] switch_core_codec.c:699 Codec PROXY Exists
> but not at the desired implementation. 90000hz 0ms
> 2013-08-20 19:14:58.820110 [ERR] switch_core_media.c:1837 Can't load codec?
> 
> Getting this error with following settings:
> 
> 1. media proxy mode enabled
> 2. late negotiation enabled
> 3. <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU,PCMA,GSM,H264"/>
> 
> 
> Both sip clients are are h264 capable. The same configuration work fine on freeswitch running over ubuntu 12.04 but generates the above log and error on window 7 x64. 
> 
> I found that the freeswitch version I was using on linux was 1.2 and on windows it was 1.5 with git 39ad799
> On Linux I checked out same revision and built it again. I could reproduce the issue on linux too. but in the latest revision the problem seems to be fixed. So now I am going to build the the current revision code for windows.
> 
> Any comment on this or have anybody faced this issue ? 
> 
> Cheers,
> Pankaj
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 brian at freeswitch.org  Wed Aug 21 21:01:59 2013
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Aug 2013 12:01:59 -0500
Subject: [Freeswitch-users] Validating configuration freeswitch.xml.fsxml
In-Reply-To: <FB7D97A214987F458242ACBDF87614073B1BD199FB@clubvirtual40.ClubMessage.local>
References: <FB7D97A214987F458242ACBDF87614073B1BD199FB@clubvirtual40.ClubMessage.local>
Message-ID: <2057FFED-5D9E-4378-A15D-AC684D6A0B86@freeswitch.org>

Anything in X-PRE-XXXX directives aren't preserved into freeswith.xml.fsxml  so you won't see those.

/b

On Aug 21, 2013, at 2:27 AM, Grant Bagdasarian <GB at cm.nl> wrote:

> Hello,
>  
> I?ve started to configure FS from scratch by making my own configuration files, based on the files in the vanilla directory.
> I don?t know if this is normal behavior but whenever I start FreeSwitch and check the freeswith.xml.fsxml file, the XML elements in the freeswitch/conf/customsbc/freeswitch.xml and vars.xml are not shown. Only the commentary is shown in this .fsxml file. Is this normal?
>  
> Also when running the netstat ?unlp command, I see freeswitch isn?t binding to the address defined in the bind_server_ip variable. Is this because the configuration isn?t properly loaded?
> Htop shows FS is running.
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 brian at freeswitch.org  Wed Aug 21 21:04:33 2013
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Aug 2013 12:04:33 -0500
Subject: [Freeswitch-users] TLS/Freeswitch self signed certs
In-Reply-To: <47059AC9-3C1A-4FE0-B82E-275EC3500818@freeswitch.org>
References: <CABkS=xockpPROOhnbSaYJxt0uw2bUtKbyYus_CWqEsWz=sjB6Q@mail.gmail.com>
	<CAC0j6rfkNbJ7KMKMo7VVfGRG0uox3xZjqaw8heX2=HE09TcrEg@mail.gmail.com>
	<CABkS=xqg_Nw=X97szavNccZOT+cHA5PZ2sQ0-Tg=7DW+cqZTrg@mail.gmail.com>
	<CABkS=xqE=3D_PuSVPLWXM8gFzf_7+RPYjj4PTNAUUTU=F1iyog@mail.gmail.com>
	<FC8B9B69-D238-428C-9799-C3EFBA64DDB5@jerris.com>
	<CABkS=xpqry45O7XX7s4R5ib-uwD-wyW7TqAZtTVRujr0XpzaBg@mail.gmail.com>
	<47059AC9-3C1A-4FE0-B82E-275EC3500818@freeswitch.org>
Message-ID: <DD9DE0F8-B525-4B0D-B20B-F181D4765A63@freeswitch.org>

I've extracted the info, please re-test the gentls_cert from the generated .in file please.

/b

On Aug 21, 2013, at 11:16 AM, Brian West <brian at freeswitch.org> wrote:

> Your diff file on your jira is backwards, can you please patch the .in file itself then git diff and attach that.

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From jpyle at fidelityvoice.com  Wed Aug 21 21:22:29 2013
From: jpyle at fidelityvoice.com (Jeff Pyle)
Date: Wed, 21 Aug 2013 13:22:29 -0400
Subject: [Freeswitch-users] in/outbound-use-uuid-as-callid as channel
	variables
In-Reply-To: <CAFiqYumfRbnwe6EEBL8m_VwBw1U-=GUZO-Cx13yV6QBb2Nyaqw@mail.gmail.com>
References: <CACYJG3J93Ge1Qca+=jacNfP57eWPTKu7kFaovVgkSSbgcbEcEA@mail.gmail.com>
	<CAFiqYum+93poAaGpvqO3MfzkYgf49tityWqQd2hcEUkNQF4o0w@mail.gmail.com>
	<CACYJG3LeF8s9cPTDDbGzEAf3jnfi_s=W5Ns=KOsV4uP1bt=uqA@mail.gmail.com>
	<CAFiqYumfRbnwe6EEBL8m_VwBw1U-=GUZO-Cx13yV6QBb2Nyaqw@mail.gmail.com>
Message-ID: <CACYJG3JaPCNkqp14ZqD8cBgDwX5rdETxcuiNMHXBmFV7BFzCSA@mail.gmail.com>

You're right, I had quite a bit wrong.  This is very helpful.

Exporting sip_call_id works.  Thanks!



- Jeff



On Wed, Aug 21, 2013 at 9:53 AM, Steven Ayre <steveayre at gmail.com> wrote:

> They didn't seem to have any effect - I still saw a new SIP callid on the
>> b-leg INVITE
>
>
> Perhaps you're misunderstanding the purpose of them.
>
> Every channel in FreeSWITCH has a UUID. On a bridge you have two different
> channels (inbound & outbound). Their UUIDs -must- be different, to tell
> them apart.
>
> The setting you found sets the UUID from the Call-ID on the inbound
> channel, and visa versa for the outbound channel. That mostly makes it
> easier to match up packet traces to calls, and your CDRs to a customers
> (since they know the Call-ID but not your internal UUID).
>
> If you set both the UUID of the incoming channel will be the incoming
> Call-ID, the outgoing UUID will be generated by FreeSWITCH and the outgoing
> Call-ID will match the outgoing UUID. You cannot set the same UUID for both.
>
> Have you tried exporting the sip_call_id variable from the aleg to the
> bleg? I don't know if that variable is read-only though. If sofia will read
> from that should be sufficient, you won't need to tweak the UUIDs of the
> channels.
>
> If that doesn't work you could also try setting origination_uuid to
> sip_call_id in the bridge dialstring, along with the outgoing parameter you
> found.
>
> -Steve
>
>
>
> On 21 August 2013 13:21, Jeff Pyle <jpyle at fidelityvoice.com> wrote:
>
>> I was afraid of that on the inbound side.  It makes sense.
>>
>> The short answer is sometimes I want full topology-hiding, other times I
>> want the same SIP callid all the way through.  Managing it per-profile
>> could work.  It has to do with correlating voice-quality monitoring on the
>> private side with other measurements from the public side.  In cases where
>> this happens, the same callid on both sides makes things much, much easier.
>>
>> Yesterday I tried enabling both inbound-use-callid-as-uuid and
>> outbound-use-uuid-as-callid on both the public and private profiles.  They
>> didn't seem to have any effect - I still saw a new SIP callid on the b-leg
>> INVITE.  I wasn't sure how to debug it from there.
>>
>>
>> - Jeff
>>
>>
>> On Wed, Aug 21, 2013 at 6:17 AM, Steven Ayre <steveayre at gmail.com> wrote:
>>
>>> It would be too late for inbound in the dialplan as the channel already
>>> exists, therefore already has a uuid. Though you can change the uuid, you'd
>>> still have issues whenever a call hung up so quickly it never reached the
>>> dialplan (yes, I've seen that happen).
>>>
>>> Outbound you may have a chance with - but I don't know if such a
>>> variable exists.
>>>
>>> Is there a reason you can't just set it on the profile?
>>>
>>>
>>>
>>> On Tuesday, August 20, 2013, Jeff Pyle wrote:
>>>
>>>>  Hello,
>>>>
>>>> Is it possible to set the equivalent of inbound-use-uuid-as-callid
>>>> or outbound-use-uuid-as-callid per channel from within the dialplan?
>>>>
>>>>
>>>> - Jeff
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nneul at mst.edu  Wed Aug 21 21:24:45 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Wed, 21 Aug 2013 12:24:45 -0500
Subject: [Freeswitch-users] Validating configuration freeswitch.xml.fsxml
In-Reply-To: <2057FFED-5D9E-4378-A15D-AC684D6A0B86@freeswitch.org>
References: <FB7D97A214987F458242ACBDF87614073B1BD199FB@clubvirtual40.ClubMessage.local>
	<2057FFED-5D9E-4378-A15D-AC684D6A0B86@freeswitch.org>
Message-ID: <5214F7DD.3010707@mst.edu>

For debugging/tracing, it might be nice if the pre-processor directives WERE preserved, but in a non-functional manner. 
i.e. a generated comment or something like that.

-- Nathan

On 08/21/2013 12:01 PM, Brian West wrote:
> Anything in X-PRE-XXXX directives aren't preserved into freeswith.xml.fsxml  so you won't see those.
>
> /b
>
> On Aug 21, 2013, at 2:27 AM, Grant Bagdasarian <GB at cm.nl> wrote:
>
>> Hello,
>>
>> I?ve started to configure FS from scratch by making my own configuration files, based on the files in the vanilla directory.
>> I don?t know if this is normal behavior but whenever I start FreeSwitch and check the freeswith.xml.fsxml file, the XML elements in the freeswitch/conf/customsbc/freeswitch.xml and vars.xml are not shown. Only the commentary is shown in this .fsxml file. Is this normal?
>>
>> Also when running the netstat ?unlp command, I see freeswitch isn?t binding to the address defined in the bind_server_ip variable. Is this because the configuration isn?t properly loaded?
>> Htop shows FS is running.
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From jpyle at fidelityvoice.com  Wed Aug 21 22:00:45 2013
From: jpyle at fidelityvoice.com (Jeff Pyle)
Date: Wed, 21 Aug 2013 14:00:45 -0400
Subject: [Freeswitch-users] in/outbound-use-uuid-as-callid as channel
	variables
In-Reply-To: <CACYJG3JaPCNkqp14ZqD8cBgDwX5rdETxcuiNMHXBmFV7BFzCSA@mail.gmail.com>
References: <CACYJG3J93Ge1Qca+=jacNfP57eWPTKu7kFaovVgkSSbgcbEcEA@mail.gmail.com>
	<CAFiqYum+93poAaGpvqO3MfzkYgf49tityWqQd2hcEUkNQF4o0w@mail.gmail.com>
	<CACYJG3LeF8s9cPTDDbGzEAf3jnfi_s=W5Ns=KOsV4uP1bt=uqA@mail.gmail.com>
	<CAFiqYumfRbnwe6EEBL8m_VwBw1U-=GUZO-Cx13yV6QBb2Nyaqw@mail.gmail.com>
	<CACYJG3JaPCNkqp14ZqD8cBgDwX5rdETxcuiNMHXBmFV7BFzCSA@mail.gmail.com>
Message-ID: <CACYJG3K2NP=gHPokC=8poahA4hp9+KcWhiirUvJBZSz9n=YRQw@mail.gmail.com>

I spoke too soon.  Exporting sip_call_id doesn't throw an error, but it
doesn't seem to do anything either.  I'm not sure what I was looking at the
first time.

Your second recommendation of setting the origination_uuid and having the
outbound param set does indeed work for.  Triple checked.  :)


Thanks,
Jeff


On Wed, Aug 21, 2013 at 1:22 PM, Jeff Pyle <jpyle at fidelityvoice.com> wrote:

> You're right, I had quite a bit wrong.  This is very helpful.
>
> Exporting sip_call_id works.  Thanks!
>
>
>
> - Jeff
>
>
>
> On Wed, Aug 21, 2013 at 9:53 AM, Steven Ayre <steveayre at gmail.com> wrote:
>
>> They didn't seem to have any effect - I still saw a new SIP callid on the
>>> b-leg INVITE
>>
>>
>> Perhaps you're misunderstanding the purpose of them.
>>
>> Every channel in FreeSWITCH has a UUID. On a bridge you have two
>> different channels (inbound & outbound). Their UUIDs -must- be different,
>> to tell them apart.
>>
>> The setting you found sets the UUID from the Call-ID on the inbound
>> channel, and visa versa for the outbound channel. That mostly makes it
>> easier to match up packet traces to calls, and your CDRs to a customers
>> (since they know the Call-ID but not your internal UUID).
>>
>> If you set both the UUID of the incoming channel will be the incoming
>> Call-ID, the outgoing UUID will be generated by FreeSWITCH and the outgoing
>> Call-ID will match the outgoing UUID. You cannot set the same UUID for both.
>>
>> Have you tried exporting the sip_call_id variable from the aleg to the
>> bleg? I don't know if that variable is read-only though. If sofia will read
>> from that should be sufficient, you won't need to tweak the UUIDs of the
>> channels.
>>
>> If that doesn't work you could also try setting origination_uuid to
>> sip_call_id in the bridge dialstring, along with the outgoing parameter you
>> found.
>>
>> -Steve
>>
>>
>>
>> On 21 August 2013 13:21, Jeff Pyle <jpyle at fidelityvoice.com> wrote:
>>
>>> I was afraid of that on the inbound side.  It makes sense.
>>>
>>> The short answer is sometimes I want full topology-hiding, other times I
>>> want the same SIP callid all the way through.  Managing it per-profile
>>> could work.  It has to do with correlating voice-quality monitoring on the
>>> private side with other measurements from the public side.  In cases where
>>> this happens, the same callid on both sides makes things much, much easier.
>>>
>>> Yesterday I tried enabling both inbound-use-callid-as-uuid and
>>> outbound-use-uuid-as-callid on both the public and private profiles.  They
>>> didn't seem to have any effect - I still saw a new SIP callid on the b-leg
>>> INVITE.  I wasn't sure how to debug it from there.
>>>
>>>
>>> - Jeff
>>>
>>>
>>> On Wed, Aug 21, 2013 at 6:17 AM, Steven Ayre <steveayre at gmail.com>wrote:
>>>
>>>> It would be too late for inbound in the dialplan as the channel already
>>>> exists, therefore already has a uuid. Though you can change the uuid, you'd
>>>> still have issues whenever a call hung up so quickly it never reached the
>>>> dialplan (yes, I've seen that happen).
>>>>
>>>> Outbound you may have a chance with - but I don't know if such a
>>>> variable exists.
>>>>
>>>> Is there a reason you can't just set it on the profile?
>>>>
>>>>
>>>>
>>>> On Tuesday, August 20, 2013, Jeff Pyle wrote:
>>>>
>>>>>  Hello,
>>>>>
>>>>> Is it possible to set the equivalent of inbound-use-uuid-as-callid
>>>>> or outbound-use-uuid-as-callid per channel from within the dialplan?
>>>>>
>>>>>
>>>>> - Jeff
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 a.daydreamer82 at gmail.com  Wed Aug 21 21:44:03 2013
From: a.daydreamer82 at gmail.com (Master Can)
Date: Wed, 21 Aug 2013 19:44:03 +0200
Subject: [Freeswitch-users] How to disable TCP Keepalive on a TLS connection?
Message-ID: <CA+T6Pv_SH5mcuYRxHj-2CvM_UAjcQS_NuUKP5E33yy-Ppi2zoQ@mail.gmail.com>

Hello,

I'm running freeswitch 1.2.10, with tls-only.
I've tried 2 SIP Useragents now: PhonerLite and CSipSimple.

My linux server shows me with netstat --timers that both useragents (both
server sockets) use keepalive, with a value of 30 seconds.

How can I disable keepalive on the TCP layer completely? My useragents take
care of sending keepalive packets anyway, so Freeswitch does not need to do
that. It's not mobile friendly, it's eating up battery power if the
useragents keep receiving keepalive every 30 seconds.

I've tried to set
<param name="tcp-keepalive" value="disable"/>
in internal.xml but to no avail. It didn't change a thing. Setting this to
60000 didn't change the output of netstat --timers either.

Any advice?

best regards,
Can
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From brian at freeswitch.org  Wed Aug 21 22:18:39 2013
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Aug 2013 13:18:39 -0500
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
Message-ID: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>

Dear FreeSWITCHERS,
	If you have a color fax machine with REAL COPPER OLD SCHOOL PSTN connectivity please respond with your phone number... I want to test if I can send you a color fax .. that is if you wanna test it out with me pretty please.  (US ONLY)

Thanks,
--
Brian West
brian at freeswitch.org
FreeSWITCH Solutions, LLC
PO BOX PO BOX 2531
Brookfield, WI 53008-2531
Twitter: @FreeSWITCH_Wire (Follow us today!)
http://www.freeswitchbook.com
http://www.freeswitchcookbook.com

T: +1.918.420.9001  |  F: +1.918.420.9002  |  M: +1.918.424.WEST
iNUM: +883 5100 1420 9001
ISN: 410*543
Skype:briankwest
PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)










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From grcamauer at gmail.com  Wed Aug 21 22:25:18 2013
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Wed, 21 Aug 2013 15:25:18 -0300
Subject: [Freeswitch-users] Error with SQLDB on FS startup
Message-ID: <CAPNTnnrg6QBB5x0Zahjp=7QsLy_54_CVE_GQ6kLgbJaQOFTRDw@mail.gmail.com>

I am getting the following errors when starting up FS:

[root at Freeswitch freeswitch]# /usr/local/freeswitch/bin/freeswitch
2013-08-21 15:13:55.201104 [INFO] switch_event.c:596 Activate Eventing
Engine.
2013-08-21 15:13:55.201308 [WARNING] switch_event.c:570 Create additional
event dispatch thread 0
2013-08-21 15:13:55.262832 [INFO] switch_nat.c:420 Scanning for NAT
2013-08-21 15:13:55.262950 [DEBUG] switch_nat.c:170 Checking for PMP 1/5
2013-08-21 15:13:55.513013 [DEBUG] switch_nat.c:170 Checking for PMP 2/5
2013-08-21 15:13:56.012990 [DEBUG] switch_nat.c:170 Checking for PMP 3/5
2013-08-21 15:13:57.013009 [DEBUG] switch_nat.c:170 Checking for PMP 4/5
2013-08-21 15:13:59.013009 [DEBUG] switch_nat.c:170 Checking for PMP 5/5
2013-08-21 15:14:03.013009 [ERR] switch_nat.c:201 Error checking for PMP
[general error]
2013-08-21 15:14:03.013039 [DEBUG] switch_nat.c:425 Checking for UPnP
2013-08-21 15:14:06.066363 [INFO] switch_nat.c:441 No PMP or UPnP NAT
devices detected!
2013-08-21 15:14:06.068084 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR [no
such table: channels]
drop table channels
2013-08-21 15:14:06.068108 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR [no
such table: calls]
drop table calls
2013-08-21 15:14:06.068125 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR [no
such table: detailed_calls]
drop view detailed_calls
2013-08-21 15:14:06.068140 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR [no
such table: basic_calls]
drop view basic_calls
2013-08-21 15:14:06.068155 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR [no
such table: interfaces]
drop table interfaces
2013-08-21 15:14:06.068169 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR [no
such table: tasks]
drop table tasks
2013-08-21 15:14:06.081283 [CONSOLE] switch_core.c:1293 Created ip list lan
default (allow)
2013-08-21 15:14:06.081302 [CONSOLE] switch_core.c:1293 Created ip list
domains default (deny)
...

I am on FreeSWITCH Version 1.2.12+git~20130821T172756Z~c73179c6ad (git
c73179c 2013-08-21 17:27:56Z). (just ran make current, but it was
happenning before also).

FS continues and loads.  What are the possible side effects of these
errors?  Is there something very wrong with my setup?


-- 
Guillermo Ruiz Camauer
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From jmesquita at freeswitch.org  Wed Aug 21 22:28:14 2013
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 21 Aug 2013 15:28:14 -0300
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
Message-ID: <CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>

:( Argentina won't suffice? :D

Jo?o Mesquita
FreeSWITCH? Solutions


On Wed, Aug 21, 2013 at 3:18 PM, Brian West <brian at freeswitch.org> wrote:

> Dear FreeSWITCHERS,
>         If you have a color fax machine with REAL COPPER OLD SCHOOL PSTN
> connectivity please respond with your phone number... I want to test if I
> can send you a color fax .. that is if you wanna test it out with me pretty
> please.  (US ONLY)
>
> Thanks,
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire (Follow us today!)
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> T: +1.918.420.9001  |  F: +1.918.420.9002  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9001
> ISN: 410*543
> Skype:briankwest
> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
>
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ga at steadfasttelecom.com  Wed Aug 21 22:36:04 2013
From: ga at steadfasttelecom.com (Gilad Abada)
Date: Wed, 21 Aug 2013 14:36:04 -0400
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
	<CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
Message-ID: <B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>

Hey Brian,

--  


SteadFast Telecommunications, Inc.

Call us to find out how much you can save with VoIP!
                  
V: 212.589.1001

  
For over 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises. Make your virtual office a reality. Enjoy the freedom to travel while remaining connected to your office.  


On Wednesday, August 21, 2013 at 2:28 PM, Jo?o Mesquita wrote:

> :( Argentina won't suffice? :D
>  
> Jo?o Mesquita
> FreeSWITCH? Solutions
>  
>  
> On Wed, Aug 21, 2013 at 3:18 PM, Brian West <brian at freeswitch.org (mailto:brian at freeswitch.org)> wrote:
> > Dear FreeSWITCHERS,
> >         If you have a color fax machine with REAL COPPER OLD SCHOOL PSTN connectivity please respond with your phone number... I want to test if I can send you a color fax .. that is if you wanna test it out with me pretty please.  (US ONLY)
> >  
> > Thanks,
> > --
> > Brian West
> > brian at freeswitch.org (mailto:brian at freeswitch.org)
> > FreeSWITCH Solutions, LLC
> > PO BOX PO BOX 2531
> > Brookfield, WI 53008-2531
> > Twitter: @FreeSWITCH_Wire (Follow us today!)
> > http://www.freeswitchbook.com
> > http://www.freeswitchcookbook.com
> >  
> > T: +1.918.420.9001  |  F: +1.918.420.9002  |  M: +1.918.424.WEST
> > iNUM: +883 5100 1420 9001
> > ISN: 410*543
> > Skype:briankwest
> > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
> >  
> >  
> >  
> >  
> >  
> >  
> >  
> >  
> >  
> >  
> >  
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > http://www.freeswitchsolutions.com
> >  
> > 
> > 
> >  
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >  
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> http://www.freeswitchsolutions.com
>  
> 
> 
>  
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>  
>  


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From ga at steadfasttelecom.com  Wed Aug 21 22:36:30 2013
From: ga at steadfasttelecom.com (Gilad Abada)
Date: Wed, 21 Aug 2013 14:36:30 -0400
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
	<CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
	<B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
Message-ID: <9EBCB89319B84637AFE785ED2C2F8B78@steadfasttelecom.com>

Sorry?

Hey Brian,

Message me on IRC I have one setup for you to play with.

Gill  

--  


SteadFast Telecommunications, Inc.

Call us to find out how much you can save with VoIP!
                  
V: 212.589.1001

  
For over 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises. Make your virtual office a reality. Enjoy the freedom to travel while remaining connected to your office.  


On Wednesday, August 21, 2013 at 2:36 PM, Gilad Abada wrote:

> Hey Brian,
>  
> --  
>  
>  
> SteadFast Telecommunications, Inc.
>  
> Call us to find out how much you can save with VoIP!
>                   
> V: 212.589.1001
>  
>   
> For over 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises. Make your virtual office a reality. Enjoy the freedom to travel while remaining connected to your office.  
>  
>  
> On Wednesday, August 21, 2013 at 2:28 PM, Jo?o Mesquita wrote:
>  
> > :( Argentina won't suffice? :D
> >  
> > Jo?o Mesquita
> > FreeSWITCH? Solutions
> >  
> >  
> > On Wed, Aug 21, 2013 at 3:18 PM, Brian West <brian at freeswitch.org (mailto:brian at freeswitch.org)> wrote:
> > > Dear FreeSWITCHERS,
> > >         If you have a color fax machine with REAL COPPER OLD SCHOOL PSTN connectivity please respond with your phone number... I want to test if I can send you a color fax .. that is if you wanna test it out with me pretty please.  (US ONLY)
> > >  
> > > Thanks,
> > > --
> > > Brian West
> > > brian at freeswitch.org (mailto:brian at freeswitch.org)
> > > FreeSWITCH Solutions, LLC
> > > PO BOX PO BOX 2531
> > > Brookfield, WI 53008-2531
> > > Twitter: @FreeSWITCH_Wire (Follow us today!)
> > > http://www.freeswitchbook.com
> > > http://www.freeswitchcookbook.com
> > >  
> > > T: +1.918.420.9001  |  F: +1.918.420.9002  |  M: +1.918.424.WEST
> > > iNUM: +883 5100 1420 9001
> > > ISN: 410*543
> > > Skype:briankwest
> > > PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
> > >  
> > >  
> > >  
> > >  
> > >  
> > >  
> > >  
> > >  
> > >  
> > >  
> > >  
> > > _________________________________________________________________________
> > > Professional FreeSWITCH Consulting Services:
> > > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > > http://www.freeswitchsolutions.com
> > >  
> > > 
> > > 
> > >  
> > > Official FreeSWITCH Sites
> > > http://www.freeswitch.org
> > > http://wiki.freeswitch.org
> > > http://www.cluecon.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
> > >  
> >  
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org (mailto:consulting at freeswitch.org)
> > http://www.freeswitchsolutions.com
> >  
> > 
> > 
> >  
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >  
> >  
> >  
>  
>  

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From bdfoster at davri.com  Wed Aug 21 22:51:36 2013
From: bdfoster at davri.com (Brian Foster)
Date: Wed, 21 Aug 2013 14:51:36 -0400
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
	<CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
	<B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
Message-ID: <CAHgk4BWdp9iQHsZfQmNWu0JLzidQQCtGn0kzD_AAyapKOohzQA@mail.gmail.com>

Damn. T38 only here.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 21, 2013 2:42 PM, "Gilad Abada" <ga at steadfasttelecom.com> wrote:

> Hey Brian,
>
> --
>
>
> SteadFast Telecommunications, Inc.
>
> Call us to find out how much you can save with VoIP!
>
> V: 212.589.1001
>
>
> For over 35 years, Steadfast Telecommunications has been providing
> state-of-the-art communications technology to businesses and government
> agencies - large and small. Steadfast Telecommunications tailors Unified
> Communications and Voice-Over IP Solutions to single-site offices or
> multi-site and worldwide enterprises. Make your virtual office a reality.
> Enjoy the freedom to travel while remaining connected to your office.
>
> On Wednesday, August 21, 2013 at 2:28 PM, Jo?o Mesquita wrote:
>
> :( Argentina won't suffice? :D
>
> Jo?o Mesquita
> FreeSWITCH? Solutions
>
>
> On Wed, Aug 21, 2013 at 3:18 PM, Brian West <brian at freeswitch.org> wrote:
>
> Dear FreeSWITCHERS,
>         If you have a color fax machine with REAL COPPER OLD SCHOOL PSTN
> connectivity please respond with your phone number... I want to test if I
> can send you a color fax .. that is if you wanna test it out with me pretty
> please.  (US ONLY)
>
> Thanks,
> --
> Brian West
> brian at freeswitch.org
> FreeSWITCH Solutions, LLC
> PO BOX PO BOX 2531
> Brookfield, WI 53008-2531
> Twitter: @FreeSWITCH_Wire (Follow us today!)
> http://www.freeswitchbook.com
> http://www.freeswitchcookbook.com
>
> T: +1.918.420.9001  |  F: +1.918.420.9002  |  M: +1.918.424.WEST
> iNUM: +883 5100 1420 9001
> ISN: 410*543
> Skype:briankwest
> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
>
>
>
>
>
>
>
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ostolyar at netflix.com  Wed Aug 21 23:59:20 2013
From: ostolyar at netflix.com (Oleg Stolyar)
Date: Wed, 21 Aug 2013 12:59:20 -0700
Subject: [Freeswitch-users] Using Unix Environment Variables in FreeSWITCH
	vars
Message-ID: <CA+x4QzCwPGY3aYWu=m-6g69FLm0-Su6zD6hrmqqXj-iPTyVjvg@mail.gmail.com>

Hi,

I have a somewhat unusual question.  Can I use Unix variables somehow in
vars.xml?

I have an environment variable $FOO.  I want to define a variable in
vars.xml like this

<X-PRE-PROCESS cmd="set" data="my_var=value_$FOO"/>

Thank you
*Oleg*
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From vipkilla at gmail.com  Thu Aug 22 00:28:49 2013
From: vipkilla at gmail.com (Vik Killa)
Date: Wed, 21 Aug 2013 16:28:49 -0400
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <CAHgk4BWdp9iQHsZfQmNWu0JLzidQQCtGn0kzD_AAyapKOohzQA@mail.gmail.com>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
	<CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
	<B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
	<CAHgk4BWdp9iQHsZfQmNWu0JLzidQQCtGn0kzD_AAyapKOohzQA@mail.gmail.com>
Message-ID: <CAC-LwPPCLeX80DLcutwqQi4JDzJazLChivT=mPSXjTi6J19xdg@mail.gmail.com>

FS cannot send color faxes yet?

<snip>
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From kris at kriskinc.com  Thu Aug 22 00:38:29 2013
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 21 Aug 2013 16:38:29 -0400
Subject: [Freeswitch-users] TLS with FreeSWITCH and Kamailio
Message-ID: <CAKDfjge1ZwadN=a8+Kk0ZC6eEX5ykzSdGf8bmPJ1X5DHoWkm-Q@mail.gmail.com>

Hello,

  I'm trying to get TLS cert validation between FreeSWITCH (client)
and Kamailio (server) up and running.  Here's my config/setup so far:

FreeSWITCH 1.2.12 (client) configured with:

    <!-- TLS: disabled by default, set to "true" to enable -->
    <param name="tls" value="true"/>
    <!-- additional bind parameters for TLS -->
    <param name="tls-bind-params" value="transport=tls"/>
    <!-- Port to listen on for TLS requests. (5061 will be used if
unspecified) -->
    <param name="tls-sip-port" value="5081"/>
    <!-- Location of the agent.pem and cafile.pem ssl certificates
(needed for TLS server) -->
    <param name="tls-cert-dir" value="[my cert dir]"/>
    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may
not work with TLSv1 -->
    <param name="tls-version" value="tlsv1"/>
    <param name="tls-verify-policy" value="out"/>

I have a gateway configured with ;transport=tls

Kamailio 4.0 (also tried 4.1, etc) configured with (tls.cfg):

[server:default]
method = TLSv1
verify_certificate = no
require_certificate = yes
private_key = /etc/kamailio/generic-sip.key
certificate = /etc/kamailio/generic-sip.pem
ca_list = /etc/kamailio/generic-cacert.pem
cipher_list = AES

  I'm using my own CA with self-signed certs.  I've verified that they
check out by comparing the modulus on the cert and key pairs and
verifying the CA chain with 'openssl verify ...'.

  When I run without tls-verify-policy=none and require_certificate=no
everything is golden and TLS works all day long.  However, this is
less than ideal and I'd like to at least make sure that my TLS clients
are presenting a valid cert.  Unfortunately when FS tries to connect
to Kamailio it reports the following errors:

ERROR: tls [tls_server.c:1190]: TLS accept:error:140890B2:SSL
routines:SSL3_GET_CLIENT_CERTIFICATE:no certificate returned
ERROR: <core> [tcp_read.c:1275]: ERROR: tcp_read_req: error reading

  What's interesting is that FreeSWITCH reports a successful
registration and seems to exchange OPTIONS pings (over UDP!) with the
remote Kamailio instance.  However, Kamailio does not show the
endpoint as registered (verified with 'kamctl ul show').  That seems
like a bug and worthy of a JIRA but my main concern at this point is
getting TLS with certificate validation up and running.

  Any ideas?  Thanks!

-- 
Kristian Kielhofner


From k4kaleem at gmail.com  Thu Aug 22 00:55:52 2013
From: k4kaleem at gmail.com (kaleem rehman)
Date: Wed, 21 Aug 2013 21:55:52 +0100
Subject: [Freeswitch-users] Conferencing solution - need advice
Message-ID: <CAFWsKqz1t+V1dfX-HzT_adAcR8dC_T5f=EZe7zDnza2SvRYHDw@mail.gmail.com>

Hi All,

I want to implement a conferencing solution using freeswitch with Avaya.

We currently have a multi-country hosted telephony platform, I want to
introduce freeswitch to it for internal use to see how it goes.

I want a freeswitch server in UK, one in Europe, one in Americas and one in
South Africa.

i would like all freeswitch server to talk to each other over SIP over
internal Cloud then i want UK conference users to join to UK freeswitch and
for same conference other countries users to join their local freeswitch
server and talk together.

this way i want to achieve less cloud load and in reality one call between
all servers where users all users from their country will connect to local
freeswitch and that freeswitch links then to a conference with other
country.

please advise on how i can achieve this.

i am open to suggestions and ideas.

regards,
Kaleem
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From krice at freeswitch.org  Thu Aug 22 01:02:21 2013
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 21 Aug 2013 16:02:21 -0500
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <CAC-LwPPCLeX80DLcutwqQi4JDzJazLChivT=mPSXjTi6J19xdg@mail.gmail.com>
Message-ID: <CE3A950D.79234%krice@freeswitch.org>

What do you think he is testing now? heh


On 8/21/13 3:28 PM, "Vik Killa" <vipkilla at gmail.com> wrote:

> FS cannot send color faxes yet?
> 
> <snip>
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From brian at freeswitch.org  Thu Aug 22 01:03:19 2013
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Aug 2013 16:03:19 -0500
Subject: [Freeswitch-users] TLS with FreeSWITCH and Kamailio
In-Reply-To: <CAKDfjge1ZwadN=a8+Kk0ZC6eEX5ykzSdGf8bmPJ1X5DHoWkm-Q@mail.gmail.com>
References: <CAKDfjge1ZwadN=a8+Kk0ZC6eEX5ykzSdGf8bmPJ1X5DHoWkm-Q@mail.gmail.com>
Message-ID: <E4F0379B-E8FF-44B9-9D67-F847B06DD24A@freeswitch.org>

How art thou generated the certs?

On Aug 21, 2013, at 3:38 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:

> Hello,
> 
>  I'm trying to get TLS cert validation between FreeSWITCH (client)
> and Kamailio (server) up and running.  Here's my config/setup so far:
> 
> FreeSWITCH 1.2.12 (client) configured with:
> 
>    <!-- TLS: disabled by default, set to "true" to enable -->
>    <param name="tls" value="true"/>
>    <!-- additional bind parameters for TLS -->
>    <param name="tls-bind-params" value="transport=tls"/>
>    <!-- Port to listen on for TLS requests. (5061 will be used if
> unspecified) -->
>    <param name="tls-sip-port" value="5081"/>
>    <!-- Location of the agent.pem and cafile.pem ssl certificates
> (needed for TLS server) -->
>    <param name="tls-cert-dir" value="[my cert dir]"/>
>    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may
> not work with TLSv1 -->
>    <param name="tls-version" value="tlsv1"/>
>    <param name="tls-verify-policy" value="out"/>
> 
> I have a gateway configured with ;transport=tls
> 
> Kamailio 4.0 (also tried 4.1, etc) configured with (tls.cfg):
> 
> [server:default]
> method = TLSv1
> verify_certificate = no
> require_certificate = yes
> private_key = /etc/kamailio/generic-sip.key
> certificate = /etc/kamailio/generic-sip.pem
> ca_list = /etc/kamailio/generic-cacert.pem
> cipher_list = AES
> 
>  I'm using my own CA with self-signed certs.  I've verified that they
> check out by comparing the modulus on the cert and key pairs and
> verifying the CA chain with 'openssl verify ...'.
> 
>  When I run without tls-verify-policy=none and require_certificate=no
> everything is golden and TLS works all day long.  However, this is
> less than ideal and I'd like to at least make sure that my TLS clients
> are presenting a valid cert.  Unfortunately when FS tries to connect
> to Kamailio it reports the following errors:
> 
> ERROR: tls [tls_server.c:1190]: TLS accept:error:140890B2:SSL
> routines:SSL3_GET_CLIENT_CERTIFICATE:no certificate returned
> ERROR: <core> [tcp_read.c:1275]: ERROR: tcp_read_req: error reading
> 
>  What's interesting is that FreeSWITCH reports a successful
> registration and seems to exchange OPTIONS pings (over UDP!) with the
> remote Kamailio instance.  However, Kamailio does not show the
> endpoint as registered (verified with 'kamctl ul show').  That seems
> like a bug and worthy of a JIRA but my main concern at this point is
> getting TLS with certificate validation up and running.
> 
>  Any ideas?  Thanks!
> 
> -- 
> Kristian Kielhofner
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 brian at freeswitch.org  Thu Aug 22 01:03:43 2013
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Aug 2013 16:03:43 -0500
Subject: [Freeswitch-users] How to disable TCP Keepalive on a TLS
	connection?
In-Reply-To: <CA+T6Pv_SH5mcuYRxHj-2CvM_UAjcQS_NuUKP5E33yy-Ppi2zoQ@mail.gmail.com>
References: <CA+T6Pv_SH5mcuYRxHj-2CvM_UAjcQS_NuUKP5E33yy-Ppi2zoQ@mail.gmail.com>
Message-ID: <70AABE02-0D10-46CA-B43E-901039E14B63@freeswitch.org>

Remove the option.

On Aug 21, 2013, at 12:44 PM, Master Can <a.daydreamer82 at gmail.com> wrote:

> Hello,
> 
> I'm running freeswitch 1.2.10, with tls-only.
> I've tried 2 SIP Useragents now: PhonerLite and CSipSimple.
> 
> My linux server shows me with netstat --timers that both useragents (both server sockets) use keepalive, with a value of 30 seconds.
> 
> How can I disable keepalive on the TCP layer completely? My useragents take care of sending keepalive packets anyway, so Freeswitch does not need to do that. It's not mobile friendly, it's eating up battery power if the useragents keep receiving keepalive every 30 seconds.
> 
> I've tried to set
> <param name="tcp-keepalive" value="disable"/>
> in internal.xml but to no avail. It didn't change a thing. Setting this to 60000 didn't change the output of netstat --timers either.
> 
> Any advice?
> 
> best regards,
> Can
> _______

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From brian at freeswitch.org  Thu Aug 22 01:05:01 2013
From: brian at freeswitch.org (Brian West)
Date: Wed, 21 Aug 2013 16:05:01 -0500
Subject: [Freeswitch-users] in/outbound-use-uuid-as-callid as channel
	variables
In-Reply-To: <CACYJG3J93Ge1Qca+=jacNfP57eWPTKu7kFaovVgkSSbgcbEcEA@mail.gmail.com>
References: <CACYJG3J93Ge1Qca+=jacNfP57eWPTKu7kFaovVgkSSbgcbEcEA@mail.gmail.com>
Message-ID: <1809E751-F75E-458F-8E65-502371AB7FA4@freeswitch.org>

That right there is some Ms. Cleo stuff... Kinda like the time we were asked to execute a command 5 seconds before a call hangs up... LOL

On Aug 20, 2013, at 12:12 PM, Jeff Pyle <jpyle at fidelityvoice.com> wrote:

> Hello,
> 
> Is it possible to set the equivalent of inbound-use-uuid-as-callid or outbound-use-uuid-as-callid per channel from within the dialplan?
> 
> 
> - Jeff

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From kris at kriskinc.com  Thu Aug 22 01:22:32 2013
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Wed, 21 Aug 2013 17:22:32 -0400
Subject: [Freeswitch-users] TLS with FreeSWITCH and Kamailio
In-Reply-To: <E4F0379B-E8FF-44B9-9D67-F847B06DD24A@freeswitch.org>
References: <CAKDfjge1ZwadN=a8+Kk0ZC6eEX5ykzSdGf8bmPJ1X5DHoWkm-Q@mail.gmail.com>
	<E4F0379B-E8FF-44B9-9D67-F847B06DD24A@freeswitch.org>
Message-ID: <CAKDfjgf+gOyp_enk80AgD6ivJcu4TC+Pg0TaRweqMWekBN2ZbA@mail.gmail.com>

Good question!

I've tried a variety of certs, going all the way back to the CA.  I
started with your gentls_cert script and eventually moved to the
openvpn-style "easy-rsa" package.  I will tell you that using
identical certs with a TLS-capable pjsip pjsua client results in a
successful TLS connection to Kamailio (using the same CA cert, client
cert, and client key used in FreeSWITCH).  Of course I'm not changing
the config in Kamailio either.

On Wed, Aug 21, 2013 at 5:03 PM, Brian West <brian at freeswitch.org> wrote:
> How art thou generated the certs?
>
> On Aug 21, 2013, at 3:38 PM, Kristian Kielhofner <kris at kriskinc.com> wrote:
>
>> Hello,
>>
>>  I'm trying to get TLS cert validation between FreeSWITCH (client)
>> and Kamailio (server) up and running.  Here's my config/setup so far:
>>
>> FreeSWITCH 1.2.12 (client) configured with:
>>
>>    <!-- TLS: disabled by default, set to "true" to enable -->
>>    <param name="tls" value="true"/>
>>    <!-- additional bind parameters for TLS -->
>>    <param name="tls-bind-params" value="transport=tls"/>
>>    <!-- Port to listen on for TLS requests. (5061 will be used if
>> unspecified) -->
>>    <param name="tls-sip-port" value="5081"/>
>>    <!-- Location of the agent.pem and cafile.pem ssl certificates
>> (needed for TLS server) -->
>>    <param name="tls-cert-dir" value="[my cert dir]"/>
>>    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may
>> not work with TLSv1 -->
>>    <param name="tls-version" value="tlsv1"/>
>>    <param name="tls-verify-policy" value="out"/>
>>
>> I have a gateway configured with ;transport=tls
>>
>> Kamailio 4.0 (also tried 4.1, etc) configured with (tls.cfg):
>>
>> [server:default]
>> method = TLSv1
>> verify_certificate = no
>> require_certificate = yes
>> private_key = /etc/kamailio/generic-sip.key
>> certificate = /etc/kamailio/generic-sip.pem
>> ca_list = /etc/kamailio/generic-cacert.pem
>> cipher_list = AES
>>
>>  I'm using my own CA with self-signed certs.  I've verified that they
>> check out by comparing the modulus on the cert and key pairs and
>> verifying the CA chain with 'openssl verify ...'.
>>
>>  When I run without tls-verify-policy=none and require_certificate=no
>> everything is golden and TLS works all day long.  However, this is
>> less than ideal and I'd like to at least make sure that my TLS clients
>> are presenting a valid cert.  Unfortunately when FS tries to connect
>> to Kamailio it reports the following errors:
>>
>> ERROR: tls [tls_server.c:1190]: TLS accept:error:140890B2:SSL
>> routines:SSL3_GET_CLIENT_CERTIFICATE:no certificate returned
>> ERROR: <core> [tcp_read.c:1275]: ERROR: tcp_read_req: error reading
>>
>>  What's interesting is that FreeSWITCH reports a successful
>> registration and seems to exchange OPTIONS pings (over UDP!) with the
>> remote Kamailio instance.  However, Kamailio does not show the
>> endpoint as registered (verified with 'kamctl ul show').  That seems
>> like a bug and worthy of a JIRA but my main concern at this point is
>> getting TLS with certificate validation up and running.
>>
>>  Any ideas?  Thanks!
>>
>> --
>> Kristian Kielhofner
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
>
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Kristian Kielhofner


From ccachor at gmail.com  Thu Aug 22 01:53:13 2013
From: ccachor at gmail.com (Chris Cachor)
Date: Wed, 21 Aug 2013 16:53:13 -0500
Subject: [Freeswitch-users] Build error Mac OS X Mountain Lion
Message-ID: <459CF61A-F314-4318-A9B9-489A5979F03E@gmail.com>

Hello,

I'm trying to build Freeswitch on my machine and when I run the make command, I get an error towards the end:

/var/folders/zr/r7177m510r71w3z81959j9sc0000gn/T//ccUHViiz.s:58:suffix or operands invalid for `lea'
/var/folders/zr/r7177m510r71w3z81959j9sc0000gn/T//ccUHViiz.s:63:suffix or operands invalid for `movq'
make[7]: *** [gsm0610_rpe.lo] Error 1
make[6]: *** [all] Error 2
make[5]: *** [all-recursive] Error 1
make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la] Error 2
make[3]: *** [mod_spandsp-all] Error 1
make[2]: *** [all-recursive] Error 1
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2

I've built Freeswitch on this same machine about 2 months ago, so not sure why I'm running into this issue. Has anyone had luck with recent builds on OS X Mountain Lion?

- Chris

From msc at freeswitch.org  Thu Aug 22 03:09:13 2013
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Aug 2013 16:09:13 -0700
Subject: [Freeswitch-users] Conferencing solution - need advice
In-Reply-To: <CAFWsKqz1t+V1dfX-HzT_adAcR8dC_T5f=EZe7zDnza2SvRYHDw@mail.gmail.com>
References: <CAFWsKqz1t+V1dfX-HzT_adAcR8dC_T5f=EZe7zDnza2SvRYHDw@mail.gmail.com>
Message-ID: <CAKzWOxVM0G=XrTfX_ZKi2HGGbWWqDxQz3wkyfTd5yvnpGGOYAw@mail.gmail.com>

Will the four individual conferences always be connected to each other?
-MC


On Wed, Aug 21, 2013 at 1:55 PM, kaleem rehman <k4kaleem at gmail.com> wrote:

> Hi All,
>
> I want to implement a conferencing solution using freeswitch with Avaya.
>
> We currently have a multi-country hosted telephony platform, I want to
> introduce freeswitch to it for internal use to see how it goes.
>
> I want a freeswitch server in UK, one in Europe, one in Americas and one
> in South Africa.
>
> i would like all freeswitch server to talk to each other over SIP over
> internal Cloud then i want UK conference users to join to UK freeswitch and
> for same conference other countries users to join their local freeswitch
> server and talk together.
>
> this way i want to achieve less cloud load and in reality one call between
> all servers where users all users from their country will connect to local
> freeswitch and that freeswitch links then to a conference with other
> country.
>
> please advise on how i can achieve this.
>
> i am open to suggestions and ideas.
>
> regards,
> Kaleem
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From eidevm5 at gmail.com  Thu Aug 22 03:14:33 2013
From: eidevm5 at gmail.com (Peter)
Date: Thu, 22 Aug 2013 09:14:33 +1000
Subject: [Freeswitch-users] Video via a FreeSWITCH SBS
In-Reply-To: <70DC1387-6599-40EA-A576-127237ECC02D@jerris.com>
References: <CABkS=xpmK3G9gRd1Nce5tVeW1fteJUXeUAv3PueGVXsxxCoo7w@mail.gmail.com>
	<70DC1387-6599-40EA-A576-127237ECC02D@jerris.com>
Message-ID: <CABkS=xruz5NbZ7uNog34xWkDHSjyZAvkCL+b+8JxD0OjViQpOw@mail.gmail.com>

I was under the impression that proxy_media was needed.

So if my understanding is correct, given that I already have audio stream
going through the FS SBC, video shouldn't really be any different except it
runs on different RTP ports and is initiated via a reinvite.

Is that correct?


On Wed, Aug 21, 2013 at 11:37 PM, Michael Jerris <mike at jerris.com> wrote:

> It certainly won't work with proxy_media=true, but why would you need to
> use proxy_media=true?
>
> On Aug 21, 2013, at 1:31 AM, Peter <eidevm5 at gmail.com> wrote:
>
> > Is is possible to get video going through a FreeSWITCH SBS where one
> side is using SRTP and the other RTP?
> >
> > I suspect the answer is no, as it requires proxy_media=true to be set,
> although I could be wrong.
> >
> > When I try with proxy_media=true, I always get a:
> >
> > 415 Unsupported Media Type
> >
> > Note that I have all the appropriate codec enabled on the FS servers and
> the SIP clients.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 eidevm5 at gmail.com  Thu Aug 22 03:16:09 2013
From: eidevm5 at gmail.com (Peter)
Date: Thu, 22 Aug 2013 09:16:09 +1000
Subject: [Freeswitch-users] TLS/Freeswitch self signed certs
In-Reply-To: <DD9DE0F8-B525-4B0D-B20B-F181D4765A63@freeswitch.org>
References: <CABkS=xockpPROOhnbSaYJxt0uw2bUtKbyYus_CWqEsWz=sjB6Q@mail.gmail.com>
	<CAC0j6rfkNbJ7KMKMo7VVfGRG0uox3xZjqaw8heX2=HE09TcrEg@mail.gmail.com>
	<CABkS=xqg_Nw=X97szavNccZOT+cHA5PZ2sQ0-Tg=7DW+cqZTrg@mail.gmail.com>
	<CABkS=xqE=3D_PuSVPLWXM8gFzf_7+RPYjj4PTNAUUTU=F1iyog@mail.gmail.com>
	<FC8B9B69-D238-428C-9799-C3EFBA64DDB5@jerris.com>
	<CABkS=xpqry45O7XX7s4R5ib-uwD-wyW7TqAZtTVRujr0XpzaBg@mail.gmail.com>
	<47059AC9-3C1A-4FE0-B82E-275EC3500818@freeswitch.org>
	<DD9DE0F8-B525-4B0D-B20B-F181D4765A63@freeswitch.org>
Message-ID: <CABkS=xqyyF+pCd+KdNBgNz20-_3YFp6R5evPsJYkXXtbE+8OZw@mail.gmail.com>

Note that I'm not a developer so I only have a very basic understanding of
git, jira and patching.

Is there a guide that details how to submit patches?


On Thu, Aug 22, 2013 at 3:04 AM, Brian West <brian at freeswitch.org> wrote:

> I've extracted the info, please re-test the gentls_cert from the generated
> .in file please.
>
> /b
>
> On Aug 21, 2013, at 11:16 AM, Brian West <brian at freeswitch.org> wrote:
>
> > Your diff file on your jira is backwards, can you please patch the .in
> file itself then git diff and attach that.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Thu Aug 22 03:20:38 2013
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 21 Aug 2013 16:20:38 -0700
Subject: [Freeswitch-users] TLS/Freeswitch self signed certs
In-Reply-To: <CABkS=xqyyF+pCd+KdNBgNz20-_3YFp6R5evPsJYkXXtbE+8OZw@mail.gmail.com>
References: <CABkS=xockpPROOhnbSaYJxt0uw2bUtKbyYus_CWqEsWz=sjB6Q@mail.gmail.com>
	<CAC0j6rfkNbJ7KMKMo7VVfGRG0uox3xZjqaw8heX2=HE09TcrEg@mail.gmail.com>
	<CABkS=xqg_Nw=X97szavNccZOT+cHA5PZ2sQ0-Tg=7DW+cqZTrg@mail.gmail.com>
	<CABkS=xqE=3D_PuSVPLWXM8gFzf_7+RPYjj4PTNAUUTU=F1iyog@mail.gmail.com>
	<FC8B9B69-D238-428C-9799-C3EFBA64DDB5@jerris.com>
	<CABkS=xpqry45O7XX7s4R5ib-uwD-wyW7TqAZtTVRujr0XpzaBg@mail.gmail.com>
	<47059AC9-3C1A-4FE0-B82E-275EC3500818@freeswitch.org>
	<DD9DE0F8-B525-4B0D-B20B-F181D4765A63@freeswitch.org>
	<CABkS=xqyyF+pCd+KdNBgNz20-_3YFp6R5evPsJYkXXtbE+8OZw@mail.gmail.com>
Message-ID: <CAKzWOxUjwDmc-ZqHxeVkiEoMqOui5qAMvwYWjx31qYe7W=LQeA@mail.gmail.com>

I'd start with this page:



On Wed, Aug 21, 2013 at 4:16 PM, Peter <eidevm5 at gmail.com> wrote:

> Note that I'm not a developer so I only have a very basic understanding of
> git, jira and patching.
>
> Is there a guide that details how to submit patches?
>
>
> On Thu, Aug 22, 2013 at 3:04 AM, Brian West <brian at freeswitch.org> wrote:
>
>> I've extracted the info, please re-test the gentls_cert from the
>> generated .in file please.
>>
>> /b
>>
>> On Aug 21, 2013, at 11:16 AM, Brian West <brian at freeswitch.org> wrote:
>>
>> > Your diff file on your jira is backwards, can you please patch the .in
>> file itself then git diff and attach that.
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jleung at v10networks.ca  Thu Aug 22 03:26:10 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Wed, 21 Aug 2013 16:26:10 -0700
Subject: [Freeswitch-users] Error with SQLDB on FS startup
In-Reply-To: <CAPNTnnrg6QBB5x0Zahjp=7QsLy_54_CVE_GQ6kLgbJaQOFTRDw@mail.gmail.com>
References: <CAPNTnnrg6QBB5x0Zahjp=7QsLy_54_CVE_GQ6kLgbJaQOFTRDw@mail.gmail.com>
Message-ID: <A70AC0C1B7E7FD478A6BF19705461AA0C8C9@IMBER.smb.curriegrad2004.ca>

If there's no tables when FS starts up on a SQLite database, it'll
create it automatically. You can safely ignore this.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Guillermo Ruiz Camauer
Sent: Wednesday, August 21, 2013 11:25 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Error with SQLDB on FS startup

 

I am getting the following errors when starting up FS:

 

[root at Freeswitch freeswitch]# /usr/local/freeswitch/bin/freeswitch
2013-08-21 15:13:55.201104 [INFO] switch_event.c:596 Activate Eventing
Engine.
2013-08-21 15:13:55.201308 [WARNING] switch_event.c:570 Create
additional event dispatch thread 0
2013-08-21 15:13:55.262832 [INFO] switch_nat.c:420 Scanning for NAT
2013-08-21 15:13:55.262950 [DEBUG] switch_nat.c:170 Checking for PMP 1/5
2013-08-21 15:13:55.513013 [DEBUG] switch_nat.c:170 Checking for PMP 2/5
2013-08-21 15:13:56.012990 [DEBUG] switch_nat.c:170 Checking for PMP 3/5
2013-08-21 15:13:57.013009 [DEBUG] switch_nat.c:170 Checking for PMP 4/5
2013-08-21 15:13:59.013009 [DEBUG] switch_nat.c:170 Checking for PMP 5/5
2013-08-21 15:14:03.013009 [ERR] switch_nat.c:201 Error checking for PMP
[general error]
2013-08-21 15:14:03.013039 [DEBUG] switch_nat.c:425 Checking for UPnP
2013-08-21 15:14:06.066363 [INFO] switch_nat.c:441 No PMP or UPnP NAT
devices detected!
2013-08-21 15:14:06.068084 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR
[no such table: channels]
drop table channels
2013-08-21 15:14:06.068108 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR
[no such table: calls]
drop table calls
2013-08-21 15:14:06.068125 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR
[no such table: detailed_calls]
drop view detailed_calls
2013-08-21 15:14:06.068140 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR
[no such table: basic_calls]
drop view basic_calls
2013-08-21 15:14:06.068155 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR
[no such table: interfaces]
drop table interfaces
2013-08-21 15:14:06.068169 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR
[no such table: tasks]
drop table tasks
2013-08-21 15:14:06.081283 [CONSOLE] switch_core.c:1293 Created ip list
lan default (allow)
2013-08-21 15:14:06.081302 [CONSOLE] switch_core.c:1293 Created ip list
domains default (deny)
...

 

I am on FreeSWITCH Version 1.2.12+git~20130821T172756Z~c73179c6ad (git
c73179c 2013-08-21 17:27:56Z). (just ran make current, but it was
happenning before also).

 

FS continues and loads.  What are the possible side effects of these
errors?  Is there something very wrong with my setup?

 


-- 
Guillermo Ruiz Camauer

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From karl at xtronics.com  Thu Aug 22 03:26:28 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Wed, 21 Aug 2013 18:26:28 -0500
Subject: [Freeswitch-users] CN and VAD for freetdm (ftdm)
In-Reply-To: <459CF61A-F314-4318-A9B9-489A5979F03E@gmail.com>
References: <459CF61A-F314-4318-A9B9-489A5979F03E@gmail.com>
Message-ID: <52154CA4.8060408@xtronics.com>

I'm testing outbound calls over a land line from a  Digium, Inc. Wildcard AEX410 card and I get the 
beginning of words cut off.  This appears to be something like VAD going on, but I don't see any 
settings for ftdm.

My understanding is that VAD and CNG are off by default - and as per the demo?


And the silence is too silent (no CNG) .. I don't see anything about CNG or VAD in any of the ftdm 
setup info?

Could this be from echo cancellation?




--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

Let us live so that when we come to die even the undertaker will be sorry.
-- Mark Twain
--------------------------------------------------------------------------------


From lists at kavun.ch  Thu Aug 22 03:45:23 2013
From: lists at kavun.ch (Emrah)
Date: Thu, 22 Aug 2013 01:45:23 +0200
Subject: [Freeswitch-users] Cannot load mod_spandsp.so: undefined
	symbol: lzma_stream_encoder
In-Reply-To: <9FDF70D3-8552-49A8-A94A-446F2901852F@freeswitch.org>
References: <8E465A5B-2B1E-42B1-A6C5-A445AAB2EA8A@kavun.ch>
	<5214D112.9070607@coppice.org>
	<414A04D3-B216-4509-81DA-08D0AC3F2E45@kavun.ch>
	<9FDF70D3-8552-49A8-A94A-446F2901852F@freeswitch.org>
Message-ID: <F97F3B1A-6354-41FA-B424-AA309C6CD772@kavun.ch>

Yes I did. It compiles fine and than can't load.

Thanks
On Aug 21, 2013, at 6:56 PM, Brian West <brian at freeswitch.org> wrote:

> 
> Have you tried to 'make spandsp-reconf' to make sure your spandsp source is in order?
> 
> /b
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ms at mstn.com  Thu Aug 22 04:00:39 2013
From: ms at mstn.com (Jose Suero)
Date: Wed, 21 Aug 2013 20:00:39 -0400
Subject: [Freeswitch-users] You guys are terrible
Message-ID: <f6cffc7f0becfe253c8bfe4761781315@mstn.com>

I'm swimming in a freeswitch log looking for a problem when all of the 
sudden this comes up:

2013-08-21 19:51:08.002385 [DEBUG] sofia.c:6928 1 .. 2 .. Freddie's 
commin' for you...

Didn't find the freeswitch problem, but now have a machete masked 
fellow on his way :-)


Made me smile




From bdfoster at davri.com  Thu Aug 22 04:18:35 2013
From: bdfoster at davri.com (Brian Foster)
Date: Wed, 21 Aug 2013 20:18:35 -0400
Subject: [Freeswitch-users] You guys are terrible
In-Reply-To: <f6cffc7f0becfe253c8bfe4761781315@mstn.com>
References: <f6cffc7f0becfe253c8bfe4761781315@mstn.com>
Message-ID: <CAHgk4BU+OMEMxGgZKn0pJWSUqTiLiOcF6iqFRq-K4uQ1E3o72g@mail.gmail.com>

That's whay happens when Sofia gets all bitchy and cops an attitude...

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 21, 2013 8:05 PM, "Jose Suero" <ms at mstn.com> wrote:

> I'm swimming in a freeswitch log looking for a problem when all of the
> sudden this comes up:
>
> 2013-08-21 19:51:08.002385 [DEBUG] sofia.c:6928 1 .. 2 .. Freddie's
> commin' for you...
>
> Didn't find the freeswitch problem, but now have a machete masked
> fellow on his way :-)
>
>
> Made me smile
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at davri.com  Thu Aug 22 04:21:38 2013
From: bdfoster at davri.com (Brian Foster)
Date: Wed, 21 Aug 2013 20:21:38 -0400
Subject: [Freeswitch-users] You guys are terrible
In-Reply-To: <f6cffc7f0becfe253c8bfe4761781315@mstn.com>
References: <f6cffc7f0becfe253c8bfe4761781315@mstn.com>
Message-ID: <CAHgk4BXPSFda7TABZKKeoB3DkaAXSDsoJfnnk9k_snSmKR-cFQ@mail.gmail.com>

Btw look in the source code around line 6928 might give you a clue as to
what's happening.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 21, 2013 8:05 PM, "Jose Suero" <ms at mstn.com> wrote:

> I'm swimming in a freeswitch log looking for a problem when all of the
> sudden this comes up:
>
> 2013-08-21 19:51:08.002385 [DEBUG] sofia.c:6928 1 .. 2 .. Freddie's
> commin' for you...
>
> Didn't find the freeswitch problem, but now have a machete masked
> fellow on his way :-)
>
>
> Made me smile
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jleung at v10networks.ca  Thu Aug 22 05:54:00 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Wed, 21 Aug 2013 18:54:00 -0700
Subject: [Freeswitch-users] You guys are terrible
In-Reply-To: <CAHgk4BXPSFda7TABZKKeoB3DkaAXSDsoJfnnk9k_snSmKR-cFQ@mail.gmail.com>
References: <f6cffc7f0becfe253c8bfe4761781315@mstn.com>
	<CAHgk4BXPSFda7TABZKKeoB3DkaAXSDsoJfnnk9k_snSmKR-cFQ@mail.gmail.com>
Message-ID: <A70AC0C1B7E7FD478A6BF19705461AA0C8CA@IMBER.smb.curriegrad2004.ca>

I've seen worse on what sofia sometimes does when it wants to quit on
us.

 

More like a suicide note from sofia is what I've seen before

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Brian Foster
Sent: Wednesday, August 21, 2013 5:22 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] You guys are terrible

 

Btw look in the source code around line 6928 might give you a clue as to
what's happening. 

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.

On Aug 21, 2013 8:05 PM, "Jose Suero" <ms at mstn.com> wrote:

I'm swimming in a freeswitch log looking for a problem when all of the
sudden this comes up:

2013-08-21 19:51:08.002385 [DEBUG] sofia.c:6928 1 .. 2 .. Freddie's
commin' for you...

Didn't find the freeswitch problem, but now have a machete masked
fellow on his way :-)


Made me smile



________________________________________________________________________
_
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 steveu at coppice.org  Thu Aug 22 06:32:50 2013
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 22 Aug 2013 10:32:50 +0800
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <CAC-LwPPCLeX80DLcutwqQi4JDzJazLChivT=mPSXjTi6J19xdg@mail.gmail.com>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
	<CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
	<B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
	<CAHgk4BWdp9iQHsZfQmNWu0JLzidQQCtGn0kzD_AAyapKOohzQA@mail.gmail.com>
	<CAC-LwPPCLeX80DLcutwqQi4JDzJazLChivT=mPSXjTi6J19xdg@mail.gmail.com>
Message-ID: <52157852.7010604@coppice.org>

On 08/22/2013 04:28 AM, Vik Killa wrote:
> FS cannot send color faxes yet?
>
Very few computer FAX platforms can send or receive colour FAXes. As of 
this week FS can, but the support should be treated as experimental at 
this stage.

It seems like colour FAX should be trivial, if you have error corrected 
FAX support, but it isn't. The developers of the colour FAX standard 
chose to use JPEG compression, but with a very unusual colour 
representation. This means you need to develop a messy bunch of code if 
you want the user to be able to work with conventional colour image 
files on their computers. It is this support which we are shaking out 
now. When its looking stable I will add the relevant information about 
controlling colour FAXes to the wiki pages.

Regards,
Steve


From nandy1925 at gmail.com  Thu Aug 22 07:22:36 2013
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 22 Aug 2013 11:22:36 +0800
Subject: [Freeswitch-users] Transfer / Forward calls - from an operator
 to an extension after the call is picked
In-Reply-To: <5213A356.4030805@yahoo.com>
References: <1376827283.11657.YahooMailNeo@web160501.mail.bf1.yahoo.com>
	<CAHKqMb5SK8V5h-CYf8uaivj_LkvMR-xjQ1k+gG7UA_0z_5g44w@mail.gmail.com>
	<521258E8.90900@yahoo.com>
	<CAHKqMb4Xzwswcr3PtbA5OT-AVGSNZ4+BiiW5v4B8e0=g1o1PUQ@mail.gmail.com>
	<5213A356.4030805@yahoo.com>
Message-ID: <CAHKqMb5KpNWPSAT_3j5dT=AAbezZSFLQMR_6vm=LL30X_sHhJQ@mail.gmail.com>

Hi Ravi,

The Local_extension entries in default.xml dialplan is meant for analog
phones only where the star(*) key is monitored. In IP phones, it's all done
in the SIP messages.

/Nandy

On Wed, Aug 21, 2013 at 1:11 AM, Ravi <sravi123 at yahoo.com> wrote:

>  Thanks again, Nandy. The TRANSFER and HOLD buttons have been there for
> long, but went unnoticed :) For some reason, I was thinking that there
> should be some provision in freeswitch to transfer calls. But it has turned
> out to be much simpler. I have Grandstream phones, and I am able to
> transfer calls.
> Since this solution worked, I have not checked the default.xml for call
> transfers. I shall check them soon.
>
> Thanks.
> Ravi
>
>
>
> On 20/08/13 1:53 AM, Nandy Dagondon wrote:
>
> Hi Ravi,
>
>  Nice to know it helped. What  you're describing is Supervised Call
> Transfer. Locate your phone's HOLD and TRANSFER buttons.
> 1. Press HOLD to place the caller on-hold. She'll enjoy the music :-)
> 2. You'll hear a dialtone. Dial extension 1010. Talk to the 3rd party.
> 3. If she accepts it, press the TRANSFER button then hang-up.
> 4. If not, press the HOLD button to resume talking to the caller.
>
>  For analog phones, the vanilla default.xml dialplan has an example in
> the Local extension section what keys are bound to call transfers.
>
>  /Nandy
>
>
> On Tue, Aug 20, 2013 at 1:42 AM, Ravi <sravi123 at yahoo.com> wrote:
>
>>  Thank you Nandy for the response. It helped, and I am able to route
>> calls.
>>
>> I am able to transfer the call, when it comes in, to the operator through
>> Freeswitch. After the operator takes the call, the operator talks to the
>> caller for a couple of minutes, and then has to transfer the call to one of
>> the extensions say 1010. How is that done ? Or as a simple case, how can i
>> transfer a call that I have received to another extension ?
>>
>> Perhaps this is very easy, but I am not sure where to find this
>> information.
>>
>> Thanks.
>> Ravi
>>
>>
>> On 19/08/13 9:24 AM, Nandy Dagondon wrote:
>>
>>  Hi,
>>
>> 1. Copy your working dialplan entry.
>>  2. Add <condition field="caller_id_number" data="4302000"/>
>>  3. Change the bridge/transfer application to connect to the operator
>>
>>  Take note to place the above entry ahead of your working entry. I hope
>> this helps.
>>  /Nandy
>>
>>
>> On Sun, Aug 18, 2013 at 8:01 PM, Ravi <sravi123 at yahoo.com> wrote:
>>
>>>  Hello Everyone !
>>>
>>> I have successfully configured my PRI connection. I am able to send and
>>> receive calls. Now I am trying to have an operator handle one number - say
>>> 4302000. When any call is received the operator will talk to the caller and
>>> need to transfer the call to an extension. I tried looking at dialplan
>>> documentation, and I think I find only instances where the transfer is done
>>> within the program and not when some one picks up the call.
>>>
>>> Can some one please point me to right place/ documents where I can find
>>> information regarding this.
>>>
>>> I did find a link with some similar request, but the information is
>>> limited.
>>>
>>> http://lists.freeswitch.org/pipermail/freeswitch-users/2012-October/088579.html
>>>
>>> Thanks for your help.
>>> Ravi
>>>
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>>
>> 
>>
>> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com
>
> 
>
> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nandy1925 at gmail.com  Thu Aug 22 07:36:16 2013
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 22 Aug 2013 11:36:16 +0800
Subject: [Freeswitch-users] Original vs Transfer Call Channel Variables
Message-ID: <CAHKqMb6UCmeEsTqi1-HOH1iO=oEwaPY82Xy0vRkW6zK1RHEAxg@mail.gmail.com>

Hi to everybody,

Which channel variables to examine to differentiate an original vs
transferred (REFERed) call?

Thanks,
/Nandy
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From karl at xtronics.com  Thu Aug 22 07:36:56 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Wed, 21 Aug 2013 22:36:56 -0500
Subject: [Freeswitch-users] CN and VAD for freetdm (ftdm)
In-Reply-To: <52154CA4.8060408@xtronics.com>
References: <459CF61A-F314-4318-A9B9-489A5979F03E@gmail.com>
	<52154CA4.8060408@xtronics.com>
Message-ID: <52158758.3030908@xtronics.com>

This audio problem turned out to be due to linphone-3.5.2

Not seeing this in the 3.6 version - but there is no backport available..


--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

All blame is a waste of time. No matter how much
fault you find with another, and regardless of how
much you blame him, it will not change you.
Wayne Dyer
--------------------------------------------------------------------------------


From nandy1925 at gmail.com  Thu Aug 22 07:58:41 2013
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Thu, 22 Aug 2013 11:58:41 +0800
Subject: [Freeswitch-users] How to compile and install the prerequisites
 gsmlib & libctb for freeswitch to use gsm modem
In-Reply-To: <CAMvhjJtuBEySOdRDjGSugJZYRRwPOUprUKNMWc2JZuyj8xajNg@mail.gmail.com>
References: <CAMvhjJtuBEySOdRDjGSugJZYRRwPOUprUKNMWc2JZuyj8xajNg@mail.gmail.com>
Message-ID: <CAHKqMb7rZxFvGLrTGi3M_-s+y+ciPiXJdLuh4j1cksPJxtkZLg@mail.gmail.com>

Hi Ashish,

Go to FS source directory, then:
$cd src/mod/endpoints/mod_gsmopen

Proceed building the libraries per instruction in
http://wiki.freeswitch.org/wiki/GSMopen#Prerequisites

/Nandy

I have seen the prerequisites of making freeswitch use gsm modem to make
call but cannot actually figure out how to compile and install gsmlib &
libctb...kindly help

>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mehroz.ashraf85 at gmail.com  Thu Aug 22 11:25:11 2013
From: mehroz.ashraf85 at gmail.com (mehroz)
Date: Thu, 22 Aug 2013 00:25:11 -0700 (PDT)
Subject: [Freeswitch-users] TLS setup failed (after changing key type)
In-Reply-To: <3093BAC3-571B-438F-96D7-8CE14091399A@freeswitch.org>
References: <1371825449535-7592067.post@n2.nabble.com>
	<3093BAC3-571B-438F-96D7-8CE14091399A@freeswitch.org>
Message-ID: <1377156311500-7594190.post@n2.nabble.com>

Thanks Brian!

Ill follow that from now onwards. Jira post have been updated with the diff
patch as requested.
I am actually tryin to do SUITE B standard encryption. 
Here <http://en.wikipedia.org/wiki/NSA_Suite_B_Cryptography>   it is!

Please comment ..



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/TLS-setup-failed-after-changing-key-type-tp7592067p7594190.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From k4kaleem at gmail.com  Thu Aug 22 11:45:33 2013
From: k4kaleem at gmail.com (Kaleem)
Date: Thu, 22 Aug 2013 08:45:33 +0100
Subject: [Freeswitch-users] Conferencing solution - need advice
In-Reply-To: <CAKzWOxVM0G=XrTfX_ZKi2HGGbWWqDxQz3wkyfTd5yvnpGGOYAw@mail.gmail.com>
References: <CAFWsKqz1t+V1dfX-HzT_adAcR8dC_T5f=EZe7zDnza2SvRYHDw@mail.gmail.com>
	<CAKzWOxVM0G=XrTfX_ZKi2HGGbWWqDxQz3wkyfTd5yvnpGGOYAw@mail.gmail.com>
Message-ID: <A3579698-5452-48CD-B400-29408B5AAD9F@gmail.com>

Hi Mike, thanks for looking into this, yes all servers will be active all time running on high end virtual environment
Thanks


Sent from my iPhone

On 22 Aug 2013, at 00:09, Michael Collins <msc at freeswitch.org> wrote:

> Will the four individual conferences always be connected to each other?
> -MC
> 
> 
> On Wed, Aug 21, 2013 at 1:55 PM, kaleem rehman <k4kaleem at gmail.com> wrote:
>> Hi All,
>>  
>> I want to implement a conferencing solution using freeswitch with Avaya.
>>  
>> We currently have a multi-country hosted telephony platform, I want to introduce freeswitch to it for internal use to see how it goes.
>>  
>> I want a freeswitch server in UK, one in Europe, one in Americas and one in South Africa.
>>  
>> i would like all freeswitch server to talk to each other over SIP over internal Cloud then i want UK conference users to join to UK freeswitch and for same conference other countries users to join their local freeswitch server and talk together.
>>  
>> this way i want to achieve less cloud load and in reality one call between all servers where users all users from their country will connect to local freeswitch and that freeswitch links then to a conference with other country.
>>  
>> please advise on how i can achieve this.
>>  
>> i am open to suggestions and ideas.
>>  
>> regards,
>> Kaleem
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>> 
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
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From juanito1982 at gmail.com  Thu Aug 22 12:21:25 2013
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Thu, 22 Aug 2013 10:21:25 +0200
Subject: [Freeswitch-users] Not able to send SIP chat
Message-ID: <CAHLYDCvchcYRSgiOXM1YwpNvRbkoP3w8Mc3zO7xxH7DT8Or=EQ@mail.gmail.com>

Hello,

   I am doing some test to send SIP MESSAGE using chat api without success.

SIP domain is 10.3.1.231

Destination reg info is:

Call-ID:        E950F7906ACA4DC4B2CB5ADEC4C4882D0x0a030191
User:           cabinacr1 at 10.3.1.231
Contact:        "user" <sip:cabinacr1 at 10.3.1.145>
Agent:          SJphone/1.65.377a (SJ Labs)
Status:         Registered(UDP)(unknown) EXP(2013-08-22 10:23:00)
EXPSECS(294)
Host:           fscentos6
IP:             10.3.1.145
Port:           5060
Auth-User:      cabinacr1
Auth-Realm:     10.3.1.231
MWI-Account:    cabinacr1 at 10.3.1.231

Command 'chat sip|cabinacr2 at 10.3.1.231|cabinacr1 at 10.3.1.231|Hola' returns
'Error! Message Not Sent'

Command 'chat sip|sip:cabinacr2 at 10.3.1.231|sip:cabinacr1 at 10.3.1.231|Hola'
returns 'sofia_presence.c:4309 Not sending message to ourselves!'

What is the correct way to send the message?

Regards
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From ssinyagin at yahoo.com  Thu Aug 22 13:23:49 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Thu, 22 Aug 2013 02:23:49 -0700 (PDT)
Subject: [Freeswitch-users] Conferencing solution - need advice
In-Reply-To: <A3579698-5452-48CD-B400-29408B5AAD9F@gmail.com>
References: <CAFWsKqz1t+V1dfX-HzT_adAcR8dC_T5f=EZe7zDnza2SvRYHDw@mail.gmail.com>	<CAKzWOxVM0G=XrTfX_ZKi2HGGbWWqDxQz3wkyfTd5yvnpGGOYAw@mail.gmail.com>
	<A3579698-5452-48CD-B400-29408B5AAD9F@gmail.com>
Message-ID: <1377163429.38648.YahooMailNeo@web126205.mail.ne1.yahoo.com>

this would require active SIP channels between the conference instances, but not counting as participants. 

Also some development work needs to be done to distribute the conference control between the servers:
-- moderator joined, everyone can talk
-- moderator control, like kicking off or muting participants
-- moderator ending and shutting down the conference session
-- total count and listing of all participants

This sounds like ~50 to 100 hours of developer's work if you need a full conferencing functionality across multiple servers.









>________________________________
> From: Kaleem <k4kaleem at gmail.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Thursday, August 22, 2013 9:45 AM
>Subject: Re: [Freeswitch-users] Conferencing solution - need advice
> 
>
>
>Hi Mike, thanks for looking into this, yes all servers will be active all time running on high end virtual environment
>Thanks
>
>
>Sent from my iPhone
>
>On 22 Aug 2013, at 00:09, Michael Collins <msc at freeswitch.org> wrote:
>
>
>Will the four individual conferences always be connected to each other?
>>-MC
>>
>>
>>
>>
>>On Wed, Aug 21, 2013 at 1:55 PM, kaleem rehman <k4kaleem at gmail.com> wrote:
>>
>>Hi All,
>>>?
>>>I want to implement a conferencing solution using freeswitch with Avaya.
>>>?
>>>We currently have a multi-country hosted telephony platform, I want to introduce freeswitch to it for internal use to see how it goes.
>>>?
>>>I want a freeswitch server in UK, one in Europe, one in Americas and one in South Africa.
>>>?
>>>i would like all freeswitch server to talk to each other over SIP over internal Cloud then i want UK conference users to join to UK freeswitch and for same conference other countries users to join their local freeswitch server and talk together.
>>>?
>>>this way i want to achieve less cloud load and in reality one call between all servers where users all users from their country will connect to local freeswitch and that freeswitch links then to a conference with other country.
>>>?
>>>please advise on how i can achieve this.
>>>?
>>>i am open to suggestions and ideas.
>>>?
>>>regards,
>>>Kaleem
>>>_________________________________________________________________________
>>>Professional FreeSWITCH Consulting Services:
>>>consulting at freeswitch.org
>>>http://www.freeswitchsolutions.com
>>>
>>>
>>>
>>>
>>>Official FreeSWITCH Sites
>>>http://www.freeswitch.org
>>>http://wiki.freeswitch.org
>>>http://www.cluecon.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
>>>
>>>
>>
>>
>>-- 
>>Michael S Collins
>>Twitter: @mercutioviz
>>http://www.FreeSWITCH.org
>>http://www.ClueCon.com
>>http://www.OSTAG.org
>>
>>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 beskrovny at gmail.com  Thu Aug 22 17:05:48 2013
From: beskrovny at gmail.com (Veniamin (Benjamin) Beskrovny)
Date: Thu, 22 Aug 2013 15:05:48 +0200
Subject: [Freeswitch-users] Conferencing solution - need advice
In-Reply-To: <1377163429.38648.YahooMailNeo@web126205.mail.ne1.yahoo.com>
References: <CAFWsKqz1t+V1dfX-HzT_adAcR8dC_T5f=EZe7zDnza2SvRYHDw@mail.gmail.com>
	<CAKzWOxVM0G=XrTfX_ZKi2HGGbWWqDxQz3wkyfTd5yvnpGGOYAw@mail.gmail.com>
	<A3579698-5452-48CD-B400-29408B5AAD9F@gmail.com>
	<1377163429.38648.YahooMailNeo@web126205.mail.ne1.yahoo.com>
Message-ID: <CAPhuNs_KvmL_WH6_rFXonV-_PT0OpBOFBW6km5MB-aNYf8REpQ@mail.gmail.com>

Indeed, Stan is right. I would rather concentrate on joining one
(maybe dedicated instance of FS somewhere in the middle) instead of
keeping full mesh of talkers, thus eating bandwidth and resources...

On 8/22/13, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
> this would require active SIP channels between the conference instances, but
> not counting as participants.
>
> Also some development work needs to be done to distribute the conference
> control between the servers:
> -- moderator joined, everyone can talk
> -- moderator control, like kicking off or muting participants
> -- moderator ending and shutting down the conference session
> -- total count and listing of all participants
>
> This sounds like ~50 to 100 hours of developer's work if you need a full
> conferencing functionality across multiple servers.
>
>
>
>
>
>
>
>
>
>>________________________________
>> From: Kaleem <k4kaleem at gmail.com>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>Sent: Thursday, August 22, 2013 9:45 AM
>>Subject: Re: [Freeswitch-users] Conferencing solution - need advice
>>
>>
>>
>>Hi Mike, thanks for looking into this, yes all servers will be active all
>> time running on high end virtual environment
>>Thanks
>>
>>
>>Sent from my iPhone
>>
>>On 22 Aug 2013, at 00:09, Michael Collins <msc at freeswitch.org> wrote:
>>
>>
>>Will the four individual conferences always be connected to each other?
>>>-MC
>>>
>>>
>>>
>>>
>>>On Wed, Aug 21, 2013 at 1:55 PM, kaleem rehman <k4kaleem at gmail.com> wrote:
>>>
>>>Hi All,
>>>>
>>>>I want to implement a conferencing solution using freeswitch with Avaya.
>>>>
>>>>We currently have a multi-country hosted telephony platform, I want to
>>>> introduce freeswitch to it for internal use to see how it goes.
>>>>
>>>>I want a freeswitch server in UK, one in Europe, one in Americas and one
>>>> in South Africa.
>>>>
>>>>i would like all freeswitch server to talk to each other over SIP over
>>>> internal Cloud then i want UK conference users to join to UK freeswitch
>>>> and for same conference other countries users to join their local
>>>> freeswitch server and talk together.
>>>>
>>>>this way i want to achieve less cloud load and in reality one call
>>>> between all servers where users all users from their country will
>>>> connect to local freeswitch and that freeswitch links then to a
>>>> conference with other country.
>>>>
>>>>please advise on how i can achieve this.
>>>>
>>>>i am open to suggestions and ideas.
>>>>
>>>>regards,
>>>>Kaleem
>>>>_________________________________________________________________________
>>>>Professional FreeSWITCH Consulting Services:
>>>>consulting at freeswitch.org
>>>>http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>>
>>>>Official FreeSWITCH Sites
>>>>http://www.freeswitch.org
>>>>http://wiki.freeswitch.org
>>>>http://www.cluecon.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
>>>>
>>>>
>>>
>>>
>>>--
>>>Michael S Collins
>>>Twitter: @mercutioviz
>>>http://www.FreeSWITCH.org
>>>http://www.ClueCon.com
>>>http://www.OSTAG.org
>>>
>>>
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
>>consulting at freeswitch.org
>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
>>http://www.freeswitch.org
>>http://wiki.freeswitch.org
>>http://www.cluecon.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 jayachar88 at gmail.com  Thu Aug 22 10:54:27 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Thu, 22 Aug 2013 12:24:27 +0530
Subject: [Freeswitch-users] make sounds-install,
	only downloads tar.gz sound file ?
Message-ID: <CAHDL-syn4wLXNA1H8QK2fOtq=F_428Tbjh8uq7AvDNtHqBaSCQ@mail.gmail.com>

Hi,

Built the latest Git HEAD, following the Wiki instructions (I think
meticulously). After some minor hiccups (filed a bug report in JIRA), I
think I have the all-good, see the "Come to Cluecon..." message.

I tried doing 2 things post that, i.e.:

# make sounds-install

followed by

# make samples

while the latter (installing sample config) went fine, the sounds-install
step just seemed to use wget (I think) to download
freeswitch-sounds-en-us-callie-8000-1.0.25.tar.gz file, and didn't seem to
do anything else. Maybe, it does so non-interactively. Could anyone confirm
?

My aim is to setup a SIP-GSM gateway using mod_gsmopen, and I had taken
care to enable it in modules.conf and ensured that it was built fine
(that's where I faced the minor hiccups). I hope I am good to go (of
course, need to figure out a ton of the configuration and platform
management things).

cheers,
Jay
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From a.daydreamer82 at gmail.com  Thu Aug 22 13:17:25 2013
From: a.daydreamer82 at gmail.com (Master Can)
Date: Thu, 22 Aug 2013 11:17:25 +0200
Subject: [Freeswitch-users] How to disable TCP Keepalive on a TLS
	connection?
In-Reply-To: <70AABE02-0D10-46CA-B43E-901039E14B63@freeswitch.org>
References: <CA+T6Pv_SH5mcuYRxHj-2CvM_UAjcQS_NuUKP5E33yy-Ppi2zoQ@mail.gmail.com>
	<70AABE02-0D10-46CA-B43E-901039E14B63@freeswitch.org>
Message-ID: <CA+T6Pv8OGjBCZJ0khfQE-MH5BKRuZ7pc22M28SZk8XMVpBow+g@mail.gmail.com>

Originally the option was not there - so I tried running Freeswitch without
the option already. That didn't influence tcp keepalives though...


2013/8/21 Brian West <brian at freeswitch.org>

> Remove the option.
>
> On Aug 21, 2013, at 12:44 PM, Master Can <a.daydreamer82 at gmail.com> wrote:
>
> > Hello,
> >
> > I'm running freeswitch 1.2.10, with tls-only.
> > I've tried 2 SIP Useragents now: PhonerLite and CSipSimple.
> >
> > My linux server shows me with netstat --timers that both useragents
> (both server sockets) use keepalive, with a value of 30 seconds.
> >
> > How can I disable keepalive on the TCP layer completely? My useragents
> take care of sending keepalive packets anyway, so Freeswitch does not need
> to do that. It's not mobile friendly, it's eating up battery power if the
> useragents keep receiving keepalive every 30 seconds.
> >
> > I've tried to set
> > <param name="tcp-keepalive" value="disable"/>
> > in internal.xml but to no avail. It didn't change a thing. Setting this
> to 60000 didn't change the output of netstat --timers either.
> >
> > Any advice?
> >
> > best regards,
> > Can
> > _______
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 iland at cs.ucsb.edu  Thu Aug 22 15:56:53 2013
From: iland at cs.ucsb.edu (Danny Iland)
Date: Thu, 22 Aug 2013 04:56:53 -0700
Subject: [Freeswitch-users] Mod_skypopen ' No Skype client logged in as
	'user' has been found'
Message-ID: <CAC_uT4C+45dv8iYRRgQ6uTuFZk+pUza_oqCMOjXpmcZFh3jvWg@mail.gmail.com>

Hi,

When I load mod_skypopen in freeswitch, after running
./start_skype_clients.sh, I get a message that 'No Skype client logged in
as 'user' has been found'.

I followed the Wiki install on Ubuntu 12 x64, then launched per the wiki's
instructions. I have connected to the Skype client as each user via ssh -X
to give skypopen API permissions in Skype, following
README.skypopen_auth. I would appreciate a bit of clarification with
regards to where mod_skypopen looks for each user's Skype config files, and
what needs to be where in the .Skype directory. It would be particularly
nice to see a file tree of a multi-user skypopen install's .Skype directory.

I have checked all the obvious problems I could think of before coming to
the list. Obviously I have set my own usernames and passwords in
start_skype_clients.sh and usernames in skypopen.conf.xml

Below is a log of the behavior. The second is to show that Skype is
running, along with Xvfb.

Any help is appreciated.

Thanks,
Danny Iland
Graduate Student Researcher, University of California Santa Barbara
iland at cs.ucsb.edu



./start_skype_clients.sh
[snip]
freeswitch

freeswitch at g> log level 9

+OK

freeswitch at g> load mod_skypopen
2013-08-22 05:02:15.023031 [INFO] mod_enum.c:876 ENUM Reloaded
2013-08-22 05:02:15.023031 [INFO] switch_time.c:1191 Timezone reloaded 530
definitions
2013-08-22 05:02:15.043058 [WARNING] mod_skypopen.c:1762
[543dc3c|22dd4bf] [WARNINGA     1762 ][skype101       ][IDLE,IDLE] STARTING
interface_id=1
2013-08-22 05:02:15.243058 [NOTICE] mod_skypopen.c:1791
 [543dc3c|22dd4bf] [NOTICA       1791 ][skype101       ][IDLE,IDLE] WAITING
roughly 10 seconds to find a running Skype client and connect to its SKYPE
API for interface_id=1
2013-08-22 05:02:15.243058 [NOTICE] mod_skypopen.c:1800
 [543dc3c|22dd4bf] [NOTICA       1800 ][skype101       ][IDLE,IDLE] Found a
running Skype client, connected to its SKYPE API for interface_id=1,
waiting 60 seconds for CURRENTUSERHANDLE==user
2013-08-22 05:03:15.283035 [ERR] mod_skypopen.c:1841
 [543dc3c|22dd4bf] [ERRORA       1841 ][skype101       ][IDLE,IDLE] The
Skype client to which we are connected FAILED to gave us
CURRENTUSERHANDLE=user, interface_id=1 FAILED to start. No Skype client
logged in as 'user' has been found. Please (re)launch a Skype client logged
in as 'user'. Skypopen exiting now

+OK Reloading XML
-ERR [module load file routine returned an error]

2013-08-22 05:03:16.283037 [CRIT] switch_loadable_module.c:1383 Error
Loading module /usr/lib/freeswitch/mod/mod_skypopen.so
**Module load routine returned an error**

# ps -aAl | grep Xvfb
0 S     0  3264     1  0  80   0 - 40832 poll_s pts/4    00:00:00 Xvfb
0 S     0  3298     1  0  80   0 - 40833 poll_s pts/4    00:00:00 Xvfb
0 S     0  3330     1  0  80   0 - 40830 poll_s pts/4    00:00:00 Xvfb

# ps -aAl | grep skype
4 S     0  3283     1  0  80   0 - 14469 poll_s ?        00:00:00 skype101
4 S     0  3317     1  0  80   0 - 14469 poll_s ?        00:00:00 skype102
4 S     0  3349     1  0  80   0 - 14451 poll_s ?        00:00:00 skype103


<snip>kill them all</snip>

root at g:~# ps -aAl | grep skype
root at g:~# ps -aAl | grep Xvfb
root at g:~# ps -aAl | grep free
root at g:~#



# ./start_skype_clients.sh
# freeswitch

freeswitch at g> log level 9

+OK
freeswitch at gnuradio08> load mod_skypopen
2013-08-22 05:12:15.803041 [INFO] mod_enum.c:876 ENUM Reloaded
2013-08-22 05:12:15.803041 [INFO] switch_time.c:1191 Timezone reloaded 530
definitions
2013-08-22 05:12:15.823035 [WARNING] mod_skypopen.c:1762
[543dc3c|22dd4bf] [WARNINGA     1762 ][skype101       ][IDLE,IDLE] STARTING
interface_id=1
2013-08-22 05:12:16.023032 [NOTICE] mod_skypopen.c:1791
 [543dc3c|22dd4bf] [NOTICA       1791 ][skype101       ][IDLE,IDLE] WAITING
roughly 10 seconds to find a running Skype client and connect to its SKYPE
API for interface_id=1
2013-08-22 05:12:16.023032 [NOTICE] mod_skypopen.c:1800
 [543dc3c|22dd4bf] [NOTICA       1800 ][skype101       ][IDLE,IDLE] Found a
running Skype client, connected to its SKYPE API for interface_id=1,
waiting 60 seconds for CURRENTUSERHANDLE==dannyiland
2013-08-22 05:13:16.063035 [ERR] mod_skypopen.c:1841
 [543dc3c|22dd4bf] [ERRORA       1841 ][skype101       ][IDLE,IDLE] The
Skype client to which we are connected FAILED to gave us
CURRENTUSERHANDLE=user, interface_id=1 FAILED to start. No Skype client
logged in as 'user' has been found. Please (re)launch a Skype client logged
in as 'user'. Skypopen exiting now

+OK Reloading XML
-ERR [module load file routine returned an error]

2013-08-22 05:13:17.063036 [CRIT] switch_loadable_module.c:1383 Error
Loading module /usr/lib/freeswitch/mod/mod_skypopen.so
**Module load routine returned an error**
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From jmesquita at freeswitch.org  Thu Aug 22 18:10:07 2013
From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Thu, 22 Aug 2013 11:10:07 -0300
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <52157852.7010604@coppice.org>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
	<CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
	<B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
	<CAHgk4BWdp9iQHsZfQmNWu0JLzidQQCtGn0kzD_AAyapKOohzQA@mail.gmail.com>
	<CAC-LwPPCLeX80DLcutwqQi4JDzJazLChivT=mPSXjTi6J19xdg@mail.gmail.com>
	<52157852.7010604@coppice.org>
Message-ID: <F3072A94-CC64-49B9-B0CE-BC10CD44B0D9@freeswitch.org>

Go Steve! Thank you for your efforts and making it OSS. 

Sent from my iPhone

On Aug 21, 2013, at 11:32 PM, Steve Underwood <steveu at coppice.org> wrote:

> On 08/22/2013 04:28 AM, Vik Killa wrote:
>> FS cannot send color faxes yet?
> Very few computer FAX platforms can send or receive colour FAXes. As of 
> this week FS can, but the support should be treated as experimental at 
> this stage.
> 
> It seems like colour FAX should be trivial, if you have error corrected 
> FAX support, but it isn't. The developers of the colour FAX standard 
> chose to use JPEG compression, but with a very unusual colour 
> representation. This means you need to develop a messy bunch of code if 
> you want the user to be able to work with conventional colour image 
> files on their computers. It is this support which we are shaking out 
> now. When its looking stable I will add the relevant information about 
> controlling colour FAXes to the wiki pages.
> 
> Regards,
> Steve
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From gmaruzz at gmail.com  Thu Aug 22 17:59:05 2013
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Thu, 22 Aug 2013 15:59:05 +0200
Subject: [Freeswitch-users] Mod_skypopen ' No Skype client logged in as
 'user' has been found'
In-Reply-To: <CAC_uT4C+45dv8iYRRgQ6uTuFZk+pUza_oqCMOjXpmcZFh3jvWg@mail.gmail.com>
References: <CAC_uT4C+45dv8iYRRgQ6uTuFZk+pUza_oqCMOjXpmcZFh3jvWg@mail.gmail.com>
Message-ID: <CALXCt0rG0mW9c7AXWG-t6_43Pwyo4pZK-CMpv_HhBmdfWCm_KA@mail.gmail.com>

use the "installer.pl" as instructed in the wiki page. It will do all
things automatically, you don't need to do anything special

-giovanni


On Thu, Aug 22, 2013 at 1:56 PM, Danny Iland <iland at cs.ucsb.edu> wrote:

> Hi,
>
> When I load mod_skypopen in freeswitch, after running
> ./start_skype_clients.sh, I get a message that 'No Skype client logged in
> as 'user' has been found'.
>
> I followed the Wiki install on Ubuntu 12 x64, then launched per the wiki's
> instructions. I have connected to the Skype client as each user via ssh -X
> to give skypopen API permissions in Skype, following
> README.skypopen_auth. I would appreciate a bit of clarification with
> regards to where mod_skypopen looks for each user's Skype config files, and
> what needs to be where in the .Skype directory. It would be particularly
> nice to see a file tree of a multi-user skypopen install's .Skype directory.
>
> I have checked all the obvious problems I could think of before coming to
> the list. Obviously I have set my own usernames and passwords in
> start_skype_clients.sh and usernames in skypopen.conf.xml
>
> Below is a log of the behavior. The second is to show that Skype is
> running, along with Xvfb.
>
> Any help is appreciated.
>
> Thanks,
> Danny Iland
> Graduate Student Researcher, University of California Santa Barbara
> iland at cs.ucsb.edu
>
>
>
> ./start_skype_clients.sh
> [snip]
> freeswitch
>
> freeswitch at g> log level 9
>
> +OK
>
> freeswitch at g> load mod_skypopen
> 2013-08-22 05:02:15.023031 [INFO] mod_enum.c:876 ENUM Reloaded
> 2013-08-22 05:02:15.023031 [INFO] switch_time.c:1191 Timezone reloaded 530
> definitions
> 2013-08-22 05:02:15.043058 [WARNING] mod_skypopen.c:1762
> [543dc3c|22dd4bf] [WARNINGA     1762 ][skype101       ][IDLE,IDLE] STARTING
> interface_id=1
> 2013-08-22 05:02:15.243058 [NOTICE] mod_skypopen.c:1791
>  [543dc3c|22dd4bf] [NOTICA       1791 ][skype101       ][IDLE,IDLE] WAITING
> roughly 10 seconds to find a running Skype client and connect to its SKYPE
> API for interface_id=1
> 2013-08-22 05:02:15.243058 [NOTICE] mod_skypopen.c:1800
>  [543dc3c|22dd4bf] [NOTICA       1800 ][skype101       ][IDLE,IDLE] Found a
> running Skype client, connected to its SKYPE API for interface_id=1,
> waiting 60 seconds for CURRENTUSERHANDLE==user
> 2013-08-22 05:03:15.283035 [ERR] mod_skypopen.c:1841
>  [543dc3c|22dd4bf] [ERRORA       1841 ][skype101       ][IDLE,IDLE] The
> Skype client to which we are connected FAILED to gave us
> CURRENTUSERHANDLE=user, interface_id=1 FAILED to start. No Skype client
> logged in as 'user' has been found. Please (re)launch a Skype client logged
> in as 'user'. Skypopen exiting now
>
> +OK Reloading XML
> -ERR [module load file routine returned an error]
>
> 2013-08-22 05:03:16.283037 [CRIT] switch_loadable_module.c:1383 Error
> Loading module /usr/lib/freeswitch/mod/mod_skypopen.so
> **Module load routine returned an error**
>
> # ps -aAl | grep Xvfb
> 0 S     0  3264     1  0  80   0 - 40832 poll_s pts/4    00:00:00 Xvfb
> 0 S     0  3298     1  0  80   0 - 40833 poll_s pts/4    00:00:00 Xvfb
> 0 S     0  3330     1  0  80   0 - 40830 poll_s pts/4    00:00:00 Xvfb
>
> # ps -aAl | grep skype
> 4 S     0  3283     1  0  80   0 - 14469 poll_s ?        00:00:00 skype101
> 4 S     0  3317     1  0  80   0 - 14469 poll_s ?        00:00:00 skype102
> 4 S     0  3349     1  0  80   0 - 14451 poll_s ?        00:00:00 skype103
>
>
> <snip>kill them all</snip>
>
> root at g:~# ps -aAl | grep skype
>  root at g:~# ps -aAl | grep Xvfb
> root at g:~# ps -aAl | grep free
> root at g:~#
>
>
>
> # ./start_skype_clients.sh
> # freeswitch
>
> freeswitch at g> log level 9
>
> +OK
> freeswitch at gnuradio08> load mod_skypopen
> 2013-08-22 05:12:15.803041 [INFO] mod_enum.c:876 ENUM Reloaded
> 2013-08-22 05:12:15.803041 [INFO] switch_time.c:1191 Timezone reloaded 530
> definitions
> 2013-08-22 05:12:15.823035 [WARNING] mod_skypopen.c:1762
> [543dc3c|22dd4bf] [WARNINGA     1762 ][skype101       ][IDLE,IDLE] STARTING
> interface_id=1
> 2013-08-22 05:12:16.023032 [NOTICE] mod_skypopen.c:1791
>  [543dc3c|22dd4bf] [NOTICA       1791 ][skype101       ][IDLE,IDLE] WAITING
> roughly 10 seconds to find a running Skype client and connect to its SKYPE
> API for interface_id=1
> 2013-08-22 05:12:16.023032 [NOTICE] mod_skypopen.c:1800
>  [543dc3c|22dd4bf] [NOTICA       1800 ][skype101       ][IDLE,IDLE] Found a
> running Skype client, connected to its SKYPE API for interface_id=1,
> waiting 60 seconds for CURRENTUSERHANDLE==dannyiland
> 2013-08-22 05:13:16.063035 [ERR] mod_skypopen.c:1841
>  [543dc3c|22dd4bf] [ERRORA       1841 ][skype101       ][IDLE,IDLE] The
> Skype client to which we are connected FAILED to gave us
> CURRENTUSERHANDLE=user, interface_id=1 FAILED to start. No Skype client
> logged in as 'user' has been found. Please (re)launch a Skype client logged
> in as 'user'. Skypopen exiting now
>
> +OK Reloading XML
> -ERR [module load file routine returned an error]
>
> 2013-08-22 05:13:17.063036 [CRIT] switch_loadable_module.c:1383 Error
> Loading module /usr/lib/freeswitch/mod/mod_skypopen.so
> **Module load routine returned an error**
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From bdfoster at davri.com  Thu Aug 22 19:58:24 2013
From: bdfoster at davri.com (Brian Foster)
Date: Thu, 22 Aug 2013 11:58:24 -0400
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <52157852.7010604@coppice.org>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
	<CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
	<B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
	<CAHgk4BWdp9iQHsZfQmNWu0JLzidQQCtGn0kzD_AAyapKOohzQA@mail.gmail.com>
	<CAC-LwPPCLeX80DLcutwqQi4JDzJazLChivT=mPSXjTi6J19xdg@mail.gmail.com>
	<52157852.7010604@coppice.org>
Message-ID: <CAHgk4BV=VoOZY3R2jR+eyRkDK=HYYeNF+EjuAsD2yjL0gK_DHQ@mail.gmail.com>

The very definition of awesome. I can't believe I'm this excited over a
(supposedly) dying technology.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 21, 2013 10:38 PM, "Steve Underwood" <steveu at coppice.org> wrote:

> On 08/22/2013 04:28 AM, Vik Killa wrote:
> > FS cannot send color faxes yet?
> >
> Very few computer FAX platforms can send or receive colour FAXes. As of
> this week FS can, but the support should be treated as experimental at
> this stage.
>
> It seems like colour FAX should be trivial, if you have error corrected
> FAX support, but it isn't. The developers of the colour FAX standard
> chose to use JPEG compression, but with a very unusual colour
> representation. This means you need to develop a messy bunch of code if
> you want the user to be able to work with conventional colour image
> files on their computers. It is this support which we are shaking out
> now. When its looking stable I will add the relevant information about
> controlling colour FAXes to the wiki pages.
>
> Regards,
> Steve
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 grcamauer at gmail.com  Thu Aug 22 20:00:50 2013
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Thu, 22 Aug 2013 13:00:50 -0300
Subject: [Freeswitch-users] Error with SQLDB on FS startup
In-Reply-To: <A70AC0C1B7E7FD478A6BF19705461AA0C8C9@IMBER.smb.curriegrad2004.ca>
References: <CAPNTnnrg6QBB5x0Zahjp=7QsLy_54_CVE_GQ6kLgbJaQOFTRDw@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C8C9@IMBER.smb.curriegrad2004.ca>
Message-ID: <-7015965643976304845@unknownmsgid>

Great, thanks.  I suspected as much, since everything seems to be working
fine.

Guillermo

Sent from my iPhone

On 21/08/2013, at 20:30, Jeff Leung <jleung at v10networks.ca> wrote:

If there?s no tables when FS starts up on a SQLite database, it?ll create
it automatically. You can safely ignore this.



*From:* freeswitch-users-bounces at lists.freeswitch.org [
mailto:freeswitch-users-bounces at lists.freeswitch.org<freeswitch-users-bounces at lists.freeswitch.org>]
*On Behalf Of *Guillermo Ruiz Camauer
*Sent:* Wednesday, August 21, 2013 11:25 AM
*To:* FreeSWITCH Users Help
*Subject:* [Freeswitch-users] Error with SQLDB on FS startup



I am getting the following errors when starting up FS:



[root at Freeswitch freeswitch]# /usr/local/freeswitch/bin/freeswitch
2013-08-21 15:13:55.201104 [INFO] switch_event.c:596 Activate Eventing
Engine.
2013-08-21 15:13:55.201308 [WARNING] switch_event.c:570 Create additional
event dispatch thread 0
2013-08-21 15:13:55.262832 [INFO] switch_nat.c:420 Scanning for NAT
2013-08-21 15:13:55.262950 [DEBUG] switch_nat.c:170 Checking for PMP 1/5
2013-08-21 15:13:55.513013 [DEBUG] switch_nat.c:170 Checking for PMP 2/5
2013-08-21 15:13:56.012990 [DEBUG] switch_nat.c:170 Checking for PMP 3/5
2013-08-21 15:13:57.013009 [DEBUG] switch_nat.c:170 Checking for PMP 4/5
2013-08-21 15:13:59.013009 [DEBUG] switch_nat.c:170 Checking for PMP 5/5
2013-08-21 15:14:03.013009 [ERR] switch_nat.c:201 Error checking for PMP
[general error]
2013-08-21 15:14:03.013039 [DEBUG] switch_nat.c:425 Checking for UPnP
2013-08-21 15:14:06.066363 [INFO] switch_nat.c:441 No PMP or UPnP NAT
devices detected!
2013-08-21 15:14:06.068084 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR [no
such table: channels]
drop table channels
2013-08-21 15:14:06.068108 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR [no
such table: calls]
drop table calls
2013-08-21 15:14:06.068125 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR [no
such table: detailed_calls]
drop view detailed_calls
2013-08-21 15:14:06.068140 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR [no
such table: basic_calls]
drop view basic_calls
2013-08-21 15:14:06.068155 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR [no
such table: interfaces]
drop table interfaces
2013-08-21 15:14:06.068169 [ERR] switch_core_sqldb.c:585 NATIVE SQL ERR [no
such table: tasks]
drop table tasks
2013-08-21 15:14:06.081283 [CONSOLE] switch_core.c:1293 Created ip list lan
default (allow)
2013-08-21 15:14:06.081302 [CONSOLE] switch_core.c:1293 Created ip list
domains default (deny)
...



I am on FreeSWITCH Version 1.2.12+git~20130821T172756Z~c73179c6ad (git
c73179c 2013-08-21 17:27:56Z). (just ran make current, but it was
happenning before also).



FS continues and loads.  What are the possible side effects of these
errors?  Is there something very wrong with my setup?




-- 
Guillermo Ruiz Camauer

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 kriskinc.com  Thu Aug 22 20:03:56 2013
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 22 Aug 2013 12:03:56 -0400
Subject: [Freeswitch-users] TLS with FreeSWITCH and Kamailio
In-Reply-To: <CAKDfjgf+gOyp_enk80AgD6ivJcu4TC+Pg0TaRweqMWekBN2ZbA@mail.gmail.com>
References: <CAKDfjge1ZwadN=a8+Kk0ZC6eEX5ykzSdGf8bmPJ1X5DHoWkm-Q@mail.gmail.com>
	<E4F0379B-E8FF-44B9-9D67-F847B06DD24A@freeswitch.org>
	<CAKDfjgf+gOyp_enk80AgD6ivJcu4TC+Pg0TaRweqMWekBN2ZbA@mail.gmail.com>
Message-ID: <CAKDfjgdktCpw9ynCxgf9rMA1ivPVb6B5eJqjmOUJQ=LZ_XXAag@mail.gmail.com>

Answering my own post:

After cranking up Sofia debugging and reading through source Sofia really
wants to see "-extensions server" on the "server" generated cert in this
case, which makes sense but it's also problematic because any SIP UA can be
a "server" or "client" with different keys (with full validation) and as
far as I can tell FreeSWITCH only allows for one key+cert per profile
(which can be a "client" or "server").

Kamailio provides for completely separate client and server TLS
configuration, including key, cert, CA, CRL, etc, etc.

Is there a mechanism to provide different client and server certs per
profile with Sofia?  It looks like that's the only way this is going to
work.

Thanks!


On Wed, Aug 21, 2013 at 5:22 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Good question!
>
> I've tried a variety of certs, going all the way back to the CA.  I
> started with your gentls_cert script and eventually moved to the
> openvpn-style "easy-rsa" package.  I will tell you that using
> identical certs with a TLS-capable pjsip pjsua client results in a
> successful TLS connection to Kamailio (using the same CA cert, client
> cert, and client key used in FreeSWITCH).  Of course I'm not changing
> the config in Kamailio either.
>
> On Wed, Aug 21, 2013 at 5:03 PM, Brian West <brian at freeswitch.org> wrote:
> > How art thou generated the certs?
> >
> > On Aug 21, 2013, at 3:38 PM, Kristian Kielhofner <kris at kriskinc.com>
> wrote:
> >
> >> Hello,
> >>
> >>  I'm trying to get TLS cert validation between FreeSWITCH (client)
> >> and Kamailio (server) up and running.  Here's my config/setup so far:
> >>
> >> FreeSWITCH 1.2.12 (client) configured with:
> >>
> >>    <!-- TLS: disabled by default, set to "true" to enable -->
> >>    <param name="tls" value="true"/>
> >>    <!-- additional bind parameters for TLS -->
> >>    <param name="tls-bind-params" value="transport=tls"/>
> >>    <!-- Port to listen on for TLS requests. (5061 will be used if
> >> unspecified) -->
> >>    <param name="tls-sip-port" value="5081"/>
> >>    <!-- Location of the agent.pem and cafile.pem ssl certificates
> >> (needed for TLS server) -->
> >>    <param name="tls-cert-dir" value="[my cert dir]"/>
> >>    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may
> >> not work with TLSv1 -->
> >>    <param name="tls-version" value="tlsv1"/>
> >>    <param name="tls-verify-policy" value="out"/>
> >>
> >> I have a gateway configured with ;transport=tls
> >>
> >> Kamailio 4.0 (also tried 4.1, etc) configured with (tls.cfg):
> >>
> >> [server:default]
> >> method = TLSv1
> >> verify_certificate = no
> >> require_certificate = yes
> >> private_key = /etc/kamailio/generic-sip.key
> >> certificate = /etc/kamailio/generic-sip.pem
> >> ca_list = /etc/kamailio/generic-cacert.pem
> >> cipher_list = AES
> >>
> >>  I'm using my own CA with self-signed certs.  I've verified that they
> >> check out by comparing the modulus on the cert and key pairs and
> >> verifying the CA chain with 'openssl verify ...'.
> >>
> >>  When I run without tls-verify-policy=none and require_certificate=no
> >> everything is golden and TLS works all day long.  However, this is
> >> less than ideal and I'd like to at least make sure that my TLS clients
> >> are presenting a valid cert.  Unfortunately when FS tries to connect
> >> to Kamailio it reports the following errors:
> >>
> >> ERROR: tls [tls_server.c:1190]: TLS accept:error:140890B2:SSL
> >> routines:SSL3_GET_CLIENT_CERTIFICATE:no certificate returned
> >> ERROR: <core> [tcp_read.c:1275]: ERROR: tcp_read_req: error reading
> >>
> >>  What's interesting is that FreeSWITCH reports a successful
> >> registration and seems to exchange OPTIONS pings (over UDP!) with the
> >> remote Kamailio instance.  However, Kamailio does not show the
> >> endpoint as registered (verified with 'kamctl ul show').  That seems
> >> like a bug and worthy of a JIRA but my main concern at this point is
> >> getting TLS with certificate validation up and running.
> >>
> >>  Any ideas?  Thanks!
> >>
> >> --
> >> Kristian Kielhofner
> >>
> >>
> _________________________________________________________________________
> >> Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.com
> >
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
>
> --
> Kristian Kielhofner
>



-- 
Kristian Kielhofner
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From vipkilla at gmail.com  Thu Aug 22 21:47:12 2013
From: vipkilla at gmail.com (Vik Killa)
Date: Thu, 22 Aug 2013 13:47:12 -0400
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <CAHgk4BV=VoOZY3R2jR+eyRkDK=HYYeNF+EjuAsD2yjL0gK_DHQ@mail.gmail.com>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
	<CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
	<B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
	<CAHgk4BWdp9iQHsZfQmNWu0JLzidQQCtGn0kzD_AAyapKOohzQA@mail.gmail.com>
	<CAC-LwPPCLeX80DLcutwqQi4JDzJazLChivT=mPSXjTi6J19xdg@mail.gmail.com>
	<52157852.7010604@coppice.org>
	<CAHgk4BV=VoOZY3R2jR+eyRkDK=HYYeNF+EjuAsD2yjL0gK_DHQ@mail.gmail.com>
Message-ID: <CAC-LwPM1StLEQW9ZvjY1T5nEk32SMbBqLLTNNK4DKe8YaRcf9Q@mail.gmail.com>

What about color faxing from FS to FS?
<snip>
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From kris at kriskinc.com  Thu Aug 22 22:13:09 2013
From: kris at kriskinc.com (Kristian Kielhofner)
Date: Thu, 22 Aug 2013 14:13:09 -0400
Subject: [Freeswitch-users] TLS with FreeSWITCH and Kamailio
In-Reply-To: <CAKDfjgdktCpw9ynCxgf9rMA1ivPVb6B5eJqjmOUJQ=LZ_XXAag@mail.gmail.com>
References: <CAKDfjge1ZwadN=a8+Kk0ZC6eEX5ykzSdGf8bmPJ1X5DHoWkm-Q@mail.gmail.com>
	<E4F0379B-E8FF-44B9-9D67-F847B06DD24A@freeswitch.org>
	<CAKDfjgf+gOyp_enk80AgD6ivJcu4TC+Pg0TaRweqMWekBN2ZbA@mail.gmail.com>
	<CAKDfjgdktCpw9ynCxgf9rMA1ivPVb6B5eJqjmOUJQ=LZ_XXAag@mail.gmail.com>
Message-ID: <CAKDfjgeysSqRdqWAW=s0sh9sHGxHoxWbZAy6LKjj0RuqsR5K5A@mail.gmail.com>

http://jira.freeswitch.org/browse/FS-5725


On Thu, Aug 22, 2013 at 12:03 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:

> Answering my own post:
>
> After cranking up Sofia debugging and reading through source Sofia really
> wants to see "-extensions server" on the "server" generated cert in this
> case, which makes sense but it's also problematic because any SIP UA can be
> a "server" or "client" with different keys (with full validation) and as
> far as I can tell FreeSWITCH only allows for one key+cert per profile
> (which can be a "client" or "server").
>
> Kamailio provides for completely separate client and server TLS
> configuration, including key, cert, CA, CRL, etc, etc.
>
> Is there a mechanism to provide different client and server certs per
> profile with Sofia?  It looks like that's the only way this is going to
> work.
>
> Thanks!
>
>
> On Wed, Aug 21, 2013 at 5:22 PM, Kristian Kielhofner <kris at kriskinc.com>wrote:
>
>> Good question!
>>
>> I've tried a variety of certs, going all the way back to the CA.  I
>> started with your gentls_cert script and eventually moved to the
>> openvpn-style "easy-rsa" package.  I will tell you that using
>> identical certs with a TLS-capable pjsip pjsua client results in a
>> successful TLS connection to Kamailio (using the same CA cert, client
>> cert, and client key used in FreeSWITCH).  Of course I'm not changing
>> the config in Kamailio either.
>>
>> On Wed, Aug 21, 2013 at 5:03 PM, Brian West <brian at freeswitch.org> wrote:
>> > How art thou generated the certs?
>> >
>> > On Aug 21, 2013, at 3:38 PM, Kristian Kielhofner <kris at kriskinc.com>
>> wrote:
>> >
>> >> Hello,
>> >>
>> >>  I'm trying to get TLS cert validation between FreeSWITCH (client)
>> >> and Kamailio (server) up and running.  Here's my config/setup so far:
>> >>
>> >> FreeSWITCH 1.2.12 (client) configured with:
>> >>
>> >>    <!-- TLS: disabled by default, set to "true" to enable -->
>> >>    <param name="tls" value="true"/>
>> >>    <!-- additional bind parameters for TLS -->
>> >>    <param name="tls-bind-params" value="transport=tls"/>
>> >>    <!-- Port to listen on for TLS requests. (5061 will be used if
>> >> unspecified) -->
>> >>    <param name="tls-sip-port" value="5081"/>
>> >>    <!-- Location of the agent.pem and cafile.pem ssl certificates
>> >> (needed for TLS server) -->
>> >>    <param name="tls-cert-dir" value="[my cert dir]"/>
>> >>    <!-- TLS version ("sslv23" (default), "tlsv1"). NOTE: Phones may
>> >> not work with TLSv1 -->
>> >>    <param name="tls-version" value="tlsv1"/>
>> >>    <param name="tls-verify-policy" value="out"/>
>> >>
>> >> I have a gateway configured with ;transport=tls
>> >>
>> >> Kamailio 4.0 (also tried 4.1, etc) configured with (tls.cfg):
>> >>
>> >> [server:default]
>> >> method = TLSv1
>> >> verify_certificate = no
>> >> require_certificate = yes
>> >> private_key = /etc/kamailio/generic-sip.key
>> >> certificate = /etc/kamailio/generic-sip.pem
>> >> ca_list = /etc/kamailio/generic-cacert.pem
>> >> cipher_list = AES
>> >>
>> >>  I'm using my own CA with self-signed certs.  I've verified that they
>> >> check out by comparing the modulus on the cert and key pairs and
>> >> verifying the CA chain with 'openssl verify ...'.
>> >>
>> >>  When I run without tls-verify-policy=none and require_certificate=no
>> >> everything is golden and TLS works all day long.  However, this is
>> >> less than ideal and I'd like to at least make sure that my TLS clients
>> >> are presenting a valid cert.  Unfortunately when FS tries to connect
>> >> to Kamailio it reports the following errors:
>> >>
>> >> ERROR: tls [tls_server.c:1190]: TLS accept:error:140890B2:SSL
>> >> routines:SSL3_GET_CLIENT_CERTIFICATE:no certificate returned
>> >> ERROR: <core> [tcp_read.c:1275]: ERROR: tcp_read_req: error reading
>> >>
>> >>  What's interesting is that FreeSWITCH reports a successful
>> >> registration and seems to exchange OPTIONS pings (over UDP!) with the
>> >> remote Kamailio instance.  However, Kamailio does not show the
>> >> endpoint as registered (verified with 'kamctl ul show').  That seems
>> >> like a bug and worthy of a JIRA but my main concern at this point is
>> >> getting TLS with certificate validation up and running.
>> >>
>> >>  Any ideas?  Thanks!
>> >>
>> >> --
>> >> Kristian Kielhofner
>> >>
>> >>
>> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
>> >> Official FreeSWITCH Sites
>> >> http://www.freeswitch.org
>> >> http://wiki.freeswitch.org
>> >> http://www.cluecon.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
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.com
>> >
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>>
>>
>>
>> --
>> Kristian Kielhofner
>>
>
>
>
> --
> Kristian Kielhofner
>



-- 
Kristian Kielhofner
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From k4kaleem at gmail.com  Thu Aug 22 22:41:04 2013
From: k4kaleem at gmail.com (Kaleem)
Date: Thu, 22 Aug 2013 19:41:04 +0100
Subject: [Freeswitch-users] Conferencing solution - need advice
In-Reply-To: <CAPhuNs_KvmL_WH6_rFXonV-_PT0OpBOFBW6km5MB-aNYf8REpQ@mail.gmail.com>
References: <CAFWsKqz1t+V1dfX-HzT_adAcR8dC_T5f=EZe7zDnza2SvRYHDw@mail.gmail.com>
	<CAKzWOxVM0G=XrTfX_ZKi2HGGbWWqDxQz3wkyfTd5yvnpGGOYAw@mail.gmail.com>
	<A3579698-5452-48CD-B400-29408B5AAD9F@gmail.com>
	<1377163429.38648.YahooMailNeo@web126205.mail.ne1.yahoo.com>
	<CAPhuNs_KvmL_WH6_rFXonV-_PT0OpBOFBW6km5MB-aNYf8REpQ@mail.gmail.com>
Message-ID: <6A85391E-763C-42DA-8547-24CA76326F88@gmail.com>

Hi, in that case, how about a star network of freeswitch?

One FS is central and other FS servers around transfer calls to it, in a way that they setup one stream only to main and merge all connected callers to it.

Sent from my iPhone

On 22 Aug 2013, at 14:05, "Veniamin (Benjamin) Beskrovny" <beskrovny at gmail.com> wrote:

> Indeed, Stan is right. I would rather concentrate on joining one
> (maybe dedicated instance of FS somewhere in the middle) instead of
> keeping full mesh of talkers, thus eating bandwidth and resources...
> 
> On 8/22/13, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>> this would require active SIP channels between the conference instances, but
>> not counting as participants.
>> 
>> Also some development work needs to be done to distribute the conference
>> control between the servers:
>> -- moderator joined, everyone can talk
>> -- moderator control, like kicking off or muting participants
>> -- moderator ending and shutting down the conference session
>> -- total count and listing of all participants
>> 
>> This sounds like ~50 to 100 hours of developer's work if you need a full
>> conferencing functionality across multiple servers.
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>> 
>>> ________________________________
>>> From: Kaleem <k4kaleem at gmail.com>
>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>> Sent: Thursday, August 22, 2013 9:45 AM
>>> Subject: Re: [Freeswitch-users] Conferencing solution - need advice
>>> 
>>> 
>>> 
>>> Hi Mike, thanks for looking into this, yes all servers will be active all
>>> time running on high end virtual environment
>>> Thanks
>>> 
>>> 
>>> Sent from my iPhone
>>> 
>>> On 22 Aug 2013, at 00:09, Michael Collins <msc at freeswitch.org> wrote:
>>> 
>>> 
>>> Will the four individual conferences always be connected to each other?
>>>> -MC
>>>> 
>>>> 
>>>> 
>>>> 
>>>> On Wed, Aug 21, 2013 at 1:55 PM, kaleem rehman <k4kaleem at gmail.com> wrote:
>>>> 
>>>> Hi All,
>>>>> 
>>>>> I want to implement a conferencing solution using freeswitch with Avaya.
>>>>> 
>>>>> We currently have a multi-country hosted telephony platform, I want to
>>>>> introduce freeswitch to it for internal use to see how it goes.
>>>>> 
>>>>> I want a freeswitch server in UK, one in Europe, one in Americas and one
>>>>> in South Africa.
>>>>> 
>>>>> i would like all freeswitch server to talk to each other over SIP over
>>>>> internal Cloud then i want UK conference users to join to UK freeswitch
>>>>> and for same conference other countries users to join their local
>>>>> freeswitch server and talk together.
>>>>> 
>>>>> this way i want to achieve less cloud load and in reality one call
>>>>> between all servers where users all users from their country will
>>>>> connect to local freeswitch and that freeswitch links then to a
>>>>> conference with other country.
>>>>> 
>>>>> please advise on how i can achieve this.
>>>>> 
>>>>> i am open to suggestions and ideas.
>>>>> 
>>>>> regards,
>>>>> Kaleem
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>> 
>>>>> 
>>>>> 
>>>>> 
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>> 
>>>>> 
>>>> 
>>>> 
>>>> --
>>>> Michael S Collins
>>>> Twitter: @mercutioviz
>>>> http://www.FreeSWITCH.org
>>>> http://www.ClueCon.com
>>>> http://www.OSTAG.org
>>>> 
>>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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 guga.salazar.loor at gmail.com  Thu Aug 22 23:17:07 2013
From: guga.salazar.loor at gmail.com (Gustavo Salazar)
Date: Thu, 22 Aug 2013 14:17:07 -0500
Subject: [Freeswitch-users] Can't call from Firefox 22 to Freeswitch
	using sipml5
In-Reply-To: <CAKbxfG_GUuHwT_V_H0HYRPA=sQNp9VGu9txtWXYZ=A0OspR65g@mail.gmail.com>
References: <CAMKwYLASs++ZxCNiAQ1yJMB+oROwwPAr8MuZ65g4dyQ-6nY_fg@mail.gmail.com>
	<CAFArqsbMZXD4pK2W65sNOQpgn7TdCR9wec8YFwM6NLYFyftdQw@mail.gmail.com>
	<CAMKwYLAdOvXUz0XLcJX653MwDZbYaoGO=q7kH8QYm7xx0urGBA@mail.gmail.com>
	<51FFC89A.4020807@orange-vallee.net>
	<CAFArqsYXECis-KFXstadKGSNjHhs-oePr-UbSYy67bqKndDuHg@mail.gmail.com>
	<CAKbxfG8JiU20-Atc58Wyt3pHSoUptsecMgoQq47iyu-DV2aHgA@mail.gmail.com>
	<CAFArqsYho0u56kmenj9PHPSjAMykpVSbh_1fC6KGd1tf5u0iKg@mail.gmail.com>
	<CAKbxfG-bSG7sgZN039WaZ5GhpcR9sP19hy+-D2FkweMzEURnhw@mail.gmail.com>
	<CAFArqsZA1DtMoK_OuEDEjcOVJ+mEeF_g+a9-K5Jf2sWf7aDFFA@mail.gmail.com>
	<CAKbxfG_GUuHwT_V_H0HYRPA=sQNp9VGu9txtWXYZ=A0OspR65g@mail.gmail.com>
Message-ID: <CAMKwYLD-p5_1MFAsAHmg1FV-=tmseW6PT8vcP6X5Ntk2bYBa7A@mail.gmail.com>

Karsten, I executed the openssl commands   against my local server and
webrtc.freeswitch.org,  both show the same output that you described
against a broken host.

Using Firefox I still have the same results:

Firefox -> 5000 ->  webrtc.freeswitch.org ....... GOOD
Firefox -> 5000 -> my local vm .......................NO AUDIO

And now using Chrome:

Chrome  -> 5000 -> webrtc.freeswitch.org ....... NO AUDIO This worked before
Chrome ->  -> 5000 -> my local vm ......................GOOD
I am using Chrome 29.0.1547.57 m

Karsten, did the suggestions from Anthony work for you?

Anthony, In the freeswitch cli I see this message many times
2013-08-22 11:42:55.504980 [ALERT] switch_rtp.c:5662 Skip sending audio
packet 98 bytes (dtls not ready!)
Do you have something related to dtls configured in webrtc.freeswitch.org?






2013/8/5 Anthony Minessale <anthony.minessale at gmail.com>

> Your example might have different network conditions than the demo.
> You can check by enabling sip trace on FS and bring up the call and look
> at the ports that are advertised out.
> You can try turning off selinux or iptables as a test.
>
>
>
>
> On Mon, Aug 5, 2013 at 3:41 PM, Karsten Horsmann <khorsmann at gmail.com>wrote:
>
>> Hi Anthony,
>>
>> how is the best way to check that? With wireshark most stuff is ssl
>> unlike normal UDP/SIP stuff.
>>
>> Iam still confused, that it works with webrtc.freeswitch.org at work and
>> at home, but not with my example setup.
>>
>> I checked for iptables (everything fine), and its no cloud virtual foo
>> where the server is running.
>>
>> Thanks in advance,
>> Regards
>>
>>
>> 2013/8/5 Anthony Minessale <anthony.minessale at gmail.com>
>>
>>> Make sure its going to the ports that are advertised in the sdp and that
>>> there are no nat or firewall issues in between.
>>>
>>>
>>
>>
>> --
>> Mit freundlichen Gr??en
>> *Karsten Horsmann*
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Gustavo Salazar
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From ga at steadfasttelecom.com  Thu Aug 22 23:21:33 2013
From: ga at steadfasttelecom.com (Gilad Abada)
Date: Thu, 22 Aug 2013 15:21:33 -0400
Subject: [Freeswitch-users] Snom and valet_parking
In-Reply-To: <5213EBC2.4000704@stubbornroses.com>
References: <52128129.4050004@stubbornroses.com>
	<5213EBC2.4000704@stubbornroses.com>
Message-ID: <A48EFC2B9269423EBEBF37A91FAB6231@steadfasttelecom.com>

Hey Guys, 

I can confirm James's issue on a Snom 300 firmware version 8.7.3.19.

You should open a JIRA James.

Thanks again for everyones help/support

Gill 
> 
> -------- Original Message -------- 
> Subject: 
> [Freeswitch-users] Snom and valet_parking
> 
> Date: 
> Mon, 19 Aug 2013 15:33:45 -0500
> 
> From: 
> soapee01.fs at stubbornroses.com (mailto:soapee01.fs at stubbornroses.com)
> 
> Reply-To: 
> FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> (mailto:freeswitch-users at lists.freeswitch.org)
> 
> To: 
> freeswitch-users at lists.freeswitch.org (mailto:freeswitch-users at lists.freeswitch.org)
> 
> 
> 
> 
> 
> Hi list: I have an issue with a new Snom 821. I've configured valet_parking for auto park mode (eg: my_lot auto in). After the attended transfer, I always receive a Temporarily Unavailable response on the Snom phone from FS. This does not hold true on my yealink/cisco/linksys/etc phones. I can do proper attended transfers from extension to extension on the Snom. I've tried this on a separate machine with latest stable, and latest git head. Dialplan: <extension name="valet_auto_park_in" > <condition field="destination_number" expression="^(7900)$" > <action application="valet_park" data="my_lot@${domain_name} auto in 7901 7999" /> </condition> </extension> <extension name="valet_auto_park_out" > <condition field="destination_number" expression="^(79\d\d)$" > <action application="answer" /> <action application="valet_park" data="my_lot@${domain_name} $1" /> </condition> </extension> screenshot from wireshark: http://tinypic.com/r/jtp5hc/5 Relevant Log: http://pastebin.freeswitch.org/
21322 Snom Firmware Version: snom821-SIP 8.7.3.19 The call flow analysis on wireshark doesn't include any of the sip messages that follow the 480, so it doesn't really provide much there. Any pointers would be greatly appreciated. Duplication (for my own personal sanity) would also be appreciated. Could be I just got a bad phone... Regards, James PS: I know the timestamps on the wireshark don't match the log. They are from different tests, but the results are always the same. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org (mailto:consulting at freeswitch.org) http://www.freeswitchsolutions.com   Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) ht
tp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org 

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From bdfoster at davri.com  Thu Aug 22 23:39:44 2013
From: bdfoster at davri.com (Brian Foster)
Date: Thu, 22 Aug 2013 15:39:44 -0400
Subject: [Freeswitch-users] Conferencing solution - need advice
In-Reply-To: <6A85391E-763C-42DA-8547-24CA76326F88@gmail.com>
References: <CAFWsKqz1t+V1dfX-HzT_adAcR8dC_T5f=EZe7zDnza2SvRYHDw@mail.gmail.com>
	<CAKzWOxVM0G=XrTfX_ZKi2HGGbWWqDxQz3wkyfTd5yvnpGGOYAw@mail.gmail.com>
	<A3579698-5452-48CD-B400-29408B5AAD9F@gmail.com>
	<1377163429.38648.YahooMailNeo@web126205.mail.ne1.yahoo.com>
	<CAPhuNs_KvmL_WH6_rFXonV-_PT0OpBOFBW6km5MB-aNYf8REpQ@mail.gmail.com>
	<6A85391E-763C-42DA-8547-24CA76326F88@gmail.com>
Message-ID: <CAHgk4BX253=W5sxFNAxhs2gUa+-j_xPJ2WjZYf7AzDfMH5Hgqg@mail.gmail.com>

It doesn't quite work that way. Every call to main conference server will
have it's own audio. Defeats the purpose of having a distributed setup.

But I don't think the control issue is a big deal. You could always
implement a web gui that kept track of participants and use that to control
the conference on the edge servers. A little ESL, some PHP/AJAX and you
should be fine if yiu're comfortable with those technologies.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 22, 2013 2:45 PM, "Kaleem" <k4kaleem at gmail.com> wrote:

> Hi, in that case, how about a star network of freeswitch?
>
> One FS is central and other FS servers around transfer calls to it, in a
> way that they setup one stream only to main and merge all connected callers
> to it.
>
> Sent from my iPhone
>
> On 22 Aug 2013, at 14:05, "Veniamin (Benjamin) Beskrovny" <
> beskrovny at gmail.com> wrote:
>
> > Indeed, Stan is right. I would rather concentrate on joining one
> > (maybe dedicated instance of FS somewhere in the middle) instead of
> > keeping full mesh of talkers, thus eating bandwidth and resources...
> >
> > On 8/22/13, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
> >> this would require active SIP channels between the conference
> instances, but
> >> not counting as participants.
> >>
> >> Also some development work needs to be done to distribute the conference
> >> control between the servers:
> >> -- moderator joined, everyone can talk
> >> -- moderator control, like kicking off or muting participants
> >> -- moderator ending and shutting down the conference session
> >> -- total count and listing of all participants
> >>
> >> This sounds like ~50 to 100 hours of developer's work if you need a full
> >> conferencing functionality across multiple servers.
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>
> >>> ________________________________
> >>> From: Kaleem <k4kaleem at gmail.com>
> >>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> >>> Sent: Thursday, August 22, 2013 9:45 AM
> >>> Subject: Re: [Freeswitch-users] Conferencing solution - need advice
> >>>
> >>>
> >>>
> >>> Hi Mike, thanks for looking into this, yes all servers will be active
> all
> >>> time running on high end virtual environment
> >>> Thanks
> >>>
> >>>
> >>> Sent from my iPhone
> >>>
> >>> On 22 Aug 2013, at 00:09, Michael Collins <msc at freeswitch.org> wrote:
> >>>
> >>>
> >>> Will the four individual conferences always be connected to each other?
> >>>> -MC
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> On Wed, Aug 21, 2013 at 1:55 PM, kaleem rehman <k4kaleem at gmail.com>
> wrote:
> >>>>
> >>>> Hi All,
> >>>>>
> >>>>> I want to implement a conferencing solution using freeswitch with
> Avaya.
> >>>>>
> >>>>> We currently have a multi-country hosted telephony platform, I want
> to
> >>>>> introduce freeswitch to it for internal use to see how it goes.
> >>>>>
> >>>>> I want a freeswitch server in UK, one in Europe, one in Americas and
> one
> >>>>> in South Africa.
> >>>>>
> >>>>> i would like all freeswitch server to talk to each other over SIP
> over
> >>>>> internal Cloud then i want UK conference users to join to UK
> freeswitch
> >>>>> and for same conference other countries users to join their local
> >>>>> freeswitch server and talk together.
> >>>>>
> >>>>> this way i want to achieve less cloud load and in reality one call
> >>>>> between all servers where users all users from their country will
> >>>>> connect to local freeswitch and that freeswitch links then to a
> >>>>> conference with other country.
> >>>>>
> >>>>> please advise on how i can achieve this.
> >>>>>
> >>>>> i am open to suggestions and ideas.
> >>>>>
> >>>>> regards,
> >>>>> Kaleem
> >>>>>
> _________________________________________________________________________
> >>>>> Professional FreeSWITCH Consulting Services:
> >>>>> consulting at freeswitch.org
> >>>>> http://www.freeswitchsolutions.com
> >>>>>
> >>>>> 
> >>>>> 
> >>>>>
> >>>>> Official FreeSWITCH Sites
> >>>>> http://www.freeswitch.org
> >>>>> http://wiki.freeswitch.org
> >>>>> http://www.cluecon.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
> >>>>>
> >>>>>
> >>>>
> >>>>
> >>>> --
> >>>> Michael S Collins
> >>>> Twitter: @mercutioviz
> >>>> http://www.FreeSWITCH.org
> >>>> http://www.ClueCon.com
> >>>> http://www.OSTAG.org
> >>>>
> >>>>
> >>>
> _________________________________________________________________________
> >>> Professional FreeSWITCH Consulting Services:
> >>> consulting at freeswitch.org
> >>> http://www.freeswitchsolutions.com
> >>>
> >>> 
> >>> 
> >>>
> >>> Official FreeSWITCH Sites
> >>> http://www.freeswitch.org
> >>> http://wiki.freeswitch.org
> >>> http://www.cluecon.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
> >>>
> >>>
> >>>
> >
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ssinyagin at yahoo.com  Fri Aug 23 00:14:51 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Thu, 22 Aug 2013 13:14:51 -0700 (PDT)
Subject: [Freeswitch-users] Conferencing solution - need advice
In-Reply-To: <CAHgk4BX253=W5sxFNAxhs2gUa+-j_xPJ2WjZYf7AzDfMH5Hgqg@mail.gmail.com>
References: <CAFWsKqz1t+V1dfX-HzT_adAcR8dC_T5f=EZe7zDnza2SvRYHDw@mail.gmail.com>	<CAKzWOxVM0G=XrTfX_ZKi2HGGbWWqDxQz3wkyfTd5yvnpGGOYAw@mail.gmail.com>	<A3579698-5452-48CD-B400-29408B5AAD9F@gmail.com>	<1377163429.38648.YahooMailNeo@web126205.mail.ne1.yahoo.com>	<CAPhuNs_KvmL_WH6_rFXonV-_PT0OpBOFBW6km5MB-aNYf8REpQ@mail.gmail.com>	<6A85391E-763C-42DA-8547-24CA76326F88@gmail.com>
	<CAHgk4BX253=W5sxFNAxhs2gUa+-j_xPJ2WjZYf7AzDfMH5Hgqg@mail.gmail.com>
Message-ID: <1377202491.60638.YahooMailNeo@web126202.mail.ne1.yahoo.com>

it's possible to run local conferences on the hub servers, and let them automatically dial-out toward the central server.

But you don't have the moderator controls. The typical scenario is that the conference only starts when the moderator logs in. In this hub and spoke design, you need some external program which watches the ESL sockets and performs the control actions.

If bandwidth is so much of concern, you can use Speex codec, as it gives a pretty good voice quality and moderate bandwidth usage. The CPU usage will grow, of course. Then the hub boxes would compress the voice and send each call to the central conference bridge. In this case, you would keep the full moderator control, and nothing needs to be developed on top of today's FreeSWITCH.





?





>________________________________
> From: Brian Foster <bdfoster at davri.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Thursday, August 22, 2013 9:39 PM
>Subject: Re: [Freeswitch-users] Conferencing solution - need advice
> 
>
>
>It doesn't quite work that way. Every call to main conference server will have it's own audio. Defeats the purpose of having a distributed setup.
>But I don't think the control issue is a big deal. You could always implement a web gui that kept track of participants and use that to control the conference on the edge servers. A little ESL, some PHP/AJAX and you should be fine if yiu're comfortable with those technologies.
>Thank you,
>Brian Foster
>Project Manager/Owner's Rep.
>Davri Investments, Inc.
>O: 317-787-2686 x2102
>M: 317-600-9753
>E: bdfoster at davri.com
>Indianapolis, Indiana
>Sent from a mobile device.
>On Aug 22, 2013 2:45 PM, "Kaleem" <k4kaleem at gmail.com> wrote:
>
>Hi, in that case, how about a star network of freeswitch?
>>
>>One FS is central and other FS servers around transfer calls to it, in a way that they setup one stream only to main and merge all connected callers to it.
>>
>>Sent from my iPhone
>>
>>On 22 Aug 2013, at 14:05, "Veniamin (Benjamin) Beskrovny" <beskrovny at gmail.com> wrote:
>>
>>> Indeed, Stan is right. I would rather concentrate on joining one
>>> (maybe dedicated instance of FS somewhere in the middle) instead of
>>> keeping full mesh of talkers, thus eating bandwidth and resources...
>>>
>>> On 8/22/13, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>>>> this would require active SIP channels between the conference instances, but
>>>> not counting as participants.
>>>>
>>>> Also some development work needs to be done to distribute the conference
>>>> control between the servers:
>>>> -- moderator joined, everyone can talk
>>>> -- moderator control, like kicking off or muting participants
>>>> -- moderator ending and shutting down the conference session
>>>> -- total count and listing of all participants
>>>>
>>>> This sounds like ~50 to 100 hours of developer's work if you need a full
>>>> conferencing functionality across multiple servers.
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>
>>>>> ________________________________
>>>>> From: Kaleem <k4kaleem at gmail.com>
>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>> Sent: Thursday, August 22, 2013 9:45 AM
>>>>> Subject: Re: [Freeswitch-users] Conferencing solution - need advice
>>>>>
>>>>>
>>>>>
>>>>> Hi Mike, thanks for looking into this, yes all servers will be active all
>>>>> time running on high end virtual environment
>>>>> Thanks
>>>>>
>>>>>
>>>>> Sent from my iPhone
>>>>>
>>>>> On 22 Aug 2013, at 00:09, Michael Collins <msc at freeswitch.org> wrote:
>>>>>
>>>>>
>>>>> Will the four individual conferences always be connected to each other?
>>>>>> -MC
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Wed, Aug 21, 2013 at 1:55 PM, kaleem rehman <k4kaleem at gmail.com> wrote:
>>>>>>
>>>>>> Hi All,
>>>>>>>
>>>>>>> I want to implement a conferencing solution using freeswitch with Avaya.
>>>>>>>
>>>>>>> We currently have a multi-country hosted telephony platform, I want to
>>>>>>> introduce freeswitch to it for internal use to see how it goes.
>>>>>>>
>>>>>>> I want a freeswitch server in UK, one in Europe, one in Americas and one
>>>>>>> in South Africa.
>>>>>>>
>>>>>>> i would like all freeswitch server to talk to each other over SIP over
>>>>>>> internal Cloud then i want UK conference users to join to UK freeswitch
>>>>>>> and for same conference other countries users to join their local
>>>>>>> freeswitch server and talk together.
>>>>>>>
>>>>>>> this way i want to achieve less cloud load and in reality one call
>>>>>>> between all servers where users all users from their country will
>>>>>>> connect to local freeswitch and that freeswitch links then to a
>>>>>>> conference with other country.
>>>>>>>
>>>>>>> please advise on how i can achieve this.
>>>>>>>
>>>>>>> i am open to suggestions and ideas.
>>>>>>>
>>>>>>> regards,
>>>>>>> Kaleem
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Michael S Collins
>>>>>> Twitter: @mercutioviz
>>>>>> http://www.FreeSWITCH.org
>>>>>> http://www.ClueCon.com
>>>>>> http://www.OSTAG.org
>>>>>>
>>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>>
>>>
>>>
>>
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
>>consulting at freeswitch.org
>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
>>http://www.freeswitch.org
>>http://wiki.freeswitch.org
>>http://www.cluecon.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
>>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 ssinyagin at yahoo.com  Fri Aug 23 00:22:09 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Thu, 22 Aug 2013 13:22:09 -0700 (PDT)
Subject: [Freeswitch-users] Conferencing solution - need advice
In-Reply-To: <1377202491.60638.YahooMailNeo@web126202.mail.ne1.yahoo.com>
References: <CAFWsKqz1t+V1dfX-HzT_adAcR8dC_T5f=EZe7zDnza2SvRYHDw@mail.gmail.com>	<CAKzWOxVM0G=XrTfX_ZKi2HGGbWWqDxQz3wkyfTd5yvnpGGOYAw@mail.gmail.com>	<A3579698-5452-48CD-B400-29408B5AAD9F@gmail.com>	<1377163429.38648.YahooMailNeo@web126205.mail.ne1.yahoo.com>	<CAPhuNs_KvmL_WH6_rFXonV-_PT0OpBOFBW6km5MB-aNYf8REpQ@mail.gmail.com>	<6A85391E-763C-42DA-8547-24CA76326F88@gmail.com>	<CAHgk4BX253=W5sxFNAxhs2gUa+-j_xPJ2WjZYf7AzDfMH5Hgqg@mail.gmail.com>
	<1377202491.60638.YahooMailNeo@web126202.mail.ne1.yahoo.com>
Message-ID: <1377202929.88047.YahooMailNeo@web126205.mail.ne1.yahoo.com>




>
>it's possible to run local conferences on the hub servers, and let them automatically dial-out toward the central server.
>
>I meant, conferences on the spoke servers, and dial-out to the hub server.




But you don't have the moderator controls. The typical scenario is that the conference only starts when the moderator logs in. In this hub and spoke design, you need some external program which watches the ESL sockets and performs the control actions.
>
>If bandwidth is so much of concern, you can use Speex codec, as it gives a pretty good voice quality and moderate bandwidth usage. The CPU usage will grow, of course. Then the hub boxes would compress the voice and send each call to the central conference bridge. In this case, you would keep the full moderator control, and nothing needs to be developed on top of today's
 FreeSWITCH.
>
>
>
>
>
>?
>
>
>
>
>
>
>>________________________________
>> From: Brian Foster <bdfoster at davri.com>
>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>>Sent: Thursday, August 22, 2013 9:39 PM
>>Subject: Re: [Freeswitch-users] Conferencing solution - need advice
>> 
>>
>>
>>It doesn't quite work that way. Every call to main conference server will have it's own audio. Defeats the purpose of having a distributed setup.
>>But I don't think the control issue is a big deal. You could always implement a web gui that kept track of participants and use that to control the conference on the edge servers. A little ESL, some PHP/AJAX and you should be fine if yiu're comfortable with those technologies.
>>Thank you,
>>Brian Foster
>>Project Manager/Owner's Rep.
>>Davri Investments, Inc.
>>O: 317-787-2686 x2102
>>M: 317-600-9753
>>E: bdfoster at davri.com
>>Indianapolis, Indiana
>>Sent from a mobile device.
>>On Aug 22, 2013 2:45 PM, "Kaleem" <k4kaleem at gmail.com> wrote:
>>
>>Hi, in that case, how about a star network of freeswitch?
>>>
>>>One FS is central and other FS servers around transfer calls to it, in a way that they setup one stream only to main and merge all connected callers to it.
>>>
>>>Sent from my iPhone
>>>
>>>On 22 Aug 2013, at 14:05, "Veniamin (Benjamin) Beskrovny" <beskrovny at gmail.com> wrote:
>>>
>>>> Indeed, Stan is right. I would rather concentrate on joining one
>>>> (maybe dedicated instance of FS somewhere in the middle) instead of
>>>> keeping full mesh of talkers, thus eating bandwidth and resources...
>>>>
>>>> On 8/22/13, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>>>>> this would require active SIP channels between the conference instances, but
>>>>> not counting as participants.
>>>>>
>>>>> Also some development work needs to be done to distribute the conference
>>>>> control between the servers:
>>>>> -- moderator joined, everyone can talk
>>>>> -- moderator control, like kicking off or muting participants
>>>>> -- moderator ending and shutting down the conference session
>>>>> -- total count and listing of all participants
>>>>>
>>>>> This sounds like ~50 to 100 hours of developer's work if you need a full
>>>>> conferencing functionality across multiple servers.
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>> ________________________________
>>>>>> From: Kaleem <k4kaleem at gmail.com>
>>>>>> To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>>>>>> Sent: Thursday, August 22, 2013 9:45 AM
>>>>>> Subject: Re: [Freeswitch-users] Conferencing solution - need advice
>>>>>>
>>>>>>
>>>>>>
>>>>>> Hi Mike, thanks for looking into this, yes all servers will be active all
>>>>>> time running on high end virtual environment
>>>>>> Thanks
>>>>>>
>>>>>>
>>>>>> Sent from my iPhone
>>>>>>
>>>>>> On 22 Aug 2013, at 00:09, Michael Collins <msc at freeswitch.org> wrote:
>>>>>>
>>>>>>
>>>>>> Will the four individual conferences always be connected to each other?
>>>>>>> -MC
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Wed, Aug 21, 2013 at 1:55 PM, kaleem rehman <k4kaleem at gmail.com> wrote:
>>>>>>>
>>>>>>> Hi All,
>>>>>>>>
>>>>>>>> I want to implement a conferencing solution using freeswitch with Avaya.
>>>>>>>>
>>>>>>>> We currently have a multi-country hosted telephony platform, I want to
>>>>>>>> introduce freeswitch to it for internal use to see how it goes.
>>>>>>>>
>>>>>>>> I want a freeswitch server in UK, one in Europe, one in Americas and one
>>>>>>>> in South Africa.
>>>>>>>>
>>>>>>>> i would like all freeswitch server to talk to each other over SIP over
>>>>>>>> internal Cloud then i want UK conference users to join to UK freeswitch
>>>>>>>> and for same conference other countries users to join their local
>>>>>>>> freeswitch server and talk together.
>>>>>>>>
>>>>>>>> this way i want to achieve less cloud load and in reality one call
>>>>>>>> between all servers where users all users from their country will
>>>>>>>> connect to local freeswitch and that freeswitch links then to a
>>>>>>>> conference with other country.
>>>>>>>>
>>>>>>>> please advise on how i can achieve this.
>>>>>>>>
>>>>>>>> i am open to suggestions and ideas.
>>>>>>>>
>>>>>>>> regards,
>>>>>>>> Kaleem
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Michael S Collins
>>>>>>> Twitter: @mercutioviz
>>>>>>> http://www.FreeSWITCH.org
>>>>>>> http://www.ClueCon.com
>>>>>>> http://www.OSTAG.org
>>>>>>>
>>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>>
>>>>
>>>>
>>>
>>>_________________________________________________________________________
>>>Professional FreeSWITCH Consulting Services:
>>>consulting at freeswitch.org
>>>http://www.freeswitchsolutions.com
>>>
>>>
>>>
>>>
>>>Official FreeSWITCH Sites
>>>http://www.freeswitch.org
>>>http://wiki.freeswitch.org
>>>http://www.cluecon.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
>>>
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
>>consulting at freeswitch.org
>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
>>http://www.freeswitch.org
>>http://wiki.freeswitch.org
>>http://www.cluecon.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
>>
>>
>>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 mike at jerris.com  Fri Aug 23 01:00:00 2013
From: mike at jerris.com (Michael Jerris)
Date: Thu, 22 Aug 2013 17:00:00 -0400
Subject: [Freeswitch-users] You guys are terrible
In-Reply-To: <f6cffc7f0becfe253c8bfe4761781315@mstn.com>
References: <f6cffc7f0becfe253c8bfe4761781315@mstn.com>
Message-ID: <18B289D4-FDB4-403A-BE7A-18CCFD1BFA34@jerris.com>

Its the nightmare transfer on elm street?  For what its worth? sip shouldn't allow transfers to be done like this? the phones should really just "do the right thing" spike lee style when someone hangs up mid attended transfer, but most don't for some reason.

Mike

On Aug 21, 2013, at 8:00 PM, Jose Suero <ms at mstn.com> wrote:

> I'm swimming in a freeswitch log looking for a problem when all of the 
> sudden this comes up:
> 
> 2013-08-21 19:51:08.002385 [DEBUG] sofia.c:6928 1 .. 2 .. Freddie's 
> commin' for you...
> 
> Didn't find the freeswitch problem, but now have a machete masked 
> fellow on his way :-)
> 
> 
> Made me smile



From anthony.minessale at gmail.com  Fri Aug 23 01:11:52 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 22 Aug 2013 16:11:52 -0500
Subject: [Freeswitch-users] You guys are terrible
In-Reply-To: <18B289D4-FDB4-403A-BE7A-18CCFD1BFA34@jerris.com>
References: <f6cffc7f0becfe253c8bfe4761781315@mstn.com>
	<18B289D4-FDB4-403A-BE7A-18CCFD1BFA34@jerris.com>
Message-ID: <CAKbxfG80O2phmOfytL2tmN3XuOsTS6bXD_cMSKmF0LA8bv4v8Q@mail.gmail.com>

This is the one where you do an att xfer to BOX A but the call you need to
replace is on BOX B, so you have to call BOX B with invite +replaces then
xfer onto that new call.
ITs a nightmare cos its really complicated  and breaks all the time.  I
don't even have a way to test it very often.  you need to have a proxy that
places one call from the phone to BOX A and the next to BOX B.

Good old bug 3710 from asterisk days on mantis



On Thu, Aug 22, 2013 at 4:00 PM, Michael Jerris <mike at jerris.com> wrote:

> Its the nightmare transfer on elm street?  For what its worth? sip
> shouldn't allow transfers to be done like this? the phones should really
> just "do the right thing" spike lee style when someone hangs up mid
> attended transfer, but most don't for some reason.
>
> Mike
>
> On Aug 21, 2013, at 8:00 PM, Jose Suero <ms at mstn.com> wrote:
>
> > I'm swimming in a freeswitch log looking for a problem when all of the
> > sudden this comes up:
> >
> > 2013-08-21 19:51:08.002385 [DEBUG] sofia.c:6928 1 .. 2 .. Freddie's
> > commin' for you...
> >
> > Didn't find the freeswitch problem, but now have a machete masked
> > fellow on his way :-)
> >
> >
> > Made me smile
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From khalid.hosein at platform28.com  Thu Aug 22 21:51:07 2013
From: khalid.hosein at platform28.com (Khalid Hosein)
Date: Thu, 22 Aug 2013 13:51:07 -0400
Subject: [Freeswitch-users] Fwd: A FreeSWITCH Nagios plugin
In-Reply-To: <CAH0trcv-2=i6PwefpN2vtKdhr1woC7PNUOhXaQxUbFCe9GdixA@mail.gmail.com>
References: <CAH0trcv-2=i6PwefpN2vtKdhr1woC7PNUOhXaQxUbFCe9GdixA@mail.gmail.com>
Message-ID: <CAH0trctM+7AtvywWXq_18xpMb0Ddoe729jtVgXFKBQvYUyL8Sg@mail.gmail.com>

Hello everyone,

Like others before me, I've looked around for a Nagios plugin to monitor
FreeSWITCH, but didn't find anything that met our needs. There were some
discussions from years ago on this list (see links below), but no plugins
or scripts.


http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/064600.html

http://lists.freeswitch.org/pipermail/freeswitch-users/2010-April/055640.html

A search for 'freeswitch' on the Nagios Exchange also comes up empty.

Since we really needed to use Nagios to monitor our FS servers, I had to
build something, which I did in Perl. It uses Ton Voon's excellent
Nagios::Plugin Perl module.

Admittedly, it's limited in features, but if what you need to be monitored
can be gleaned from "fs_cli -x" calls, then this plugin can be easily
extended to do it.

I'm a complete FreeSWITCH newb, so I may have gone about this entirely the
wrong way, so please go easy on me ;-)

  https://github.com/kjhosein/nagios-freeswitch-plugin

I do hope this helps someone out there.
Cheers,

_Khalid
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From julesa at pcf.com  Thu Aug 22 21:58:09 2013
From: julesa at pcf.com (Jules Agee)
Date: Thu, 22 Aug 2013 10:58:09 -0700
Subject: [Freeswitch-users] You guys are terrible
In-Reply-To: <A70AC0C1B7E7FD478A6BF19705461AA0C8CA@IMBER.smb.curriegrad2004.ca>
References: <f6cffc7f0becfe253c8bfe4761781315@mstn.com>
	<CAHgk4BXPSFda7TABZKKeoB3DkaAXSDsoJfnnk9k_snSmKR-cFQ@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C8CA@IMBER.smb.curriegrad2004.ca>
Message-ID: <52165131.2010006@pcf.com>

I think it's a reference to Pink Floyd's "The Wall."

On 08/21/2013 06:54 PM, Jeff Leung wrote:
>
> I've seen worse on what sofia sometimes does when it wants to quit on us.
>
> More like a suicide note from sofia is what I've seen before
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org 
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of 
> *Brian Foster
> *Sent:* Wednesday, August 21, 2013 5:22 PM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] You guys are terrible
>
> Btw look in the source code around line 6928 might give you a clue as 
> to what's happening.
>
> Thank you,
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com <mailto:bdfoster at davri.com>
> Indianapolis, Indiana
>
> Sent from a mobile device.
>
> On Aug 21, 2013 8:05 PM, "Jose Suero" <ms at mstn.com 
> <mailto:ms at mstn.com>> wrote:
>
> I'm swimming in a freeswitch log looking for a problem when all of the
> sudden this comes up:
>
> 2013-08-21 19:51:08.002385 [DEBUG] sofia.c:6928 1 .. 2 .. Freddie's
> commin' for you...
>
> Didn't find the freeswitch problem, but now have a machete masked
> fellow on his way :-)
>
>
> Made me smile
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
--
Jules Agee
Senior System Administrator
Pacific Coast Feather Co.
julesa at pcf.com       x284

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From jayachar88 at gmail.com  Thu Aug 22 22:16:55 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Thu, 22 Aug 2013 23:46:55 +0530
Subject: [Freeswitch-users] Help understand error message -- gsmopen fails
Message-ID: <CAHDL-szUF1yMzuYoK5VOe5phhpMHVmhyxRoyPM4_cHo28fXC2Q@mail.gmail.com>

Latest FS (Git HEAD, as of yesterday), cloned and built successfully with
mod_gsmopen. Have used a fully unlocked E1550 dongle (tested in Mobile
Partner softeware in Windows, able to make calls, send SMSs). Started FS
with default configurations, as per Wiki instructions. Using the default
configration for gsmopen as well (maybe, that is the problem) ???

I get the highlighted errors, while loading mod_gsmopen ! My naive
interpretation (started with FS yesterday, and no experience with Asterisk
either), is that in the gsmopen module default config, I find:

  <per_interface_settings>
    <interface id="1" name="gsm01">
      <param name="controldevice_name" value="/dev/ttyUSB3"/>
      <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>

However from the error logs printed in FS console, I see:

4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
f.tty_data_device = |/dev/ttyUSB2|
2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
f.tty_audio_device = |/dev/ttyUSB1|

Probably that is autodetected !! But how, without the IMSI/IMEI being
specified?

---

freeswitch at dabbian1> load mod_gsmopen
2013-08-22 23:29:51.807103 [INFO] mod_enum.c:876 ENUM Reloaded
2013-08-22 23:29:51.847112 [INFO] switch_time.c:1191 Timezone reloaded 530
definitions
2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1106 rev
4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1106 ][none      ][-1,-1,-1] GSMOPEN
Charset Output Test 0 ?????
2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1107 rev
4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1107 ][none      ][-1,-1,-1] GSMOPEN
Charset Output Test 1 ???^?
2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1108 rev
4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1108 ][none      ][-1,-1,-1] GSMOPEN
Charset Output Test 2 ???????
2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1109 rev
4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1109 ][none      ][-1,-1,-1] GSMOPEN
Charset Output Test 3 ??????
2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1110 rev
4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1110 ][none      ][-1,-1,-1] GSMOPEN
Charset Output Test 4 ???
2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1111 rev
4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1111 ][none      ][-1,-1,-1] GSMOPEN
Charset Output Test 5 ??
2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1112 rev
4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1112 ][none      ][-1,-1,-1] GSMOPEN
Charset Output Test 6 ??
2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3053 rev
4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3053 ][none      ][-1,-1,-1]
************************************************
2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3054 rev
4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3054 ][none      ][-1,-1,-1]
f.imei=|354638041679399|
2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3055 rev
4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3055 ][none      ][-1,-1,-1]
f.imsi=|405034007414619|
2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3056 rev
4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
f.tty_data_device = |/dev/ttyUSB2|
2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
f.tty_audio_device = |/dev/ttyUSB1|
2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3058 rev
4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3058 ][none      ][-1,-1,-1]
************************************************
2013-08-22 23:29:53.027123 [WARNING] mod_gsmopen.cpp:1589 rev
4a3d1a0|4a3d1a0[(nil)|37     ][WARNINGA  1589 ][gsm01     ][-1, 0, 0]
STARTING interface_id=1
2013-08-22 23:29:53.027123 [ERR] gsmopen_protocol.cpp:137 rev
4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
/dev/ttyUSB3, NOT open
2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:1608 rev
4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
STARTING interface_id=1 FAILED: gsmopen_serial_init failed

+OK Reloading XML
+OK

2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:2684 rev
4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
on interface gsm01:
freeswitch at dabbian1> 2013-08-22 23:29:53.027123 [CONSOLE]
switch_loadable_module.c:1401 Successfully Loaded [mod_gsmopen]
2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:147 Adding
Endpoint 'gsmopen'
2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding API
Function 'gsm'
2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding API
Function 'gsmopen'
2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding API
Function 'gsmopen_boost_audio'
2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding API
Function 'gsmopen_dump'
2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding API
Function 'gsmopen_sendsms'
2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:417 Adding
Chat interface 'sms'

freeswitch at dabbian1>

---


With this understanding, I went ahead and modified the gsmopen config file,
to:
  <per_interface_settings>
    <interface id="1" name="gsm01">
      <param name="controldevice_name" value="/dev/ttyUSB2"/>
      <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>

then unloaded mod_gsmopen, and reloaded it, but now, I get this error:

2013-08-22 23:41:43.587104 [ERR] gsmopen_protocol.cpp:137 rev
4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
/dev/ttyUSB3, NOT open
2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:1608 rev
4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
STARTING interface_id=1 FAILED: gsmopen_serial_init failed
2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:2684 rev
4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
on interface gsm01:

What does it need /dev/ttyUSB3 for ?

Maybe -- after all, my non linear progression into setting up FS and using
it by somewhat of trial-n-error isn't helping!

thanks,
Jay
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From brian at freeswitch.org  Fri Aug 23 01:31:36 2013
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Aug 2013 16:31:36 -0500
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <CAHgk4BWdp9iQHsZfQmNWu0JLzidQQCtGn0kzD_AAyapKOohzQA@mail.gmail.com>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
	<CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
	<B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
	<CAHgk4BWdp9iQHsZfQmNWu0JLzidQQCtGn0kzD_AAyapKOohzQA@mail.gmail.com>
Message-ID: <7E49300F-E95E-4970-A58F-F9FD48CACAFA@freeswitch.org>

It will work on T.38 just fine as long as ECM is supported.

/b

On Aug 21, 2013, at 1:51 PM, Brian Foster <bdfoster at davri.com> wrote:

> Damn. T38 only here.
> 
> Thank you,
> 
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana
> 
> Sent from a mobile device.
> 
> On Aug 21, 2013 2:42 PM, "Gilad Abada" <ga at steadfasttelecom.com> wrote:
> Hey Brian,
> 
> 
> -- 
> 
> 
> SteadFast Telecommunications, Inc.
> 
> Call us to find out how much you can save with VoIP!
>                  
> V: 212.589.1001
> 
>  
> For over 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises. Make your virtual office a reality. Enjoy the freedom to travel while remaining connected to your office.
> 
> On Wednesday, August 21, 2013 at 2:28 PM, Jo?o Mesquita wrote:
> 
>> :( Argentina won't suffice? :D
>> 
>> Jo?o Mesquita
>> FreeSWITCH? Solutions
>> 
>> 
>> On Wed, Aug 21, 2013 at 3:18 PM, Brian West <brian at freeswitch.org> wrote:
>>> Dear FreeSWITCHERS,
>>>         If you have a color fax machine with REAL COPPER OLD SCHOOL PSTN connectivity please respond with your phone number... I want to test if I can send you a color fax .. that is if you wanna test it out with me pretty please.  (US ONLY)
>>> 
>>> Thanks,
>>> --
>>> Brian West
>>> brian at freeswitch.org
>>> FreeSWITCH Solutions, LLC
>>> PO BOX PO BOX 2531
>>> Brookfield, WI 53008-2531
>>> Twitter: @FreeSWITCH_Wire (Follow us today!)
>>> http://www.freeswitchbook.com
>>> http://www.freeswitchcookbook.com
>>> 
>>> T: +1.918.420.9001  |  F: +1.918.420.9002  |  M: +1.918.424.WEST
>>> iNUM: +883 5100 1420 9001
>>> ISN: 410*543
>>> Skype:briankwest
>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 brian at freeswitch.org  Fri Aug 23 01:31:57 2013
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Aug 2013 16:31:57 -0500
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <CAC-LwPPCLeX80DLcutwqQi4JDzJazLChivT=mPSXjTi6J19xdg@mail.gmail.com>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
	<CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
	<B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
	<CAHgk4BWdp9iQHsZfQmNWu0JLzidQQCtGn0kzD_AAyapKOohzQA@mail.gmail.com>
	<CAC-LwPPCLeX80DLcutwqQi4JDzJazLChivT=mPSXjTi6J19xdg@mail.gmail.com>
Message-ID: <95706213-7DBF-49F7-A1DC-B0D8C559695B@freeswitch.org>

But it can.  I did one yesterday LULZ!


On Aug 21, 2013, at 3:28 PM, Vik Killa <vipkilla at gmail.com> wrote:

> FS cannot send color faxes yet?
> 
> <snip>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 brian at freeswitch.org  Fri Aug 23 01:32:49 2013
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Aug 2013 16:32:49 -0500
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <CAHgk4BV=VoOZY3R2jR+eyRkDK=HYYeNF+EjuAsD2yjL0gK_DHQ@mail.gmail.com>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
	<CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
	<B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
	<CAHgk4BWdp9iQHsZfQmNWu0JLzidQQCtGn0kzD_AAyapKOohzQA@mail.gmail.com>
	<CAC-LwPPCLeX80DLcutwqQi4JDzJazLChivT=mPSXjTi6J19xdg@mail.gmail.com>
	<52157852.7010604@coppice.org>
	<CAHgk4BV=VoOZY3R2jR+eyRkDK=HYYeNF+EjuAsD2yjL0gK_DHQ@mail.gmail.com>
Message-ID: <5CD9F9EC-8510-4865-8B98-8EBE86B87992@freeswitch.org>

Faxes will not die anytime in the next 30 years.

On Aug 22, 2013, at 10:58 AM, Brian Foster <bdfoster at davri.com> wrote:

> The very definition of awesome. I can't believe I'm this excited over a (supposedly) dying technology.
> 
> Thank you,
> 
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana
> 
> Sent from a mobile device.
> 
> On Aug 21, 2013 10:38 PM, "Steve Underwood" <steveu at coppice.org> wrote:
> On 08/22/2013 04:28 AM, Vik Killa wrote:
> > FS cannot send color faxes yet?
> >
> Very few computer FAX platforms can send or receive colour FAXes. As of
> this week FS can, but the support should be treated as experimental at
> this stage.
> 
> It seems like colour FAX should be trivial, if you have error corrected
> FAX support, but it isn't. The developers of the colour FAX standard
> chose to use JPEG compression, but with a very unusual colour
> representation. This means you need to develop a messy bunch of code if
> you want the user to be able to work with conventional colour image
> files on their computers. It is this support which we are shaking out
> now. When its looking stable I will add the relevant information about
> controlling colour FAXes to the wiki pages.
> 
> Regards,
> Steve
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 brian at freeswitch.org  Fri Aug 23 01:33:20 2013
From: brian at freeswitch.org (Brian West)
Date: Thu, 22 Aug 2013 16:33:20 -0500
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <CAC-LwPM1StLEQW9ZvjY1T5nEk32SMbBqLLTNNK4DKe8YaRcf9Q@mail.gmail.com>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
	<CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
	<B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
	<CAHgk4BWdp9iQHsZfQmNWu0JLzidQQCtGn0kzD_AAyapKOohzQA@mail.gmail.com>
	<CAC-LwPPCLeX80DLcutwqQi4JDzJazLChivT=mPSXjTi6J19xdg@mail.gmail.com>
	<52157852.7010604@coppice.org>
	<CAHgk4BV=VoOZY3R2jR+eyRkDK=HYYeNF+EjuAsD2yjL0gK_DHQ@mail.gmail.com>
	<CAC-LwPM1StLEQW9ZvjY1T5nEk32SMbBqLLTNNK4DKe8YaRcf9Q@mail.gmail.com>
Message-ID: <F01D2610-0500-4763-BECB-D9E3EB181904@freeswitch.org>

HEY there is this think called EMAIL.. I'm sure you've heard about it! lulu

/b
PS it will work if you're that crazy.

On Aug 22, 2013, at 12:47 PM, Vik Killa <vipkilla at gmail.com> wrote:

> What about color faxing from FS to FS? 
> <snip>

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From msc at freeswitch.org  Fri Aug 23 01:36:25 2013
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 22 Aug 2013 14:36:25 -0700
Subject: [Freeswitch-users] Fwd: A FreeSWITCH Nagios plugin
In-Reply-To: <CAH0trctM+7AtvywWXq_18xpMb0Ddoe729jtVgXFKBQvYUyL8Sg@mail.gmail.com>
References: <CAH0trcv-2=i6PwefpN2vtKdhr1woC7PNUOhXaQxUbFCe9GdixA@mail.gmail.com>
	<CAH0trctM+7AtvywWXq_18xpMb0Ddoe729jtVgXFKBQvYUyL8Sg@mail.gmail.com>
Message-ID: <CAKzWOxX5FUS633iB+hp=QkDpG8BbtCCJbpRpT491JrLTVgC3VQ@mail.gmail.com>

Awesome, thanks for sharing! That code is really well laid out and easy to
read. Proof positive that Perl code need not be ugly and unreadable.

-MC


On Thu, Aug 22, 2013 at 10:51 AM, Khalid Hosein <
khalid.hosein at platform28.com> wrote:

>
> Hello everyone,
>
> Like others before me, I've looked around for a Nagios plugin to monitor
> FreeSWITCH, but didn't find anything that met our needs. There were some
> discussions from years ago on this list (see links below), but no plugins
> or scripts.
>
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/064600.html
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-April/055640.html
>
> A search for 'freeswitch' on the Nagios Exchange also comes up empty.
>
> Since we really needed to use Nagios to monitor our FS servers, I had to
> build something, which I did in Perl. It uses Ton Voon's excellent
> Nagios::Plugin Perl module.
>
> Admittedly, it's limited in features, but if what you need to be monitored
> can be gleaned from "fs_cli -x" calls, then this plugin can be easily
> extended to do it.
>
> I'm a complete FreeSWITCH newb, so I may have gone about this entirely the
> wrong way, so please go easy on me ;-)
>
>   https://github.com/kjhosein/nagios-freeswitch-plugin
>
> I do hope this helps someone out there.
> Cheers,
>
> _Khalid
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From mike at jerris.com  Fri Aug 23 01:46:28 2013
From: mike at jerris.com (Michael Jerris)
Date: Thu, 22 Aug 2013 17:46:28 -0400
Subject: [Freeswitch-users] Fwd: A FreeSWITCH Nagios plugin
In-Reply-To: <CAKzWOxX5FUS633iB+hp=QkDpG8BbtCCJbpRpT491JrLTVgC3VQ@mail.gmail.com>
References: <CAH0trcv-2=i6PwefpN2vtKdhr1woC7PNUOhXaQxUbFCe9GdixA@mail.gmail.com>
	<CAH0trctM+7AtvywWXq_18xpMb0Ddoe729jtVgXFKBQvYUyL8Sg@mail.gmail.com>
	<CAKzWOxX5FUS633iB+hp=QkDpG8BbtCCJbpRpT491JrLTVgC3VQ@mail.gmail.com>
Message-ID: <7E86861C-22A5-48E0-B6D6-9C6A1257D5EB@jerris.com>

Its readable?  I can fix that!!

On Aug 22, 2013, at 5:36 PM, Michael Collins <msc at freeswitch.org> wrote:

> Awesome, thanks for sharing! That code is really well laid out and easy to read. Proof positive that Perl code need not be ugly and unreadable.
> 
> -MC
> 
> 
> On Thu, Aug 22, 2013 at 10:51 AM, Khalid Hosein <khalid.hosein at platform28.com> wrote:
> 
> Hello everyone,
> 
> Like others before me, I've looked around for a Nagios plugin to monitor FreeSWITCH, but didn't find anything that met our needs. There were some discussions from years ago on this list (see links below), but no plugins or scripts. 
> 
>   http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/064600.html
>   http://lists.freeswitch.org/pipermail/freeswitch-users/2010-April/055640.html
> 
> A search for 'freeswitch' on the Nagios Exchange also comes up empty.
> 
> Since we really needed to use Nagios to monitor our FS servers, I had to build something, which I did in Perl. It uses Ton Voon's excellent Nagios::Plugin Perl module.
> 
> Admittedly, it's limited in features, but if what you need to be monitored can be gleaned from "fs_cli -x" calls, then this plugin can be easily extended to do it. 
> 
> I'm a complete FreeSWITCH newb, so I may have gone about this entirely the wrong way, so please go easy on me ;-) 
> 
>   https://github.com/kjhosein/nagios-freeswitch-plugin
> 
> I do hope this helps someone out there.
> Cheers,
> 
> _Khalid
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> 
> 
> -- 
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveayre at gmail.com  Fri Aug 23 01:48:24 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Thu, 22 Aug 2013 22:48:24 +0100
Subject: [Freeswitch-users] Fwd: A FreeSWITCH Nagios plugin
In-Reply-To: <CAH0trctM+7AtvywWXq_18xpMb0Ddoe729jtVgXFKBQvYUyL8Sg@mail.gmail.com>
References: <CAH0trcv-2=i6PwefpN2vtKdhr1woC7PNUOhXaQxUbFCe9GdixA@mail.gmail.com>
	<CAH0trctM+7AtvywWXq_18xpMb0Ddoe729jtVgXFKBQvYUyL8Sg@mail.gmail.com>
Message-ID: <CAFiqYukkmKZu=2T0BqjbsVOgSiKixtpvh6S0zOb8CPwYd7UnTw@mail.gmail.com>

Some monitoring can also be gathered from mod_snmp (contributed by
pressureman). It registers with snmpd via AgentX and would let you remotely
query FS via SNMP.

It's limited in the information available (it does have at
least calls/channels and basic status though). Any patches to add more
would I'm sure be welcome.


On Thursday, August 22, 2013, Khalid Hosein wrote:

>
> Hello everyone,
>
> Like others before me, I've looked around for a Nagios plugin to monitor
> FreeSWITCH, but didn't find anything that met our needs. There were some
> discussions from years ago on this list (see links below), but no plugins
> or scripts.
>
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/064600.html
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-April/055640.html
>
> A search for 'freeswitch' on the Nagios Exchange also comes up empty.
>
> Since we really needed to use Nagios to monitor our FS servers, I had to
> build something, which I did in Perl. It uses Ton Voon's excellent
> Nagios::Plugin Perl module.
>
> Admittedly, it's limited in features, but if what you need to be monitored
> can be gleaned from "fs_cli -x" calls, then this plugin can be easily
> extended to do it.
>
> I'm a complete FreeSWITCH newb, so I may have gone about this entirely the
> wrong way, so please go easy on me ;-)
>
>   https://github.com/kjhosein/nagios-freeswitch-plugin
>
> I do hope this helps someone out there.
> Cheers,
>
> _Khalid
>
>
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From bjamm90 at gmail.com  Fri Aug 23 01:53:37 2013
From: bjamm90 at gmail.com (bjamm90)
Date: Thu, 22 Aug 2013 14:53:37 -0700 (PDT)
Subject: [Freeswitch-users]
 =?utf-8?q?Error_message_=22_=5BERR=5D_switch?=
 =?utf-8?q?=5Fcpp=2Ecpp=3A48_Cannot_queue_any_more_events=E2=80=9C?=
In-Reply-To: <CAKbxfG-gXMvkWFn6AhTU-OJw13oypPmeC25Y8FOPwRzPfMED0w@mail.gmail.com>
References: <1339645894596-7579817.post@n2.nabble.com>
	<CAKbxfG9vD5KjpiPC2tMhnhHA-jGc-TkVpG7CF-7wgw84rdwvRQ@mail.gmail.com>
	<1368502318906-7590651.post@n2.nabble.com>
	<CAKbxfG-gXMvkWFn6AhTU-OJw13oypPmeC25Y8FOPwRzPfMED0w@mail.gmail.com>
Message-ID: <1377208417437-7594218.post@n2.nabble.com>

I am also having the same issue. Thank you for the explanation Anthony.
Unfortunately I am a noob and am not sure how to change this limit.

Could you provide a few short steps to get me started. Commands would be
greatly appreciated.

Cheers!



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-message-ERR-switch-cpp-cpp-48-Cannot-queue-any-more-events-tp7579817p7594218.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From grcamauer at gmail.com  Fri Aug 23 02:10:05 2013
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Thu, 22 Aug 2013 19:10:05 -0300
Subject: [Freeswitch-users]
	=?windows-1252?q?Error_message_=22_=5BERR=5D_s?=
	=?windows-1252?q?witch=5Fcpp=2Ecpp=3A48_Cannot_queue_any_more_even?=
	=?windows-1252?q?ts=93?=
In-Reply-To: <1377208417437-7594218.post@n2.nabble.com>
References: <1339645894596-7579817.post@n2.nabble.com>
	<CAKbxfG9vD5KjpiPC2tMhnhHA-jGc-TkVpG7CF-7wgw84rdwvRQ@mail.gmail.com>
	<1368502318906-7590651.post@n2.nabble.com>
	<CAKbxfG-gXMvkWFn6AhTU-OJw13oypPmeC25Y8FOPwRzPfMED0w@mail.gmail.com>
	<1377208417437-7594218.post@n2.nabble.com>
Message-ID: <CAPNTnnoUFyQKFjpOFt4py7M4pFakKEnQ-G6jBUPio9O+r73qnA@mail.gmail.com>

Also, make sure that you are filtering events and only receive those that
you actually need.

Guillermo


On Thu, Aug 22, 2013 at 6:53 PM, bjamm90 <bjamm90 at gmail.com> wrote:

> I am also having the same issue. Thank you for the explanation Anthony.
> Unfortunately I am a noob and am not sure how to change this limit.
>
> Could you provide a few short steps to get me started. Commands would be
> greatly appreciated.
>
> Cheers!
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Error-message-ERR-switch-cpp-cpp-48-Cannot-queue-any-more-events-tp7579817p7594218.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
Guillermo Ruiz Camauer
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From bdfoster at davri.com  Fri Aug 23 02:24:23 2013
From: bdfoster at davri.com (Brian Foster)
Date: Thu, 22 Aug 2013 18:24:23 -0400
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <5CD9F9EC-8510-4865-8B98-8EBE86B87992@freeswitch.org>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
	<CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
	<B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
	<CAHgk4BWdp9iQHsZfQmNWu0JLzidQQCtGn0kzD_AAyapKOohzQA@mail.gmail.com>
	<CAC-LwPPCLeX80DLcutwqQi4JDzJazLChivT=mPSXjTi6J19xdg@mail.gmail.com>
	<52157852.7010604@coppice.org>
	<CAHgk4BV=VoOZY3R2jR+eyRkDK=HYYeNF+EjuAsD2yjL0gK_DHQ@mail.gmail.com>
	<5CD9F9EC-8510-4865-8B98-8EBE86B87992@freeswitch.org>
Message-ID: <CAHgk4BUh9Op1rH594Z3Y+ddVB6zBjNsje3aVro6=MoYfDw9o5A@mail.gmail.com>

Hey many people who I work with have no clue what email is... hence the
reason I receive 10-20 faxes a day just for me.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 22, 2013 5:48 PM, "Brian West" <brian at freeswitch.org> wrote:

> Faxes will not die anytime in the next 30 years.
>
> On Aug 22, 2013, at 10:58 AM, Brian Foster <bdfoster at davri.com> wrote:
>
> > The very definition of awesome. I can't believe I'm this excited over a
> (supposedly) dying technology.
> >
> > Thank you,
> >
> > Brian Foster
> > Project Manager/Owner's Rep.
> > Davri Investments, Inc.
> > O: 317-787-2686 x2102
> > M: 317-600-9753
> > E: bdfoster at davri.com
> > Indianapolis, Indiana
> >
> > Sent from a mobile device.
> >
> > On Aug 21, 2013 10:38 PM, "Steve Underwood" <steveu at coppice.org> wrote:
> > On 08/22/2013 04:28 AM, Vik Killa wrote:
> > > FS cannot send color faxes yet?
> > >
> > Very few computer FAX platforms can send or receive colour FAXes. As of
> > this week FS can, but the support should be treated as experimental at
> > this stage.
> >
> > It seems like colour FAX should be trivial, if you have error corrected
> > FAX support, but it isn't. The developers of the colour FAX standard
> > chose to use JPEG compression, but with a very unusual colour
> > representation. This means you need to develop a messy bunch of code if
> > you want the user to be able to work with conventional colour image
> > files on their computers. It is this support which we are shaking out
> > now. When its looking stable I will add the relevant information about
> > controlling colour FAXes to the wiki pages.
> >
> > Regards,
> > Steve
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 red.rain.seven at gmail.com  Fri Aug 23 04:04:25 2013
From: red.rain.seven at gmail.com (Henry Huang)
Date: Thu, 22 Aug 2013 17:04:25 -0700
Subject: [Freeswitch-users] Fwd: A FreeSWITCH Nagios plugin
In-Reply-To: <CAH0trctM+7AtvywWXq_18xpMb0Ddoe729jtVgXFKBQvYUyL8Sg@mail.gmail.com>
References: <CAH0trcv-2=i6PwefpN2vtKdhr1woC7PNUOhXaQxUbFCe9GdixA@mail.gmail.com>
	<CAH0trctM+7AtvywWXq_18xpMb0Ddoe729jtVgXFKBQvYUyL8Sg@mail.gmail.com>
Message-ID: <CAP_G394uo45u-Fsw1j08SnSGHMeNiRWb514gwv-Pa_x8N+qWGA@mail.gmail.com>

Thank you for sharing. I am currently working on similar check with Nagios
using Bash. But this looks good and has the options ready for me to just
add additional commands that I want to check on FS.

Henry


On Thu, Aug 22, 2013 at 10:51 AM, Khalid Hosein <
khalid.hosein at platform28.com> wrote:

>
> Hello everyone,
>
> Like others before me, I've looked around for a Nagios plugin to monitor
> FreeSWITCH, but didn't find anything that met our needs. There were some
> discussions from years ago on this list (see links below), but no plugins
> or scripts.
>
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/064600.html
>
> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-April/055640.html
>
> A search for 'freeswitch' on the Nagios Exchange also comes up empty.
>
> Since we really needed to use Nagios to monitor our FS servers, I had to
> build something, which I did in Perl. It uses Ton Voon's excellent
> Nagios::Plugin Perl module.
>
> Admittedly, it's limited in features, but if what you need to be monitored
> can be gleaned from "fs_cli -x" calls, then this plugin can be easily
> extended to do it.
>
> I'm a complete FreeSWITCH newb, so I may have gone about this entirely the
> wrong way, so please go easy on me ;-)
>
>   https://github.com/kjhosein/nagios-freeswitch-plugin
>
> I do hope this helps someone out there.
> Cheers,
>
> _Khalid
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Fri Aug 23 04:49:53 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Fri, 23 Aug 2013 06:19:53 +0530
Subject: [Freeswitch-users] Help understand error message -- gsmopen
	fails
In-Reply-To: <CAHDL-szUF1yMzuYoK5VOe5phhpMHVmhyxRoyPM4_cHo28fXC2Q@mail.gmail.com>
References: <CAHDL-szUF1yMzuYoK5VOe5phhpMHVmhyxRoyPM4_cHo28fXC2Q@mail.gmail.com>
Message-ID: <CAMvhjJu7-jrHFzZJdMu+A8Leg_Dc2fy-HnT6jK9BdbxXOaFjbw@mail.gmail.com>

Hi jayanth
Since you are the first person whom i have seen working with mod_gsmopen.
Can you help me with the ways to enable freeswitch use mod_gsmopen on
ubuntu machine.
This may sound ironical that i am not answering your question rather asking
a question as a reply of your question.
I will send you the error log which i come across after loading mod_gsmopen
on freeswitch.
Hope we can together help each other.

Regards
Ashish Mishra
On Aug 23, 2013 3:12 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:

> Latest FS (Git HEAD, as of yesterday), cloned and built successfully with
> mod_gsmopen. Have used a fully unlocked E1550 dongle (tested in Mobile
> Partner softeware in Windows, able to make calls, send SMSs). Started FS
> with default configurations, as per Wiki instructions. Using the default
> configration for gsmopen as well (maybe, that is the problem) ???
>
> I get the highlighted errors, while loading mod_gsmopen ! My naive
> interpretation (started with FS yesterday, and no experience with Asterisk
> either), is that in the gsmopen module default config, I find:
>
>   <per_interface_settings>
>     <interface id="1" name="gsm01">
>       <param name="controldevice_name" value="/dev/ttyUSB3"/>
>       <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>
> However from the error logs printed in FS console, I see:
>
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
> f.tty_data_device = |/dev/ttyUSB2|
> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
> f.tty_audio_device = |/dev/ttyUSB1|
>
> Probably that is autodetected !! But how, without the IMSI/IMEI being
> specified?
>
> ---
>
> freeswitch at dabbian1> load mod_gsmopen
> 2013-08-22 23:29:51.807103 [INFO] mod_enum.c:876 ENUM Reloaded
> 2013-08-22 23:29:51.847112 [INFO] switch_time.c:1191 Timezone reloaded 530
> definitions
> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1106 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1106 ][none      ][-1,-1,-1] GSMOPEN
> Charset Output Test 0 ?????
> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1107 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1107 ][none      ][-1,-1,-1] GSMOPEN
> Charset Output Test 1 ???^?
> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1108 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1108 ][none      ][-1,-1,-1] GSMOPEN
> Charset Output Test 2 ???????
> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1109 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1109 ][none      ][-1,-1,-1] GSMOPEN
> Charset Output Test 3 ??????
> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1110 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1110 ][none      ][-1,-1,-1] GSMOPEN
> Charset Output Test 4 ???
> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1111 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1111 ][none      ][-1,-1,-1] GSMOPEN
> Charset Output Test 5 ??
> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1112 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1112 ][none      ][-1,-1,-1] GSMOPEN
> Charset Output Test 6 ??
> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3053 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3053 ][none      ][-1,-1,-1]
> ************************************************
> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3054 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3054 ][none      ][-1,-1,-1]
> f.imei=|354638041679399|
> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3055 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3055 ][none      ][-1,-1,-1]
> f.imsi=|405034007414619|
> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3056 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
> f.tty_data_device = |/dev/ttyUSB2|
> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
> f.tty_audio_device = |/dev/ttyUSB1|
> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3058 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3058 ][none      ][-1,-1,-1]
> ************************************************
> 2013-08-22 23:29:53.027123 [WARNING] mod_gsmopen.cpp:1589 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][WARNINGA  1589 ][gsm01     ][-1, 0, 0]
> STARTING interface_id=1
> 2013-08-22 23:29:53.027123 [ERR] gsmopen_protocol.cpp:137 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
> /dev/ttyUSB3, NOT open
> 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:1608 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
> STARTING interface_id=1 FAILED: gsmopen_serial_init failed
>
> +OK Reloading XML
> +OK
>
> 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:2684 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
> on interface gsm01:
> freeswitch at dabbian1> 2013-08-22 23:29:53.027123 [CONSOLE]
> switch_loadable_module.c:1401 Successfully Loaded [mod_gsmopen]
> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:147 Adding
> Endpoint 'gsmopen'
> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
> API Function 'gsm'
> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
> API Function 'gsmopen'
> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
> API Function 'gsmopen_boost_audio'
> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
> API Function 'gsmopen_dump'
> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
> API Function 'gsmopen_sendsms'
> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:417 Adding
> Chat interface 'sms'
>
> freeswitch at dabbian1>
>
> ---
>
>
> With this understanding, I went ahead and modified the gsmopen config
> file, to:
>   <per_interface_settings>
>     <interface id="1" name="gsm01">
>       <param name="controldevice_name" value="/dev/ttyUSB2"/>
>       <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>
> then unloaded mod_gsmopen, and reloaded it, but now, I get this error:
>
> 2013-08-22 23:41:43.587104 [ERR] gsmopen_protocol.cpp:137 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
> /dev/ttyUSB3, NOT open
> 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:1608 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
> STARTING interface_id=1 FAILED: gsmopen_serial_init failed
> 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:2684 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
> on interface gsm01:
>
> What does it need /dev/ttyUSB3 for ?
>
> Maybe -- after all, my non linear progression into setting up FS and using
> it by somewhat of trial-n-error isn't helping!
>
> thanks,
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 alex at opensystems.net.au  Fri Aug 23 04:55:22 2013
From: alex at opensystems.net.au (Alex Ynema)
Date: Fri, 23 Aug 2013 08:55:22 +0800
Subject: [Freeswitch-users] Fwd: A FreeSWITCH Nagios plugin
In-Reply-To: <CAP_G394uo45u-Fsw1j08SnSGHMeNiRWb514gwv-Pa_x8N+qWGA@mail.gmail.com>
References: <CAH0trcv-2=i6PwefpN2vtKdhr1woC7PNUOhXaQxUbFCe9GdixA@mail.gmail.com>
	<CAH0trctM+7AtvywWXq_18xpMb0Ddoe729jtVgXFKBQvYUyL8Sg@mail.gmail.com>
	<CAP_G394uo45u-Fsw1j08SnSGHMeNiRWb514gwv-Pa_x8N+qWGA@mail.gmail.com>
Message-ID: <CAN-21F76wyfOj+oCQxHDpyJQOFPX6dMXGhag2xxRBu1y_YZDdQ@mail.gmail.com>

I did do some similar work with opsview but it should work with Nagios as
it's built on it anyway.

http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-amp-Opsview-Monitoring-td7590796.html

Work well for me as the only real state I needed was the concurrent call
rate so we could compare it to the system resources and see how it was
performing with our call campaigns. Seems to be running happily presently
at 220 concurrent calls for only a 2x CPU & 16Gb Virtual Server


*Alex Ynema** *| IT Consultant
alex at opensystems.net.au

Level 1, 409-411 Oxford Street, Mount Hawthorne WA 6016
Office: +61 8 9427 2500
Mobile: +61 404 796 894
<http://www.ynema.net/>
<http://www.ynema.net/>IT Consultant for Open Systems Support
www.opensystems.net.au


On 23 August 2013 08:04, Henry Huang <red.rain.seven at gmail.com> wrote:

> Thank you for sharing. I am currently working on similar check with Nagios
> using Bash. But this looks good and has the options ready for me to just
> add additional commands that I want to check on FS.
>
> Henry
>
>
> On Thu, Aug 22, 2013 at 10:51 AM, Khalid Hosein <
> khalid.hosein at platform28.com> wrote:
>
>>
>> Hello everyone,
>>
>> Like others before me, I've looked around for a Nagios plugin to monitor
>> FreeSWITCH, but didn't find anything that met our needs. There were some
>> discussions from years ago on this list (see links below), but no plugins
>> or scripts.
>>
>>
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-October/064600.html
>>
>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-April/055640.html
>>
>> A search for 'freeswitch' on the Nagios Exchange also comes up empty.
>>
>> Since we really needed to use Nagios to monitor our FS servers, I had to
>> build something, which I did in Perl. It uses Ton Voon's excellent
>> Nagios::Plugin Perl module.
>>
>> Admittedly, it's limited in features, but if what you need to be
>> monitored can be gleaned from "fs_cli -x" calls, then this plugin can be
>> easily extended to do it.
>>
>> I'm a complete FreeSWITCH newb, so I may have gone about this entirely
>> the wrong way, so please go easy on me ;-)
>>
>>   https://github.com/kjhosein/nagios-freeswitch-plugin
>>
>> I do hope this helps someone out there.
>> Cheers,
>>
>> _Khalid
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Fri Aug 23 06:42:12 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Fri, 23 Aug 2013 08:12:12 +0530
Subject: [Freeswitch-users] Help understand error message -- gsmopen
	fails
In-Reply-To: <CAMvhjJu7-jrHFzZJdMu+A8Leg_Dc2fy-HnT6jK9BdbxXOaFjbw@mail.gmail.com>
References: <CAHDL-szUF1yMzuYoK5VOe5phhpMHVmhyxRoyPM4_cHo28fXC2Q@mail.gmail.com>
	<CAMvhjJu7-jrHFzZJdMu+A8Leg_Dc2fy-HnT6jK9BdbxXOaFjbw@mail.gmail.com>
Message-ID: <CAHDL-swo+_D_-bgwmW_jZOv-CJGkcxe7bPXbJEz-5nFAowf2Og@mail.gmail.com>

Looks like, not many people use mod_gsmopen. I've opened a JIRA ticket for
some minor issues in the Wiki, but hasn't been acknowledged, and I see no
one responding this mail either. I'd seen your other mails in the archive
while searching for hints in previous exchanges.

Ubuntu/Debian setups should be fairly similar. Did you follow the wiki to
build it from scratch or using the repo packages ? I guess the former,
because Debian repo packages didn't have mod_gsmopen.

I plan to spend a day or two reading about FS. The wiki has a tonne of info
but not in a very well organized fashion. Also, I saw that the JIRA bug
reports have lot of useful information in them.

BTW what are you trying to do with FS ? Set up an IVR ?


On Fri, Aug 23, 2013 at 6:19 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Hi jayanth
> Since you are the first person whom i have seen working with mod_gsmopen.
> Can you help me with the ways to enable freeswitch use mod_gsmopen on
> ubuntu machine.
> This may sound ironical that i am not answering your question rather
> asking a question as a reply of your question.
> I will send you the error log which i come across after loading
> mod_gsmopen on freeswitch.
> Hope we can together help each other.
>
> Regards
> Ashish Mishra
> On Aug 23, 2013 3:12 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:
>
>> Latest FS (Git HEAD, as of yesterday), cloned and built successfully with
>> mod_gsmopen. Have used a fully unlocked E1550 dongle (tested in Mobile
>> Partner softeware in Windows, able to make calls, send SMSs). Started FS
>> with default configurations, as per Wiki instructions. Using the default
>> configration for gsmopen as well (maybe, that is the problem) ???
>>
>> I get the highlighted errors, while loading mod_gsmopen ! My naive
>> interpretation (started with FS yesterday, and no experience with Asterisk
>> either), is that in the gsmopen module default config, I find:
>>
>>   <per_interface_settings>
>>     <interface id="1" name="gsm01">
>>       <param name="controldevice_name" value="/dev/ttyUSB3"/>
>>       <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>>
>> However from the error logs printed in FS console, I see:
>>
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
>> f.tty_data_device = |/dev/ttyUSB2|
>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
>> f.tty_audio_device = |/dev/ttyUSB1|
>>
>> Probably that is autodetected !! But how, without the IMSI/IMEI being
>> specified?
>>
>> ---
>>
>> freeswitch at dabbian1> load mod_gsmopen
>> 2013-08-22 23:29:51.807103 [INFO] mod_enum.c:876 ENUM Reloaded
>> 2013-08-22 23:29:51.847112 [INFO] switch_time.c:1191 Timezone reloaded
>> 530 definitions
>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1106 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1106 ][none      ][-1,-1,-1] GSMOPEN
>> Charset Output Test 0 ?????
>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1107 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1107 ][none      ][-1,-1,-1] GSMOPEN
>> Charset Output Test 1 ???^?
>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1108 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1108 ][none      ][-1,-1,-1] GSMOPEN
>> Charset Output Test 2 ???????
>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1109 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1109 ][none      ][-1,-1,-1] GSMOPEN
>> Charset Output Test 3 ??????
>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1110 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1110 ][none      ][-1,-1,-1] GSMOPEN
>> Charset Output Test 4 ???
>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1111 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1111 ][none      ][-1,-1,-1] GSMOPEN
>> Charset Output Test 5 ??
>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1112 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1112 ][none      ][-1,-1,-1] GSMOPEN
>> Charset Output Test 6 ??
>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3053 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3053 ][none      ][-1,-1,-1]
>> ************************************************
>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3054 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3054 ][none      ][-1,-1,-1]
>> f.imei=|354638041679399|
>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3055 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3055 ][none      ][-1,-1,-1]
>> f.imsi=|405034007414619|
>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3056 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
>> f.tty_data_device = |/dev/ttyUSB2|
>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
>> f.tty_audio_device = |/dev/ttyUSB1|
>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3058 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3058 ][none      ][-1,-1,-1]
>> ************************************************
>> 2013-08-22 23:29:53.027123 [WARNING] mod_gsmopen.cpp:1589 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][WARNINGA  1589 ][gsm01     ][-1, 0, 0]
>> STARTING interface_id=1
>> 2013-08-22 23:29:53.027123 [ERR] gsmopen_protocol.cpp:137 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
>> /dev/ttyUSB3, NOT open
>> 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:1608 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
>> STARTING interface_id=1 FAILED: gsmopen_serial_init failed
>>
>> +OK Reloading XML
>> +OK
>>
>> 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:2684 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
>> on interface gsm01:
>> freeswitch at dabbian1> 2013-08-22 23:29:53.027123 [CONSOLE]
>> switch_loadable_module.c:1401 Successfully Loaded [mod_gsmopen]
>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:147 Adding
>> Endpoint 'gsmopen'
>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>> API Function 'gsm'
>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>> API Function 'gsmopen'
>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>> API Function 'gsmopen_boost_audio'
>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>> API Function 'gsmopen_dump'
>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>> API Function 'gsmopen_sendsms'
>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:417 Adding
>> Chat interface 'sms'
>>
>> freeswitch at dabbian1>
>>
>> ---
>>
>>
>> With this understanding, I went ahead and modified the gsmopen config
>> file, to:
>>   <per_interface_settings>
>>     <interface id="1" name="gsm01">
>>       <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>       <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>
>> then unloaded mod_gsmopen, and reloaded it, but now, I get this error:
>>
>> 2013-08-22 23:41:43.587104 [ERR] gsmopen_protocol.cpp:137 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
>> /dev/ttyUSB3, NOT open
>> 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:1608 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
>> STARTING interface_id=1 FAILED: gsmopen_serial_init failed
>> 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:2684 rev
>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
>> on interface gsm01:
>>
>> What does it need /dev/ttyUSB3 for ?
>>
>> Maybe -- after all, my non linear progression into setting up FS and
>> using it by somewhat of trial-n-error isn't helping!
>>
>> thanks,
>> Jay
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Fri Aug 23 09:47:34 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Fri, 23 Aug 2013 11:17:34 +0530
Subject: [Freeswitch-users] Help understand error message -- gsmopen
	fails
In-Reply-To: <CAHDL-swo+_D_-bgwmW_jZOv-CJGkcxe7bPXbJEz-5nFAowf2Og@mail.gmail.com>
References: <CAHDL-szUF1yMzuYoK5VOe5phhpMHVmhyxRoyPM4_cHo28fXC2Q@mail.gmail.com>
	<CAMvhjJu7-jrHFzZJdMu+A8Leg_Dc2fy-HnT6jK9BdbxXOaFjbw@mail.gmail.com>
	<CAHDL-swo+_D_-bgwmW_jZOv-CJGkcxe7bPXbJEz-5nFAowf2Og@mail.gmail.com>
Message-ID: <CAMvhjJvv1dni2SbB0DmdO+MF439Su_jcb8=FXFtkDkauo0iKTA@mail.gmail.com>

Yes I am trying to use FS for IVR. You are right not many rather hardly
have i seen on this community who are using mod_gsmopen. I wrote to the
person who has contributed mod_gsmopen to FS but also no reply from his end
too. I was looking for someone who could help me with this aspect of FS and
then yesterday i came across your mail. I thought you would help me with
mod_gsmopen. May i expect some help ?

Regards
Ashish Mishra
On Aug 23, 2013 8:16 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:

> Looks like, not many people use mod_gsmopen. I've opened a JIRA ticket for
> some minor issues in the Wiki, but hasn't been acknowledged, and I see no
> one responding this mail either. I'd seen your other mails in the archive
> while searching for hints in previous exchanges.
>
> Ubuntu/Debian setups should be fairly similar. Did you follow the wiki to
> build it from scratch or using the repo packages ? I guess the former,
> because Debian repo packages didn't have mod_gsmopen.
>
> I plan to spend a day or two reading about FS. The wiki has a tonne of
> info but not in a very well organized fashion. Also, I saw that the JIRA
> bug reports have lot of useful information in them.
>
> BTW what are you trying to do with FS ? Set up an IVR ?
>
>
> On Fri, Aug 23, 2013 at 6:19 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Hi jayanth
>> Since you are the first person whom i have seen working with mod_gsmopen.
>> Can you help me with the ways to enable freeswitch use mod_gsmopen on
>> ubuntu machine.
>> This may sound ironical that i am not answering your question rather
>> asking a question as a reply of your question.
>> I will send you the error log which i come across after loading
>> mod_gsmopen on freeswitch.
>> Hope we can together help each other.
>>
>> Regards
>> Ashish Mishra
>> On Aug 23, 2013 3:12 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:
>>
>>> Latest FS (Git HEAD, as of yesterday), cloned and built successfully
>>> with mod_gsmopen. Have used a fully unlocked E1550 dongle (tested in Mobile
>>> Partner softeware in Windows, able to make calls, send SMSs). Started FS
>>> with default configurations, as per Wiki instructions. Using the default
>>> configration for gsmopen as well (maybe, that is the problem) ???
>>>
>>> I get the highlighted errors, while loading mod_gsmopen ! My naive
>>> interpretation (started with FS yesterday, and no experience with Asterisk
>>> either), is that in the gsmopen module default config, I find:
>>>
>>>   <per_interface_settings>
>>>     <interface id="1" name="gsm01">
>>>       <param name="controldevice_name" value="/dev/ttyUSB3"/>
>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>>>
>>> However from the error logs printed in FS console, I see:
>>>
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
>>> f.tty_data_device = |/dev/ttyUSB2|
>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
>>> f.tty_audio_device = |/dev/ttyUSB1|
>>>
>>> Probably that is autodetected !! But how, without the IMSI/IMEI being
>>> specified?
>>>
>>> ---
>>>
>>> freeswitch at dabbian1> load mod_gsmopen
>>> 2013-08-22 23:29:51.807103 [INFO] mod_enum.c:876 ENUM Reloaded
>>> 2013-08-22 23:29:51.847112 [INFO] switch_time.c:1191 Timezone reloaded
>>> 530 definitions
>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1106 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1106 ][none      ][-1,-1,-1] GSMOPEN
>>> Charset Output Test 0 ?????
>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1107 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1107 ][none      ][-1,-1,-1] GSMOPEN
>>> Charset Output Test 1 ???^?
>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1108 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1108 ][none      ][-1,-1,-1] GSMOPEN
>>> Charset Output Test 2 ???????
>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1109 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1109 ][none      ][-1,-1,-1] GSMOPEN
>>> Charset Output Test 3 ??????
>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1110 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1110 ][none      ][-1,-1,-1] GSMOPEN
>>> Charset Output Test 4 ???
>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1111 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1111 ][none      ][-1,-1,-1] GSMOPEN
>>> Charset Output Test 5 ??
>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1112 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1112 ][none      ][-1,-1,-1] GSMOPEN
>>> Charset Output Test 6 ??
>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3053 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3053 ][none      ][-1,-1,-1]
>>> ************************************************
>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3054 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3054 ][none      ][-1,-1,-1]
>>> f.imei=|354638041679399|
>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3055 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3055 ][none      ][-1,-1,-1]
>>> f.imsi=|405034007414619|
>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3056 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
>>> f.tty_data_device = |/dev/ttyUSB2|
>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
>>> f.tty_audio_device = |/dev/ttyUSB1|
>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3058 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3058 ][none      ][-1,-1,-1]
>>> ************************************************
>>> 2013-08-22 23:29:53.027123 [WARNING] mod_gsmopen.cpp:1589 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][WARNINGA  1589 ][gsm01     ][-1, 0, 0]
>>> STARTING interface_id=1
>>> 2013-08-22 23:29:53.027123 [ERR] gsmopen_protocol.cpp:137 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
>>> /dev/ttyUSB3, NOT open
>>> 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:1608 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
>>> STARTING interface_id=1 FAILED: gsmopen_serial_init failed
>>>
>>> +OK Reloading XML
>>> +OK
>>>
>>> 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:2684 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
>>> on interface gsm01:
>>> freeswitch at dabbian1> 2013-08-22 23:29:53.027123 [CONSOLE]
>>> switch_loadable_module.c:1401 Successfully Loaded [mod_gsmopen]
>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:147 Adding
>>> Endpoint 'gsmopen'
>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>>> API Function 'gsm'
>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>>> API Function 'gsmopen'
>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>>> API Function 'gsmopen_boost_audio'
>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>>> API Function 'gsmopen_dump'
>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>>> API Function 'gsmopen_sendsms'
>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:417 Adding
>>> Chat interface 'sms'
>>>
>>> freeswitch at dabbian1>
>>>
>>> ---
>>>
>>>
>>> With this understanding, I went ahead and modified the gsmopen config
>>> file, to:
>>>   <per_interface_settings>
>>>     <interface id="1" name="gsm01">
>>>       <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>
>>> then unloaded mod_gsmopen, and reloaded it, but now, I get this error:
>>>
>>> 2013-08-22 23:41:43.587104 [ERR] gsmopen_protocol.cpp:137 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
>>> /dev/ttyUSB3, NOT open
>>> 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:1608 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
>>> STARTING interface_id=1 FAILED: gsmopen_serial_init failed
>>> 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:2684 rev
>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
>>> on interface gsm01:
>>>
>>> What does it need /dev/ttyUSB3 for ?
>>>
>>> Maybe -- after all, my non linear progression into setting up FS and
>>> using it by somewhat of trial-n-error isn't helping!
>>>
>>> thanks,
>>> Jay
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Fri Aug 23 09:55:16 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Fri, 23 Aug 2013 11:25:16 +0530
Subject: [Freeswitch-users] Help understand error message -- gsmopen
	fails
In-Reply-To: <CAMvhjJvv1dni2SbB0DmdO+MF439Su_jcb8=FXFtkDkauo0iKTA@mail.gmail.com>
References: <CAHDL-szUF1yMzuYoK5VOe5phhpMHVmhyxRoyPM4_cHo28fXC2Q@mail.gmail.com>
	<CAMvhjJu7-jrHFzZJdMu+A8Leg_Dc2fy-HnT6jK9BdbxXOaFjbw@mail.gmail.com>
	<CAHDL-swo+_D_-bgwmW_jZOv-CJGkcxe7bPXbJEz-5nFAowf2Og@mail.gmail.com>
	<CAMvhjJvv1dni2SbB0DmdO+MF439Su_jcb8=FXFtkDkauo0iKTA@mail.gmail.com>
Message-ID: <CAHDL-syoG-k94NZ2VEYeQiJmFgfLU9jVEHGFTWtqRo6adm=U+g@mail.gmail.com>

No idea if I'd be able to help. If you see my post, you'd notice that I've
not managed to get it working either :-) ... my only success so far has
been in building the module.

Post your error logs in pastebin... and if your terminal settings are
correct, you'd notice that some of the error messages are in red. In a mail
just share that part, and rest of the full log in pastebin as suggested.
Let me see if I can figure out anything... though don't be too hopeful.


On Fri, Aug 23, 2013 at 11:17 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Yes I am trying to use FS for IVR. You are right not many rather hardly
> have i seen on this community who are using mod_gsmopen. I wrote to the
> person who has contributed mod_gsmopen to FS but also no reply from his end
> too. I was looking for someone who could help me with this aspect of FS and
> then yesterday i came across your mail. I thought you would help me with
> mod_gsmopen. May i expect some help ?
>
> Regards
> Ashish Mishra
> On Aug 23, 2013 8:16 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:
>
>> Looks like, not many people use mod_gsmopen. I've opened a JIRA ticket
>> for some minor issues in the Wiki, but hasn't been acknowledged, and I see
>> no one responding this mail either. I'd seen your other mails in the
>> archive while searching for hints in previous exchanges.
>>
>> Ubuntu/Debian setups should be fairly similar. Did you follow the wiki to
>> build it from scratch or using the repo packages ? I guess the former,
>> because Debian repo packages didn't have mod_gsmopen.
>>
>> I plan to spend a day or two reading about FS. The wiki has a tonne of
>> info but not in a very well organized fashion. Also, I saw that the JIRA
>> bug reports have lot of useful information in them.
>>
>> BTW what are you trying to do with FS ? Set up an IVR ?
>>
>>
>> On Fri, Aug 23, 2013 at 6:19 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> Hi jayanth
>>> Since you are the first person whom i have seen working with
>>> mod_gsmopen. Can you help me with the ways to enable freeswitch use
>>> mod_gsmopen on ubuntu machine.
>>> This may sound ironical that i am not answering your question rather
>>> asking a question as a reply of your question.
>>> I will send you the error log which i come across after loading
>>> mod_gsmopen on freeswitch.
>>> Hope we can together help each other.
>>>
>>> Regards
>>> Ashish Mishra
>>> On Aug 23, 2013 3:12 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:
>>>
>>>> Latest FS (Git HEAD, as of yesterday), cloned and built successfully
>>>> with mod_gsmopen. Have used a fully unlocked E1550 dongle (tested in Mobile
>>>> Partner softeware in Windows, able to make calls, send SMSs). Started FS
>>>> with default configurations, as per Wiki instructions. Using the default
>>>> configration for gsmopen as well (maybe, that is the problem) ???
>>>>
>>>> I get the highlighted errors, while loading mod_gsmopen ! My naive
>>>> interpretation (started with FS yesterday, and no experience with Asterisk
>>>> either), is that in the gsmopen module default config, I find:
>>>>
>>>>   <per_interface_settings>
>>>>     <interface id="1" name="gsm01">
>>>>       <param name="controldevice_name" value="/dev/ttyUSB3"/>
>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>>>>
>>>> However from the error logs printed in FS console, I see:
>>>>
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
>>>> f.tty_data_device = |/dev/ttyUSB2|
>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
>>>> f.tty_audio_device = |/dev/ttyUSB1|
>>>>
>>>> Probably that is autodetected !! But how, without the IMSI/IMEI being
>>>> specified?
>>>>
>>>> ---
>>>>
>>>> freeswitch at dabbian1> load mod_gsmopen
>>>> 2013-08-22 23:29:51.807103 [INFO] mod_enum.c:876 ENUM Reloaded
>>>> 2013-08-22 23:29:51.847112 [INFO] switch_time.c:1191 Timezone reloaded
>>>> 530 definitions
>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1106 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1106 ][none      ][-1,-1,-1] GSMOPEN
>>>> Charset Output Test 0 ?????
>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1107 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1107 ][none      ][-1,-1,-1] GSMOPEN
>>>> Charset Output Test 1 ???^?
>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1108 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1108 ][none      ][-1,-1,-1] GSMOPEN
>>>> Charset Output Test 2 ???????
>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1109 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1109 ][none      ][-1,-1,-1] GSMOPEN
>>>> Charset Output Test 3 ??????
>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1110 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1110 ][none      ][-1,-1,-1] GSMOPEN
>>>> Charset Output Test 4 ???
>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1111 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1111 ][none      ][-1,-1,-1] GSMOPEN
>>>> Charset Output Test 5 ??
>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1112 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1112 ][none      ][-1,-1,-1] GSMOPEN
>>>> Charset Output Test 6 ??
>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3053 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3053 ][none      ][-1,-1,-1]
>>>> ************************************************
>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3054 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3054 ][none      ][-1,-1,-1]
>>>> f.imei=|354638041679399|
>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3055 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3055 ][none      ][-1,-1,-1]
>>>> f.imsi=|405034007414619|
>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3056 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
>>>> f.tty_data_device = |/dev/ttyUSB2|
>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
>>>> f.tty_audio_device = |/dev/ttyUSB1|
>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3058 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3058 ][none      ][-1,-1,-1]
>>>> ************************************************
>>>> 2013-08-22 23:29:53.027123 [WARNING] mod_gsmopen.cpp:1589 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][WARNINGA  1589 ][gsm01     ][-1, 0, 0]
>>>> STARTING interface_id=1
>>>> 2013-08-22 23:29:53.027123 [ERR] gsmopen_protocol.cpp:137 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
>>>> /dev/ttyUSB3, NOT open
>>>> 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:1608 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
>>>> STARTING interface_id=1 FAILED: gsmopen_serial_init failed
>>>>
>>>> +OK Reloading XML
>>>> +OK
>>>>
>>>> 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:2684 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
>>>> on interface gsm01:
>>>> freeswitch at dabbian1> 2013-08-22 23:29:53.027123 [CONSOLE]
>>>> switch_loadable_module.c:1401 Successfully Loaded [mod_gsmopen]
>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:147 Adding
>>>> Endpoint 'gsmopen'
>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>>>> API Function 'gsm'
>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>>>> API Function 'gsmopen'
>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>>>> API Function 'gsmopen_boost_audio'
>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>>>> API Function 'gsmopen_dump'
>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305 Adding
>>>> API Function 'gsmopen_sendsms'
>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:417 Adding
>>>> Chat interface 'sms'
>>>>
>>>> freeswitch at dabbian1>
>>>>
>>>> ---
>>>>
>>>>
>>>> With this understanding, I went ahead and modified the gsmopen config
>>>> file, to:
>>>>   <per_interface_settings>
>>>>     <interface id="1" name="gsm01">
>>>>       <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>
>>>> then unloaded mod_gsmopen, and reloaded it, but now, I get this error:
>>>>
>>>> 2013-08-22 23:41:43.587104 [ERR] gsmopen_protocol.cpp:137 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
>>>> /dev/ttyUSB3, NOT open
>>>> 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:1608 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
>>>> STARTING interface_id=1 FAILED: gsmopen_serial_init failed
>>>> 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:2684 rev
>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
>>>> on interface gsm01:
>>>>
>>>> What does it need /dev/ttyUSB3 for ?
>>>>
>>>> Maybe -- after all, my non linear progression into setting up FS and
>>>> using it by somewhat of trial-n-error isn't helping!
>>>>
>>>> thanks,
>>>> Jay
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nandy1925 at gmail.com  Fri Aug 23 10:38:40 2013
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 23 Aug 2013 14:38:40 +0800
Subject: [Freeswitch-users] Help understand error message -- gsmopen
	fails
In-Reply-To: <CAHDL-syoG-k94NZ2VEYeQiJmFgfLU9jVEHGFTWtqRo6adm=U+g@mail.gmail.com>
References: <CAHDL-szUF1yMzuYoK5VOe5phhpMHVmhyxRoyPM4_cHo28fXC2Q@mail.gmail.com>
	<CAMvhjJu7-jrHFzZJdMu+A8Leg_Dc2fy-HnT6jK9BdbxXOaFjbw@mail.gmail.com>
	<CAHDL-swo+_D_-bgwmW_jZOv-CJGkcxe7bPXbJEz-5nFAowf2Og@mail.gmail.com>
	<CAMvhjJvv1dni2SbB0DmdO+MF439Su_jcb8=FXFtkDkauo0iKTA@mail.gmail.com>
	<CAHDL-syoG-k94NZ2VEYeQiJmFgfLU9jVEHGFTWtqRo6adm=U+g@mail.gmail.com>
Message-ID: <CAHKqMb45g1WvKXRkfcs0K-pboi06LK_5WsxtxkQwW0QX8gFUkg@mail.gmail.com>

Did you check the devices if they're really correct? I mean:

# ls -l /dev/ttyUSB*

I noticed that there were 3 devices open in 1 dongle - ttyUSB1, ttyUSB2 and
ttyUSB3.  The IMSI/IMEI parameters were added to fix the device assignments
if you have 2 or more USB dongles.

/Nandy
---
No idea if I'd be able to help. If you see my post, you'd notice that I've
not managed to get it working either :-) ... my only success so far has
been in building the module.

>
> Post your error logs in pastebin... and if your terminal settings are
> correct, you'd notice that some of the error messages are in red. In a mail
> just share that part, and rest of the full log in pastebin as suggested.
> Let me see if I can figure out anything... though don't be too hopeful.
>
>
> On Fri, Aug 23, 2013 at 11:17 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Yes I am trying to use FS for IVR. You are right not many rather hardly
>> have i seen on this community who are using mod_gsmopen. I wrote to the
>> person who has contributed mod_gsmopen to FS but also no reply from his end
>> too. I was looking for someone who could help me with this aspect of FS and
>> then yesterday i came across your mail. I thought you would help me with
>> mod_gsmopen. May i expect some help ?
>>
>> Regards
>> Ashish Mishra
>> On Aug 23, 2013 8:16 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:
>>
>>> Looks like, not many people use mod_gsmopen. I've opened a JIRA ticket
>>> for some minor issues in the Wiki, but hasn't been acknowledged, and I see
>>> no one responding this mail either. I'd seen your other mails in the
>>> archive while searching for hints in previous exchanges.
>>>
>>> Ubuntu/Debian setups should be fairly similar. Did you follow the wiki
>>> to build it from scratch or using the repo packages ? I guess the former,
>>> because Debian repo packages didn't have mod_gsmopen.
>>>
>>> I plan to spend a day or two reading about FS. The wiki has a tonne of
>>> info but not in a very well organized fashion. Also, I saw that the JIRA
>>> bug reports have lot of useful information in them.
>>>
>>> BTW what are you trying to do with FS ? Set up an IVR ?
>>>
>>>
>>> On Fri, Aug 23, 2013 at 6:19 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> Hi jayanth
>>>> Since you are the first person whom i have seen working with
>>>> mod_gsmopen. Can you help me with the ways to enable freeswitch use
>>>> mod_gsmopen on ubuntu machine.
>>>> This may sound ironical that i am not answering your question rather
>>>> asking a question as a reply of your question.
>>>> I will send you the error log which i come across after loading
>>>> mod_gsmopen on freeswitch.
>>>> Hope we can together help each other.
>>>>
>>>> Regards
>>>> Ashish Mishra
>>>> On Aug 23, 2013 3:12 AM, "Jayanth Acharya" <jayachar88 at gmail.com>
>>>> wrote:
>>>>
>>>>> Latest FS (Git HEAD, as of yesterday), cloned and built successfully
>>>>> with mod_gsmopen. Have used a fully unlocked E1550 dongle (tested in Mobile
>>>>> Partner softeware in Windows, able to make calls, send SMSs). Started FS
>>>>> with default configurations, as per Wiki instructions. Using the default
>>>>> configration for gsmopen as well (maybe, that is the problem) ???
>>>>>
>>>>> I get the highlighted errors, while loading mod_gsmopen ! My naive
>>>>> interpretation (started with FS yesterday, and no experience with Asterisk
>>>>> either), is that in the gsmopen module default config, I find:
>>>>>
>>>>>   <per_interface_settings>
>>>>>     <interface id="1" name="gsm01">
>>>>>       <param name="controldevice_name" value="/dev/ttyUSB3"/>
>>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>>>>>
>>>>> However from the error logs printed in FS console, I see:
>>>>>
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
>>>>> f.tty_data_device = |/dev/ttyUSB2|
>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
>>>>> f.tty_audio_device = |/dev/ttyUSB1|
>>>>>
>>>>> Probably that is autodetected !! But how, without the IMSI/IMEI being
>>>>> specified?
>>>>>
>>>>> ---
>>>>>
>>>>> freeswitch at dabbian1> load mod_gsmopen
>>>>> 2013-08-22 23:29:51.807103 [INFO] mod_enum.c:876 ENUM Reloaded
>>>>> 2013-08-22 23:29:51.847112 [INFO] switch_time.c:1191 Timezone reloaded
>>>>> 530 definitions
>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1106 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1106 ][none      ][-1,-1,-1] GSMOPEN
>>>>> Charset Output Test 0 ?????
>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1107 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1107 ][none      ][-1,-1,-1] GSMOPEN
>>>>> Charset Output Test 1 ???^?
>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1108 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1108 ][none      ][-1,-1,-1] GSMOPEN
>>>>> Charset Output Test 2 ???????
>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1109 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1109 ][none      ][-1,-1,-1] GSMOPEN
>>>>> Charset Output Test 3 ??????
>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1110 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1110 ][none      ][-1,-1,-1] GSMOPEN
>>>>> Charset Output Test 4 ???
>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1111 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1111 ][none      ][-1,-1,-1] GSMOPEN
>>>>> Charset Output Test 5 ??
>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1112 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1112 ][none      ][-1,-1,-1] GSMOPEN
>>>>> Charset Output Test 6 ??
>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3053 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3053 ][none      ][-1,-1,-1]
>>>>> ************************************************
>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3054 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3054 ][none      ][-1,-1,-1]
>>>>> f.imei=|354638041679399|
>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3055 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3055 ][none      ][-1,-1,-1]
>>>>> f.imsi=|405034007414619|
>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3056 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
>>>>> f.tty_data_device = |/dev/ttyUSB2|
>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
>>>>> f.tty_audio_device = |/dev/ttyUSB1|
>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3058 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3058 ][none      ][-1,-1,-1]
>>>>> ************************************************
>>>>> 2013-08-22 23:29:53.027123 [WARNING] mod_gsmopen.cpp:1589 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][WARNINGA  1589 ][gsm01     ][-1, 0, 0]
>>>>> STARTING interface_id=1
>>>>> 2013-08-22 23:29:53.027123 [ERR] gsmopen_protocol.cpp:137 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
>>>>> /dev/ttyUSB3, NOT open
>>>>> 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:1608 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
>>>>> STARTING interface_id=1 FAILED: gsmopen_serial_init failed
>>>>>
>>>>> +OK Reloading XML
>>>>> +OK
>>>>>
>>>>> 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:2684 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
>>>>> on interface gsm01:
>>>>> freeswitch at dabbian1> 2013-08-22 23:29:53.027123 [CONSOLE]
>>>>> switch_loadable_module.c:1401 Successfully Loaded [mod_gsmopen]
>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:147
>>>>> Adding Endpoint 'gsmopen'
>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>> Adding API Function 'gsm'
>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>> Adding API Function 'gsmopen'
>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>> Adding API Function 'gsmopen_boost_audio'
>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>> Adding API Function 'gsmopen_dump'
>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>> Adding API Function 'gsmopen_sendsms'
>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:417
>>>>> Adding Chat interface 'sms'
>>>>>
>>>>> freeswitch at dabbian1>
>>>>>
>>>>> ---
>>>>>
>>>>>
>>>>> With this understanding, I went ahead and modified the gsmopen config
>>>>> file, to:
>>>>>   <per_interface_settings>
>>>>>     <interface id="1" name="gsm01">
>>>>>       <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>>
>>>>> then unloaded mod_gsmopen, and reloaded it, but now, I get this error:
>>>>>
>>>>> 2013-08-22 23:41:43.587104 [ERR] gsmopen_protocol.cpp:137 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
>>>>> /dev/ttyUSB3, NOT open
>>>>> 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:1608 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
>>>>> STARTING interface_id=1 FAILED: gsmopen_serial_init failed
>>>>> 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:2684 rev
>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
>>>>> on interface gsm01:
>>>>>
>>>>> What does it need /dev/ttyUSB3 for ?
>>>>>
>>>>> Maybe -- after all, my non linear progression into setting up FS and
>>>>> using it by somewhat of trial-n-error isn't helping!
>>>>>
>>>>> thanks,
>>>>> Jay
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Fri Aug 23 10:50:25 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Fri, 23 Aug 2013 12:20:25 +0530
Subject: [Freeswitch-users] Help understand error message -- gsmopen
	fails
In-Reply-To: <CAHKqMb45g1WvKXRkfcs0K-pboi06LK_5WsxtxkQwW0QX8gFUkg@mail.gmail.com>
References: <CAHDL-szUF1yMzuYoK5VOe5phhpMHVmhyxRoyPM4_cHo28fXC2Q@mail.gmail.com>
	<CAMvhjJu7-jrHFzZJdMu+A8Leg_Dc2fy-HnT6jK9BdbxXOaFjbw@mail.gmail.com>
	<CAHDL-swo+_D_-bgwmW_jZOv-CJGkcxe7bPXbJEz-5nFAowf2Og@mail.gmail.com>
	<CAMvhjJvv1dni2SbB0DmdO+MF439Su_jcb8=FXFtkDkauo0iKTA@mail.gmail.com>
	<CAHDL-syoG-k94NZ2VEYeQiJmFgfLU9jVEHGFTWtqRo6adm=U+g@mail.gmail.com>
	<CAHKqMb45g1WvKXRkfcs0K-pboi06LK_5WsxtxkQwW0QX8gFUkg@mail.gmail.com>
Message-ID: <CAHDL-swp99USsJBj45JWKJJTZZpDzTCVU=8buUOrjky9eH=jPg@mail.gmail.com>

Hi Nandy,

Doing *ls -l* on /dev/ttyUSB* did show me 3 devices, ttyUSB0, ttyUSB1 and
ttyUSB2 (i.e. starting with no. 0, not 1 as you cite, and also cited by the
Wiki).

In the gsmopen conf file, I didn't have IMEI/IMSI but from the logs, it
appears that FS did detect voice and control/data devices. BTW, are the
control and data device same ? Or are voice, data, control 3 different
device types ?

cheers,
Jay


On Fri, Aug 23, 2013 at 12:08 PM, Nandy Dagondon <nandy1925 at gmail.com>wrote:

> Did you check the devices if they're really correct? I mean:
>
> # ls -l /dev/ttyUSB*
>
> I noticed that there were 3 devices open in 1 dongle - ttyUSB1, ttyUSB2
> and ttyUSB3.  The IMSI/IMEI parameters were added to fix the device
> assignments if you have 2 or more USB dongles.
>
> /Nandy
> ---
>
> No idea if I'd be able to help. If you see my post, you'd notice that I've
> not managed to get it working either :-) ... my only success so far has
> been in building the module.
>
>>
>> Post your error logs in pastebin... and if your terminal settings are
>> correct, you'd notice that some of the error messages are in red. In a mail
>> just share that part, and rest of the full log in pastebin as suggested.
>> Let me see if I can figure out anything... though don't be too hopeful.
>>
>>
>> On Fri, Aug 23, 2013 at 11:17 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> Yes I am trying to use FS for IVR. You are right not many rather hardly
>>> have i seen on this community who are using mod_gsmopen. I wrote to the
>>> person who has contributed mod_gsmopen to FS but also no reply from his end
>>> too. I was looking for someone who could help me with this aspect of FS and
>>> then yesterday i came across your mail. I thought you would help me with
>>> mod_gsmopen. May i expect some help ?
>>>
>>> Regards
>>> Ashish Mishra
>>> On Aug 23, 2013 8:16 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:
>>>
>>>> Looks like, not many people use mod_gsmopen. I've opened a JIRA ticket
>>>> for some minor issues in the Wiki, but hasn't been acknowledged, and I see
>>>> no one responding this mail either. I'd seen your other mails in the
>>>> archive while searching for hints in previous exchanges.
>>>>
>>>> Ubuntu/Debian setups should be fairly similar. Did you follow the wiki
>>>> to build it from scratch or using the repo packages ? I guess the former,
>>>> because Debian repo packages didn't have mod_gsmopen.
>>>>
>>>> I plan to spend a day or two reading about FS. The wiki has a tonne of
>>>> info but not in a very well organized fashion. Also, I saw that the JIRA
>>>> bug reports have lot of useful information in them.
>>>>
>>>> BTW what are you trying to do with FS ? Set up an IVR ?
>>>>
>>>>
>>>> On Fri, Aug 23, 2013 at 6:19 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>
>>>>> Hi jayanth
>>>>> Since you are the first person whom i have seen working with
>>>>> mod_gsmopen. Can you help me with the ways to enable freeswitch use
>>>>> mod_gsmopen on ubuntu machine.
>>>>> This may sound ironical that i am not answering your question rather
>>>>> asking a question as a reply of your question.
>>>>> I will send you the error log which i come across after loading
>>>>> mod_gsmopen on freeswitch.
>>>>> Hope we can together help each other.
>>>>>
>>>>> Regards
>>>>> Ashish Mishra
>>>>> On Aug 23, 2013 3:12 AM, "Jayanth Acharya" <jayachar88 at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Latest FS (Git HEAD, as of yesterday), cloned and built successfully
>>>>>> with mod_gsmopen. Have used a fully unlocked E1550 dongle (tested in Mobile
>>>>>> Partner softeware in Windows, able to make calls, send SMSs). Started FS
>>>>>> with default configurations, as per Wiki instructions. Using the default
>>>>>> configration for gsmopen as well (maybe, that is the problem) ???
>>>>>>
>>>>>> I get the highlighted errors, while loading mod_gsmopen ! My naive
>>>>>> interpretation (started with FS yesterday, and no experience with Asterisk
>>>>>> either), is that in the gsmopen module default config, I find:
>>>>>>
>>>>>>   <per_interface_settings>
>>>>>>     <interface id="1" name="gsm01">
>>>>>>       <param name="controldevice_name" value="/dev/ttyUSB3"/>
>>>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>>>>>>
>>>>>> However from the error logs printed in FS console, I see:
>>>>>>
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
>>>>>> f.tty_data_device = |/dev/ttyUSB2|
>>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
>>>>>> f.tty_audio_device = |/dev/ttyUSB1|
>>>>>>
>>>>>> Probably that is autodetected !! But how, without the IMSI/IMEI being
>>>>>> specified?
>>>>>>
>>>>>> ---
>>>>>>
>>>>>> freeswitch at dabbian1> load mod_gsmopen
>>>>>> 2013-08-22 23:29:51.807103 [INFO] mod_enum.c:876 ENUM Reloaded
>>>>>> 2013-08-22 23:29:51.847112 [INFO] switch_time.c:1191 Timezone
>>>>>> reloaded 530 definitions
>>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1106 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1106 ][none      ][-1,-1,-1] GSMOPEN
>>>>>> Charset Output Test 0 ?????
>>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1107 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1107 ][none      ][-1,-1,-1] GSMOPEN
>>>>>> Charset Output Test 1 ???^?
>>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1108 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1108 ][none      ][-1,-1,-1] GSMOPEN
>>>>>> Charset Output Test 2 ???????
>>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1109 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1109 ][none      ][-1,-1,-1] GSMOPEN
>>>>>> Charset Output Test 3 ??????
>>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1110 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1110 ][none      ][-1,-1,-1] GSMOPEN
>>>>>> Charset Output Test 4 ???
>>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1111 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1111 ][none      ][-1,-1,-1] GSMOPEN
>>>>>> Charset Output Test 5 ??
>>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1112 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1112 ][none      ][-1,-1,-1] GSMOPEN
>>>>>> Charset Output Test 6 ??
>>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3053 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3053 ][none      ][-1,-1,-1]
>>>>>> ************************************************
>>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3054 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3054 ][none      ][-1,-1,-1]
>>>>>> f.imei=|354638041679399|
>>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3055 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3055 ][none      ][-1,-1,-1]
>>>>>> f.imsi=|405034007414619|
>>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3056 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
>>>>>> f.tty_data_device = |/dev/ttyUSB2|
>>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
>>>>>> f.tty_audio_device = |/dev/ttyUSB1|
>>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3058 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3058 ][none      ][-1,-1,-1]
>>>>>> ************************************************
>>>>>> 2013-08-22 23:29:53.027123 [WARNING] mod_gsmopen.cpp:1589 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][WARNINGA  1589 ][gsm01     ][-1, 0, 0]
>>>>>> STARTING interface_id=1
>>>>>> 2013-08-22 23:29:53.027123 [ERR] gsmopen_protocol.cpp:137 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
>>>>>> /dev/ttyUSB3, NOT open
>>>>>> 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:1608 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
>>>>>> STARTING interface_id=1 FAILED: gsmopen_serial_init failed
>>>>>>
>>>>>> +OK Reloading XML
>>>>>> +OK
>>>>>>
>>>>>> 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:2684 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
>>>>>> on interface gsm01:
>>>>>> freeswitch at dabbian1> 2013-08-22 23:29:53.027123 [CONSOLE]
>>>>>> switch_loadable_module.c:1401 Successfully Loaded [mod_gsmopen]
>>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:147
>>>>>> Adding Endpoint 'gsmopen'
>>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>>> Adding API Function 'gsm'
>>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>>> Adding API Function 'gsmopen'
>>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>>> Adding API Function 'gsmopen_boost_audio'
>>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>>> Adding API Function 'gsmopen_dump'
>>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>>> Adding API Function 'gsmopen_sendsms'
>>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:417
>>>>>> Adding Chat interface 'sms'
>>>>>>
>>>>>> freeswitch at dabbian1>
>>>>>>
>>>>>> ---
>>>>>>
>>>>>>
>>>>>> With this understanding, I went ahead and modified the gsmopen config
>>>>>> file, to:
>>>>>>   <per_interface_settings>
>>>>>>     <interface id="1" name="gsm01">
>>>>>>       <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>>>
>>>>>> then unloaded mod_gsmopen, and reloaded it, but now, I get this error:
>>>>>>
>>>>>> 2013-08-22 23:41:43.587104 [ERR] gsmopen_protocol.cpp:137 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
>>>>>> /dev/ttyUSB3, NOT open
>>>>>> 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:1608 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
>>>>>> STARTING interface_id=1 FAILED: gsmopen_serial_init failed
>>>>>> 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:2684 rev
>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
>>>>>> on interface gsm01:
>>>>>>
>>>>>> What does it need /dev/ttyUSB3 for ?
>>>>>>
>>>>>> Maybe -- after all, my non linear progression into setting up FS and
>>>>>> using it by somewhat of trial-n-error isn't helping!
>>>>>>
>>>>>> thanks,
>>>>>> Jay
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 GB at cm.nl  Fri Aug 23 11:15:32 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Fri, 23 Aug 2013 09:15:32 +0200
Subject: [Freeswitch-users] Tuning DTMF
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD19CD6@clubvirtual40.ClubMessage.local>

Hello,

Setup is as following:

X-Lite (Test phone) ------>  SIP Proxy ------> FS (as SBC) ------> Carrier ------> End-user phone

Whenever I place a call using X-Lite to the End-user phone, in this case my own mobile phone or office phone, and start sending DTMF tones from the remote telephone, the DMTF tones arrive with a crackling sound at the end of the tone.

Console debug:
2013-08-23 09:00:13.813207 [DEBUG] switch_rtp.c:3829 Send start packet for [2] ts=3778476449 dur=160/160/1280 seq=5098 lw=-516490847
2013-08-23 09:00:13.833208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=320/320/1280 seq=5099 lw=161
2013-08-23 09:00:13.853208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=480/480/1280 seq=5100 lw=161
2013-08-23 09:00:13.873208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=640/640/1280 seq=5101 lw=161
2013-08-23 09:00:13.893208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=800/800/1280 seq=5102 lw=161
2013-08-23 09:00:13.913208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=960/960/1280 seq=5103 lw=161
2013-08-23 09:00:13.933208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=1120/1120/1280 seq=5104 lw=161
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5105 lw=1
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5106 lw=1
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5107 lw=1
2013-08-23 09:00:14.813206 [DEBUG] switch_rtp.c:5117 RTP RECV DTMF 1:1280
2013-08-23 09:00:14.813206 [DEBUG] switch_channel.c:471 RECV DTMF 1:1280

Both internal and external sip profiles have the following configuration for DTMF:
<param name="liberal-dtmf" value="true"/>
<param name="rfc2833-pt" value="101"/>
<param name="dtmf-duration" value="2000"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="inbound-late-negotiation" value="true"/>

Internal.xml (Dialplan)
    <extension name="internal_to_external">
      <condition field="${sip_req_host}" expression="^10\.0\.0\.1$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="sofia/external/${sip_req_uri}"/>
      </condition>
    </extension>

Is there a way to tune the DTMF tones? If so, which parameters do I need to add or edit?

Thanks!

Grant

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From nandy1925 at gmail.com  Fri Aug 23 11:48:16 2013
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Fri, 23 Aug 2013 15:48:16 +0800
Subject: [Freeswitch-users] Help understand error message -- gsmopen
	fails
In-Reply-To: <CAHDL-swp99USsJBj45JWKJJTZZpDzTCVU=8buUOrjky9eH=jPg@mail.gmail.com>
References: <CAHDL-szUF1yMzuYoK5VOe5phhpMHVmhyxRoyPM4_cHo28fXC2Q@mail.gmail.com>
	<CAMvhjJu7-jrHFzZJdMu+A8Leg_Dc2fy-HnT6jK9BdbxXOaFjbw@mail.gmail.com>
	<CAHDL-swo+_D_-bgwmW_jZOv-CJGkcxe7bPXbJEz-5nFAowf2Og@mail.gmail.com>
	<CAMvhjJvv1dni2SbB0DmdO+MF439Su_jcb8=FXFtkDkauo0iKTA@mail.gmail.com>
	<CAHDL-syoG-k94NZ2VEYeQiJmFgfLU9jVEHGFTWtqRo6adm=U+g@mail.gmail.com>
	<CAHKqMb45g1WvKXRkfcs0K-pboi06LK_5WsxtxkQwW0QX8gFUkg@mail.gmail.com>
	<CAHDL-swp99USsJBj45JWKJJTZZpDzTCVU=8buUOrjky9eH=jPg@mail.gmail.com>
Message-ID: <CAHKqMb5nQG5a3ujgb2v+J3tZv1Yc+bVZhOjQ-d=PB03QY5j7mw@mail.gmail.com>

Okay. I saw ttyUSB3 whenever the dongle is re-inserted. In your last
question, I'm not very familiar with the exact functions. Please help!



On Fri, Aug 23, 2013 at 2:50 PM, Jayanth Acharya <jayachar88 at gmail.com>wrote:

> Hi Nandy,
>
> Doing *ls -l* on /dev/ttyUSB* did show me 3 devices, ttyUSB0, ttyUSB1 and
> ttyUSB2 (i.e. starting with no. 0, not 1 as you cite, and also cited by the
> Wiki).
>
> In the gsmopen conf file, I didn't have IMEI/IMSI but from the logs, it
> appears that FS did detect voice and control/data devices. BTW, are the
> control and data device same ? Or are voice, data, control 3 different
> device types ?
>
> cheers,
> Jay
>
>
> On Fri, Aug 23, 2013 at 12:08 PM, Nandy Dagondon <nandy1925 at gmail.com>wrote:
>
>> Did you check the devices if they're really correct? I mean:
>>
>> # ls -l /dev/ttyUSB*
>>
>> I noticed that there were 3 devices open in 1 dongle - ttyUSB1, ttyUSB2
>> and ttyUSB3.  The IMSI/IMEI parameters were added to fix the device
>> assignments if you have 2 or more USB dongles.
>>
>> /Nandy
>> ---
>>
>> No idea if I'd be able to help. If you see my post, you'd notice that
>> I've not managed to get it working either :-) ... my only success so far
>> has been in building the module.
>>
>>>
>>> Post your error logs in pastebin... and if your terminal settings are
>>> correct, you'd notice that some of the error messages are in red. In a mail
>>> just share that part, and rest of the full log in pastebin as suggested.
>>> Let me see if I can figure out anything... though don't be too hopeful.
>>>
>>>
>>> On Fri, Aug 23, 2013 at 11:17 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> Yes I am trying to use FS for IVR. You are right not many rather hardly
>>>> have i seen on this community who are using mod_gsmopen. I wrote to the
>>>> person who has contributed mod_gsmopen to FS but also no reply from his end
>>>> too. I was looking for someone who could help me with this aspect of FS and
>>>> then yesterday i came across your mail. I thought you would help me with
>>>> mod_gsmopen. May i expect some help ?
>>>>
>>>> Regards
>>>> Ashish Mishra
>>>> On Aug 23, 2013 8:16 AM, "Jayanth Acharya" <jayachar88 at gmail.com>
>>>> wrote:
>>>>
>>>>> Looks like, not many people use mod_gsmopen. I've opened a JIRA ticket
>>>>> for some minor issues in the Wiki, but hasn't been acknowledged, and I see
>>>>> no one responding this mail either. I'd seen your other mails in the
>>>>> archive while searching for hints in previous exchanges.
>>>>>
>>>>> Ubuntu/Debian setups should be fairly similar. Did you follow the wiki
>>>>> to build it from scratch or using the repo packages ? I guess the former,
>>>>> because Debian repo packages didn't have mod_gsmopen.
>>>>>
>>>>> I plan to spend a day or two reading about FS. The wiki has a tonne of
>>>>> info but not in a very well organized fashion. Also, I saw that the JIRA
>>>>> bug reports have lot of useful information in them.
>>>>>
>>>>> BTW what are you trying to do with FS ? Set up an IVR ?
>>>>>
>>>>>
>>>>> On Fri, Aug 23, 2013 at 6:19 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>>
>>>>>> Hi jayanth
>>>>>> Since you are the first person whom i have seen working with
>>>>>> mod_gsmopen. Can you help me with the ways to enable freeswitch use
>>>>>> mod_gsmopen on ubuntu machine.
>>>>>> This may sound ironical that i am not answering your question rather
>>>>>> asking a question as a reply of your question.
>>>>>> I will send you the error log which i come across after loading
>>>>>> mod_gsmopen on freeswitch.
>>>>>> Hope we can together help each other.
>>>>>>
>>>>>> Regards
>>>>>> Ashish Mishra
>>>>>> On Aug 23, 2013 3:12 AM, "Jayanth Acharya" <jayachar88 at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Latest FS (Git HEAD, as of yesterday), cloned and built successfully
>>>>>>> with mod_gsmopen. Have used a fully unlocked E1550 dongle (tested in Mobile
>>>>>>> Partner softeware in Windows, able to make calls, send SMSs). Started FS
>>>>>>> with default configurations, as per Wiki instructions. Using the default
>>>>>>> configration for gsmopen as well (maybe, that is the problem) ???
>>>>>>>
>>>>>>> I get the highlighted errors, while loading mod_gsmopen ! My naive
>>>>>>> interpretation (started with FS yesterday, and no experience with Asterisk
>>>>>>> either), is that in the gsmopen module default config, I find:
>>>>>>>
>>>>>>>   <per_interface_settings>
>>>>>>>     <interface id="1" name="gsm01">
>>>>>>>       <param name="controldevice_name" value="/dev/ttyUSB3"/>
>>>>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB2"/>
>>>>>>>
>>>>>>> However from the error logs printed in FS console, I see:
>>>>>>>
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
>>>>>>> f.tty_data_device = |/dev/ttyUSB2|
>>>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
>>>>>>> f.tty_audio_device = |/dev/ttyUSB1|
>>>>>>>
>>>>>>> Probably that is autodetected !! But how, without the IMSI/IMEI
>>>>>>> being specified?
>>>>>>>
>>>>>>> ---
>>>>>>>
>>>>>>> freeswitch at dabbian1> load mod_gsmopen
>>>>>>> 2013-08-22 23:29:51.807103 [INFO] mod_enum.c:876 ENUM Reloaded
>>>>>>> 2013-08-22 23:29:51.847112 [INFO] switch_time.c:1191 Timezone
>>>>>>> reloaded 530 definitions
>>>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1106 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1106 ][none      ][-1,-1,-1] GSMOPEN
>>>>>>> Charset Output Test 0 ?????
>>>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1107 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1107 ][none      ][-1,-1,-1] GSMOPEN
>>>>>>> Charset Output Test 1 ???^?
>>>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1108 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1108 ][none      ][-1,-1,-1] GSMOPEN
>>>>>>> Charset Output Test 2 ???????
>>>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1109 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1109 ][none      ][-1,-1,-1] GSMOPEN
>>>>>>> Charset Output Test 3 ??????
>>>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1110 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1110 ][none      ][-1,-1,-1] GSMOPEN
>>>>>>> Charset Output Test 4 ???
>>>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1111 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1111 ][none      ][-1,-1,-1] GSMOPEN
>>>>>>> Charset Output Test 5 ??
>>>>>>> 2013-08-22 23:29:52.427101 [NOTICE] mod_gsmopen.cpp:1112 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  1112 ][none      ][-1,-1,-1] GSMOPEN
>>>>>>> Charset Output Test 6 ??
>>>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3053 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3053 ][none      ][-1,-1,-1]
>>>>>>> ************************************************
>>>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3054 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3054 ][none      ][-1,-1,-1]
>>>>>>> f.imei=|354638041679399|
>>>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3055 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3055 ][none      ][-1,-1,-1]
>>>>>>> f.imsi=|405034007414619|
>>>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3056 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3056 ][none      ][-1,-1,-1]
>>>>>>> f.tty_data_device = |/dev/ttyUSB2|
>>>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3057 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3057 ][none      ][-1,-1,-1]
>>>>>>> f.tty_audio_device = |/dev/ttyUSB1|
>>>>>>> 2013-08-22 23:29:53.007103 [NOTICE] mod_gsmopen.cpp:3058 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][NOTICA  3058 ][none      ][-1,-1,-1]
>>>>>>> ************************************************
>>>>>>> 2013-08-22 23:29:53.027123 [WARNING] mod_gsmopen.cpp:1589 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][WARNINGA  1589 ][gsm01     ][-1, 0, 0]
>>>>>>> STARTING interface_id=1
>>>>>>> 2013-08-22 23:29:53.027123 [ERR] gsmopen_protocol.cpp:137 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
>>>>>>> /dev/ttyUSB3, NOT open
>>>>>>> 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:1608 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
>>>>>>> STARTING interface_id=1 FAILED: gsmopen_serial_init failed
>>>>>>>
>>>>>>> +OK Reloading XML
>>>>>>> +OK
>>>>>>>
>>>>>>> 2013-08-22 23:29:53.027123 [ERR] mod_gsmopen.cpp:2684 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
>>>>>>> on interface gsm01:
>>>>>>> freeswitch at dabbian1> 2013-08-22 23:29:53.027123 [CONSOLE]
>>>>>>> switch_loadable_module.c:1401 Successfully Loaded [mod_gsmopen]
>>>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:147
>>>>>>> Adding Endpoint 'gsmopen'
>>>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>>>> Adding API Function 'gsm'
>>>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>>>> Adding API Function 'gsmopen'
>>>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>>>> Adding API Function 'gsmopen_boost_audio'
>>>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>>>> Adding API Function 'gsmopen_dump'
>>>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:305
>>>>>>> Adding API Function 'gsmopen_sendsms'
>>>>>>> 2013-08-22 23:29:53.027123 [NOTICE] switch_loadable_module.c:417
>>>>>>> Adding Chat interface 'sms'
>>>>>>>
>>>>>>> freeswitch at dabbian1>
>>>>>>>
>>>>>>> ---
>>>>>>>
>>>>>>>
>>>>>>> With this understanding, I went ahead and modified the gsmopen
>>>>>>> config file, to:
>>>>>>>   <per_interface_settings>
>>>>>>>     <interface id="1" name="gsm01">
>>>>>>>       <param name="controldevice_name" value="/dev/ttyUSB2"/>
>>>>>>>       <param name="controldevice_audio_name" value="/dev/ttyUSB1"/>
>>>>>>>
>>>>>>> then unloaded mod_gsmopen, and reloaded it, but now, I get this
>>>>>>> error:
>>>>>>>
>>>>>>> 2013-08-22 23:41:43.587104 [ERR] gsmopen_protocol.cpp:137 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
>>>>>>> /dev/ttyUSB3, NOT open
>>>>>>> 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:1608 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  1608 ][gsm01     ][-1, 0, 0]
>>>>>>> STARTING interface_id=1 FAILED: gsmopen_serial_init failed
>>>>>>> 2013-08-22 23:41:43.587104 [ERR] mod_gsmopen.cpp:2684 rev
>>>>>>> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  2684 ][gsm01     ][-1, 0, 0] ALARM
>>>>>>> on interface gsm01:
>>>>>>>
>>>>>>> What does it need /dev/ttyUSB3 for ?
>>>>>>>
>>>>>>> Maybe -- after all, my non linear progression into setting up FS and
>>>>>>> using it by somewhat of trial-n-error isn't helping!
>>>>>>>
>>>>>>> thanks,
>>>>>>> Jay
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 gmangudai at gmail.com  Fri Aug 23 14:52:45 2013
From: gmangudai at gmail.com (Vincent Xia)
Date: Fri, 23 Aug 2013 18:52:45 +0800
Subject: [Freeswitch-users] is it possible to have FS forcilby unregiter a
	registered UA
Message-ID: <CAKQo6tnkgwmneo3QiG0cnBZF5LsApapsYZ4o778fT9SH+5SUGg@mail.gmail.com>

is it possible to have FS forcilby unregiter a registered UA, for example,
user/1001?

to go one step further, how can i do the unregister upon receiving another
register request?
i checked event list of the event socket but cannot find any register
event, can anyone shed any light on this?
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From vipkilla at gmail.com  Fri Aug 23 16:05:56 2013
From: vipkilla at gmail.com (Vik Killa)
Date: Fri, 23 Aug 2013 08:05:56 -0400
Subject: [Freeswitch-users] Need Test Subjects (Color Fax Testing)
In-Reply-To: <95706213-7DBF-49F7-A1DC-B0D8C559695B@freeswitch.org>
References: <F33DFF2F-1855-47DD-9D4E-1646F6F1B3CE@freeswitch.org>
	<CABH8QX3FFNU-QC-_p1nRybB864iZ0YtWa4OZ1WW37K+NPk1O6Q@mail.gmail.com>
	<B3EFA27D0336463687963566F20E1C14@steadfasttelecom.com>
	<CAHgk4BWdp9iQHsZfQmNWu0JLzidQQCtGn0kzD_AAyapKOohzQA@mail.gmail.com>
	<CAC-LwPPCLeX80DLcutwqQi4JDzJazLChivT=mPSXjTi6J19xdg@mail.gmail.com>
	<95706213-7DBF-49F7-A1DC-B0D8C559695B@freeswitch.org>
Message-ID: <CAC-LwPNFcRknxncv64bt-UH_CeoNw_NEq4mQm7w9i7FYN2vtqA@mail.gmail.com>

BKW, just wondering if anyone has tested that yet :)


On Thu, Aug 22, 2013 at 5:31 PM, Brian West <brian at freeswitch.org> wrote:

> But it can.  I did one yesterday LULZ!
>
>
> On Aug 21, 2013, at 3:28 PM, Vik Killa <vipkilla at gmail.com> wrote:
>
> > FS cannot send color faxes yet?
> >
> > <snip>
>
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From fs.user at fordior.net  Fri Aug 23 17:39:30 2013
From: fs.user at fordior.net (EL)
Date: Fri, 23 Aug 2013 15:39:30 +0200
Subject: [Freeswitch-users] is it possible to have FS forcilby unregiter
 a registered UA
In-Reply-To: <CAKQo6tnkgwmneo3QiG0cnBZF5LsApapsYZ4o778fT9SH+5SUGg@mail.gmail.com>
References: <CAKQo6tnkgwmneo3QiG0cnBZF5LsApapsYZ4o778fT9SH+5SUGg@mail.gmail.com>
Message-ID: <20130823133930.GC8814@0rdior.com>


> is it possible to have FS forcilby unregiter a registered UA, for example,
> user/1001?

Yes, try the following:

sofia profile default flush_inbound_reg user at domain

Found this using: 'sofia help' in the freeswitch console. So you can
find all specific commands using this help (it's not documented in
http://wiki.freeswitch.org/wiki/Sofia-SIP).

> to go one step further, how can i do the unregister upon receiving another
> register request?

I'm not sure what you're trying to achieve, but the wiki on sofia-sip
gives some documentation.

> i checked event list of the event socket but cannot find any register
> event, can anyone shed any light on this?

In my console (logging into it with fs_cli -n) it's shows register
events and I believe this is the standard configuration:

[WARNING] sofia_reg.c:1533 SIP auth challenge (REGISTER) on sofia
profile 'internal' for [xxxx at xxxxxx] from ip XXXXXXXX.

-- 
EL


From jayachar88 at gmail.com  Fri Aug 23 18:47:11 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Fri, 23 Aug 2013 20:17:11 +0530
Subject: [Freeswitch-users] Avoiding download of sounds/moh files (reuse
	from another build)
Message-ID: <CAHDL-syuUiDPUST1tn5FYR8MWfU1+Dbm7usjwtXPyRtYGNGt3g@mail.gmail.com>

Hi,

Already cloned Git HEAD (1.5b.5) a day back, and had successfully build FS.

I wish to try latest stable, due to some issues I am facing with the Git
HEAD version.

The longest download for me were the sound/MOH files triggered as a result
of :

make sounds-install
make moh-install

Anyway to reuse the version of the already downloaded sound/MOH files while
rebuilding stable ?

thanks,

Jay
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From mehroz.ashraf85 at gmail.com  Fri Aug 23 19:13:46 2013
From: mehroz.ashraf85 at gmail.com (mehroz)
Date: Fri, 23 Aug 2013 08:13:46 -0700 (PDT)
Subject: [Freeswitch-users] Early media with proxy mode
Message-ID: <1377270826705-7594240.post@n2.nabble.com>

Hi, 


Is it somehow possible to send RBT as early media with proxy media mode?
I have only observed early media in default media mode. I am with ZRTP, thus
cannot switch to default mode.

Please advice!




--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Early-media-with-proxy-mode-tp7594240.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From steveayre at gmail.com  Fri Aug 23 20:01:26 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Aug 2013 17:01:26 +0100
Subject: [Freeswitch-users] is it possible to have FS forcilby unregiter
 a registered UA
In-Reply-To: <CAKQo6tnkgwmneo3QiG0cnBZF5LsApapsYZ4o778fT9SH+5SUGg@mail.gmail.com>
References: <CAKQo6tnkgwmneo3QiG0cnBZF5LsApapsYZ4o778fT9SH+5SUGg@mail.gmail.com>
Message-ID: <CAFiqYukMVqxHhs6rwL7QKZBn0Y-bZ+s3bWqAfUHOdR=33iWqUQ@mail.gmail.com>

>
> to go one step further, how can i do the unregister upon receiving another
> register request?


>From the same user? By default the user can only be registered at one
location.

Bear in mind registrations are timeout based, and refreshed by sending
further REGISTER packets periodically.

That means if you have 2 phones using the same user details the 2nd will
register and replace the 1st, then the 1st reregister and replace the 2nd,
then the 2nd reregister and replace the 1st...

You can't stop them reregistering. Probably wouldn't want to either, or
you'll stop blocking legitimate reregistrations, such as when a phone moves
location.

Not something you can do much about. You shouldn't use the same account in
multiple places. Or allow them to register from multiple locations at once (
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#multiple-registrations)
but bear in mind that'll affect anything that expects them in only one
place.

-Steve



On 23 August 2013 11:52, Vincent Xia <gmangudai at gmail.com> wrote:

> is it possible to have FS forcilby unregiter a registered UA, for example,
> user/1001?
>
> to go one step further, how can i do the unregister upon receiving another
> register request?
> i checked event list of the event socket but cannot find any register
> event, can anyone shed any light on this?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveayre at gmail.com  Fri Aug 23 20:06:11 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Aug 2013 17:06:11 +0100
Subject: [Freeswitch-users] is it possible to have FS forcilby unregiter
 a registered UA
In-Reply-To: <CAKQo6tnkgwmneo3QiG0cnBZF5LsApapsYZ4o778fT9SH+5SUGg@mail.gmail.com>
References: <CAKQo6tnkgwmneo3QiG0cnBZF5LsApapsYZ4o778fT9SH+5SUGg@mail.gmail.com>
Message-ID: <CAFiqYuk5_HHer-aC0o4ZCgqd_8D4v-P0ghnbkRDxvd45efq=QA@mail.gmail.com>

>
> i checked event list of the event socket but cannot find any register
> event, can anyone shed any light on this?


Did you see these?
http://wiki.freeswitch.org/wiki/Sofia#Custom_Events




On 23 August 2013 11:52, Vincent Xia <gmangudai at gmail.com> wrote:

> is it possible to have FS forcilby unregiter a registered UA, for example,
> user/1001?
>
> to go one step further, how can i do the unregister upon receiving another
> register request?
> i checked event list of the event socket but cannot find any register
> event, can anyone shed any light on this?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 a.daydreamer82 at gmail.com  Fri Aug 23 12:57:58 2013
From: a.daydreamer82 at gmail.com (Master Can)
Date: Fri, 23 Aug 2013 10:57:58 +0200
Subject: [Freeswitch-users] How to disable TCP Keepalive on a TLS
	connection?
In-Reply-To: <CA+T6Pv8OGjBCZJ0khfQE-MH5BKRuZ7pc22M28SZk8XMVpBow+g@mail.gmail.com>
References: <CA+T6Pv_SH5mcuYRxHj-2CvM_UAjcQS_NuUKP5E33yy-Ppi2zoQ@mail.gmail.com>
	<70AABE02-0D10-46CA-B43E-901039E14B63@freeswitch.org>
	<CA+T6Pv8OGjBCZJ0khfQE-MH5BKRuZ7pc22M28SZk8XMVpBow+g@mail.gmail.com>
Message-ID: <CA+T6Pv91kuVRy228NMCAkwh8PUH8wdCxOphLK3UK7UEA-Xp_+w@mail.gmail.com>

I am not a C/C++ programmer, but I've downloaded the freeswitch source
code, and did a search for "tcp_keepalive". I found the variable of type
"int" and it is populated with the value tcp-keepalive from the XML file
(in sofia.c). But apart from that the variable is not used at all?? Am I
overseeing something?


2013/8/22 Master Can <a.daydreamer82 at gmail.com>

> Originally the option was not there - so I tried running Freeswitch
> without the option already. That didn't influence tcp keepalives though...
>
>
> 2013/8/21 Brian West <brian at freeswitch.org>
>
>> Remove the option.
>>
>> On Aug 21, 2013, at 12:44 PM, Master Can <a.daydreamer82 at gmail.com>
>> wrote:
>>
>> > Hello,
>> >
>> > I'm running freeswitch 1.2.10, with tls-only.
>> > I've tried 2 SIP Useragents now: PhonerLite and CSipSimple.
>> >
>> > My linux server shows me with netstat --timers that both useragents
>> (both server sockets) use keepalive, with a value of 30 seconds.
>> >
>> > How can I disable keepalive on the TCP layer completely? My useragents
>> take care of sending keepalive packets anyway, so Freeswitch does not need
>> to do that. It's not mobile friendly, it's eating up battery power if the
>> useragents keep receiving keepalive every 30 seconds.
>> >
>> > I've tried to set
>> > <param name="tcp-keepalive" value="disable"/>
>> > in internal.xml but to no avail. It didn't change a thing. Setting this
>> to 60000 didn't change the output of netstat --timers either.
>> >
>> > Any advice?
>> >
>> > best regards,
>> > Can
>> > _______
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 a.daydreamer82 at gmail.com  Fri Aug 23 13:35:18 2013
From: a.daydreamer82 at gmail.com (Master Can)
Date: Fri, 23 Aug 2013 11:35:18 +0200
Subject: [Freeswitch-users] How to disable TCP Keepalive on a TLS
	connection?
In-Reply-To: <CA+T6Pv91kuVRy228NMCAkwh8PUH8wdCxOphLK3UK7UEA-Xp_+w@mail.gmail.com>
References: <CA+T6Pv_SH5mcuYRxHj-2CvM_UAjcQS_NuUKP5E33yy-Ppi2zoQ@mail.gmail.com>
	<70AABE02-0D10-46CA-B43E-901039E14B63@freeswitch.org>
	<CA+T6Pv8OGjBCZJ0khfQE-MH5BKRuZ7pc22M28SZk8XMVpBow+g@mail.gmail.com>
	<CA+T6Pv91kuVRy228NMCAkwh8PUH8wdCxOphLK3UK7UEA-Xp_+w@mail.gmail.com>
Message-ID: <CA+T6Pv-=AA0BG1C-vVHAoCEx=-UzMnO_sFN3XwPShCvCRu0w0g@mail.gmail.com>

When digging further into the source code, I found in
libs\sofia-sip\libsofia-sip-ua\tport\tport_type_tcp.c:

#if defined(SO_KEEPALIVE)
  setsockopt(socket, SOL_SOCKET, SO_KEEPALIVE, (void *)&val, sizeof val);
#endif
  val = 30;
#if defined(TCP_KEEPIDLE)
  setsockopt(socket, SOL_TCP, TCP_KEEPIDLE, (void *)&val, sizeof val);
#endif
#if defined(TCP_KEEPINTVL)
  setsockopt(socket, SOL_TCP, TCP_KEEPINTVL, (void *)&val, sizeof val);
#endif

The 30 seconds are hardcoded...
Now to turn off keepalive, I'd have to change the header file where
SO_KEEPALIVE is set and to compile freeswitch?

Is there a more straightforward way?

br,
Can




2013/8/23 Master Can <a.daydreamer82 at gmail.com>

> I am not a C/C++ programmer, but I've downloaded the freeswitch source
> code, and did a search for "tcp_keepalive". I found the variable of type
> "int" and it is populated with the value tcp-keepalive from the XML file
> (in sofia.c). But apart from that the variable is not used at all?? Am I
> overseeing something?
>
>
> 2013/8/22 Master Can <a.daydreamer82 at gmail.com>
>
>> Originally the option was not there - so I tried running Freeswitch
>> without the option already. That didn't influence tcp keepalives though...
>>
>>
>> 2013/8/21 Brian West <brian at freeswitch.org>
>>
>>>  Remove the option.
>>>
>>> On Aug 21, 2013, at 12:44 PM, Master Can <a.daydreamer82 at gmail.com>
>>> wrote:
>>>
>>> > Hello,
>>> >
>>> > I'm running freeswitch 1.2.10, with tls-only.
>>> > I've tried 2 SIP Useragents now: PhonerLite and CSipSimple.
>>> >
>>> > My linux server shows me with netstat --timers that both useragents
>>> (both server sockets) use keepalive, with a value of 30 seconds.
>>> >
>>> > How can I disable keepalive on the TCP layer completely? My useragents
>>> take care of sending keepalive packets anyway, so Freeswitch does not need
>>> to do that. It's not mobile friendly, it's eating up battery power if the
>>> useragents keep receiving keepalive every 30 seconds.
>>> >
>>> > I've tried to set
>>> > <param name="tcp-keepalive" value="disable"/>
>>> > in internal.xml but to no avail. It didn't change a thing. Setting
>>> this to 60000 didn't change the output of netstat --timers either.
>>> >
>>> > Any advice?
>>> >
>>> > best regards,
>>> > Can
>>> > _______
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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 steveayre at gmail.com  Fri Aug 23 20:31:27 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 23 Aug 2013 17:31:27 +0100
Subject: [Freeswitch-users] How to disable TCP Keepalive on a TLS
	connection?
In-Reply-To: <CA+T6Pv-=AA0BG1C-vVHAoCEx=-UzMnO_sFN3XwPShCvCRu0w0g@mail.gmail.com>
References: <CA+T6Pv_SH5mcuYRxHj-2CvM_UAjcQS_NuUKP5E33yy-Ppi2zoQ@mail.gmail.com>
	<70AABE02-0D10-46CA-B43E-901039E14B63@freeswitch.org>
	<CA+T6Pv8OGjBCZJ0khfQE-MH5BKRuZ7pc22M28SZk8XMVpBow+g@mail.gmail.com>
	<CA+T6Pv91kuVRy228NMCAkwh8PUH8wdCxOphLK3UK7UEA-Xp_+w@mail.gmail.com>
	<CA+T6Pv-=AA0BG1C-vVHAoCEx=-UzMnO_sFN3XwPShCvCRu0w0g@mail.gmail.com>
Message-ID: <CAFiqYunKmMFOiUcRSvSh09z3neHRPxhKwfRM6BOH2LJ=U1g2Nw@mail.gmail.com>

SO_KEEPALIVE is defined in the system headers. You shouldn't undefine it.

Rather if you want to tune that code you should patch sofia and mod_sofia
to allow you to do so.


On 23 August 2013 10:35, Master Can <a.daydreamer82 at gmail.com> wrote:

> When digging further into the source code, I found in
> libs\sofia-sip\libsofia-sip-ua\tport\tport_type_tcp.c:
>
> #if defined(SO_KEEPALIVE)
>   setsockopt(socket, SOL_SOCKET, SO_KEEPALIVE, (void *)&val, sizeof val);
> #endif
>   val = 30;
> #if defined(TCP_KEEPIDLE)
>   setsockopt(socket, SOL_TCP, TCP_KEEPIDLE, (void *)&val, sizeof val);
> #endif
> #if defined(TCP_KEEPINTVL)
>   setsockopt(socket, SOL_TCP, TCP_KEEPINTVL, (void *)&val, sizeof val);
> #endif
>
> The 30 seconds are hardcoded...
> Now to turn off keepalive, I'd have to change the header file where
> SO_KEEPALIVE is set and to compile freeswitch?
>
> Is there a more straightforward way?
>
> br,
> Can
>
>
>
>
> 2013/8/23 Master Can <a.daydreamer82 at gmail.com>
>
>> I am not a C/C++ programmer, but I've downloaded the freeswitch source
>> code, and did a search for "tcp_keepalive". I found the variable of type
>> "int" and it is populated with the value tcp-keepalive from the XML file
>> (in sofia.c). But apart from that the variable is not used at all?? Am I
>> overseeing something?
>>
>>
>> 2013/8/22 Master Can <a.daydreamer82 at gmail.com>
>>
>>> Originally the option was not there - so I tried running Freeswitch
>>> without the option already. That didn't influence tcp keepalives though...
>>>
>>>
>>> 2013/8/21 Brian West <brian at freeswitch.org>
>>>
>>>>  Remove the option.
>>>>
>>>> On Aug 21, 2013, at 12:44 PM, Master Can <a.daydreamer82 at gmail.com>
>>>> wrote:
>>>>
>>>> > Hello,
>>>> >
>>>> > I'm running freeswitch 1.2.10, with tls-only.
>>>> > I've tried 2 SIP Useragents now: PhonerLite and CSipSimple.
>>>> >
>>>> > My linux server shows me with netstat --timers that both useragents
>>>> (both server sockets) use keepalive, with a value of 30 seconds.
>>>> >
>>>> > How can I disable keepalive on the TCP layer completely? My
>>>> useragents take care of sending keepalive packets anyway, so Freeswitch
>>>> does not need to do that. It's not mobile friendly, it's eating up battery
>>>> power if the useragents keep receiving keepalive every 30 seconds.
>>>> >
>>>> > I've tried to set
>>>> > <param name="tcp-keepalive" value="disable"/>
>>>> > in internal.xml but to no avail. It didn't change a thing. Setting
>>>> this to 60000 didn't change the output of netstat --timers either.
>>>> >
>>>> > Any advice?
>>>> >
>>>> > best regards,
>>>> > Can
>>>> > _______
>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ssinyagin at yahoo.com  Sat Aug 24 00:28:19 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Fri, 23 Aug 2013 13:28:19 -0700 (PDT)
Subject: [Freeswitch-users] domain vs. domain_name variables
Message-ID: <1377289699.93283.YahooMailNeo@web126203.mail.ne1.yahoo.com>

What is the difference between ${domain} and ${domain_name} variables? Both are used in the source code, and example configs set them to the same value.

What is the design consideration behind?

Why not merging them into one variable?


thanks
stan
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From rsaavedra at 018000web.co  Sat Aug 24 01:09:53 2013
From: rsaavedra at 018000web.co (Ricardo Saavedra)
Date: Fri, 23 Aug 2013 21:09:53 +0000 (UTC)
Subject: [Freeswitch-users] Secure Websocket Setup
In-Reply-To: <1308726942.15.1377290484854.JavaMail.root@018000web.co>
Message-ID: <2065652951.99.1377292193211.JavaMail.root@018000web.co>



Hello, 

The webrtc setup instructions are: 

<!-- uncomment for sip over websocket support --> 
<!--<param name="ws-binding" value=":5066"/>--> 

<!-- uncomment for sip over secure websocket support --> 
<!-- You need wss.pem in /usr/local/freeswitch/certs for wss --> 
<!--<param name="wss-binding" value=":7443"/>--> 

Now I have a test server for the Websocket and is working. ): 

For the secure websocket I have some questions: 

Freeswitch have any tool to generate a self-signed wss.pem file? 

Can I use the files: usr/local/freeswitch/certs/dtls-srtp.crt and 
usr/local/freeswitch/certs/ dtls-srtp.key 

to generate the wss.pem that I need? 


Thank you, 

Ricardo Saavedra 


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From mike at jerris.com  Sat Aug 24 01:33:27 2013
From: mike at jerris.com (Michael Jerris)
Date: Fri, 23 Aug 2013 17:33:27 -0400
Subject: [Freeswitch-users] Secure Websocket Setup
In-Reply-To: <2065652951.99.1377292193211.JavaMail.root@018000web.co>
References: <2065652951.99.1377292193211.JavaMail.root@018000web.co>
Message-ID: <AA86DB46-7D55-4804-B18E-5304E2907694@jerris.com>

We will self-gen a self signed wss.pem if you do not supply one.

On Aug 23, 2013, at 5:09 PM, Ricardo Saavedra <rsaavedra at 018000web.co> wrote:

> Hello,
> 
> The webrtc setup instructions are:
> 
> <!-- uncomment for sip over websocket support -->
> <!--<param name="ws-binding" value=":5066"/>-->
> 
> <!-- uncomment for sip over secure websocket support -->
> <!-- You need wss.pem in /usr/local/freeswitch/certs for wss -->
> <!--<param name="wss-binding" value=":7443"/>-->
> 
> Now I have a test server for the Websocket and is working. ):
> 
> For the secure websocket I have some questions:
> 
> Freeswitch have any tool to generate a self-signed wss.pem file?
> 
> Can I use the files: usr/local/freeswitch/certs/dtls-srtp.crt and
> usr/local/freeswitch/certs/ dtls-srtp.key
> 
> to generate the wss.pem that I need?
> 

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From karl at xtronics.com  Sat Aug 24 01:55:11 2013
From: karl at xtronics.com (Karl Schmidt)
Date: Fri, 23 Aug 2013 16:55:11 -0500
Subject: [Freeswitch-users] domain vs. domain_name variables
In-Reply-To: <1377289699.93283.YahooMailNeo@web126203.mail.ne1.yahoo.com>
References: <1377289699.93283.YahooMailNeo@web126203.mail.ne1.yahoo.com>
Message-ID: <5217DA3F.70205@xtronics.com>

On 08/23/2013 03:28 PM, Stanislav Sinyagin wrote:
> What is the difference between ${domain} and ${domain_name} variables? Both are used in the source
> code, and example configs set them to the same value.
>
> What is the design consideration behind?
>
> Why not merging them into one variable?
>

This is an issue that has me a bit puzzled as well.

domain in vars.xml is supposed to be an IP address ( IMO Should have been called domain_ip )
domain_name is supposed to be a FQDN I think. grepping the configs tells me domain_name is never 
used in the default config so you can probably ignore it.

What is also not clear is the multiple uses of the term domain:


 From the wiki -
https://wiki.freeswitch.org/wiki/Sofia.conf.xml#The_Relationship_Between_SIP_Profiles_and_Domains :

 > The domains inside the XML registry are completely different from the domains on the internet and
 > again completely different from domains in sip packets. The profiles are again entirely different
 > from any of the above. Its up to you to align them if you so choose.

OK that made my head hurt.

In some places in FS 'domain' = domain_IP_address

Here it says that SIP profile domain != IP-domain != SIP packet domain.

So I know what it isn't, but it might also help if I knew what it IS!

So, I think the domain definition in sofia MIGHT be a grouping of profiles? Then as you read on, it 
tells us that a domain is a domain.

" FreeSWITCH works off the concept of users and domains just like email."

The books nor the wiki nor googling the mailing list makes this clear.

Both the internal and external profiles set the name of the sofia-domain to all and $ sofia status 
gives the names as internal and external (after the directory? ).




--------------------------------------------------------------------------------
Karl Schmidt                                  EMail Karl at xtronics.com
Transtronics, Inc.                              WEB http://secure.transtronics.com
3209 West 9th Street                             Ph (785) 841-3089
Lawrence, KS 66049                              FAX (785) 841-0434

Reality isn't fair, but that it is where I live. -kps
--------------------------------------------------------------------------------


From anthony.minessale at gmail.com  Sat Aug 24 03:55:31 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 23 Aug 2013 18:55:31 -0500
Subject: [Freeswitch-users] Secure Websocket Setup
In-Reply-To: <AA86DB46-7D55-4804-B18E-5304E2907694@jerris.com>
References: <2065652951.99.1377292193211.JavaMail.root@018000web.co>
	<AA86DB46-7D55-4804-B18E-5304E2907694@jerris.com>
Message-ID: <CAKbxfG9+OCmi0j6Vgf0_S_RdEzm6yHxAMb_SP9wSe37i7ghEnA@mail.gmail.com>

And I don't think it actually works.  You need a valid cert/key pair that
would work in apach catted together into a pem file.



On Fri, Aug 23, 2013 at 4:33 PM, Michael Jerris <mike at jerris.com> wrote:

> We will self-gen a self signed wss.pem if you do not supply one.
>
> On Aug 23, 2013, at 5:09 PM, Ricardo Saavedra <rsaavedra at 018000web.co>
> wrote:
>
> Hello,
>
> The webrtc setup instructions are:
>
> <!-- uncomment for sip over websocket support -->
> <!--<param name="ws-binding" value=":5066"/>-->
>
> <!-- uncomment for sip over secure websocket support -->
> <!-- You need wss.pem in /usr/local/freeswitch/certs for wss -->
> <!--<param name="wss-binding" value=":7443"/>-->
>
> Now I have a test server for the Websocket and is working. ):
>
> For the secure websocket I have some questions:
>
> Freeswitch have any tool to generate a self-signed wss.pem file?
>
> Can I use the files: usr/local/freeswitch/certs/dtls-srtp.crt and
> usr/local/freeswitch/certs/ dtls-srtp.key
>
> to generate the wss.pem that I need?
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From anthony.minessale at gmail.com  Sat Aug 24 03:58:10 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 23 Aug 2013 18:58:10 -0500
Subject: [Freeswitch-users] Early media with proxy mode
In-Reply-To: <1377270826705-7594240.post@n2.nabble.com>
References: <1377270826705-7594240.post@n2.nabble.com>
Message-ID: <CAKbxfG-tBxAJLfJxUfeFA9Dkk+7wUb0BWXDbhs_HS512iQmQbg@mail.gmail.com>

I don't know what RBT means but you can't send any media from FS itself if
you want to do successful proxy media.
Attempting to send media results in a sdp being generated and we do not
support transitioning from regular media to proxy media once the call has
reached that state.




On Fri, Aug 23, 2013 at 10:13 AM, mehroz <mehroz.ashraf85 at gmail.com> wrote:

> Hi,
>
>
> Is it somehow possible to send RBT as early media with proxy media mode?
> I have only observed early media in default media mode. I am with ZRTP,
> thus
> cannot switch to default mode.
>
> Please advice!
>
>
>
>
> --
> View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Early-media-with-proxy-mode-tp7594240.html
> Sent from the freeswitch-users mailing list archive at Nabble.com.
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From anthony.minessale at gmail.com  Sat Aug 24 05:03:59 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 23 Aug 2013 20:03:59 -0500
Subject: [Freeswitch-users] domain vs. domain_name variables
In-Reply-To: <5217DA3F.70205@xtronics.com>
References: <1377289699.93283.YahooMailNeo@web126203.mail.ne1.yahoo.com>
	<5217DA3F.70205@xtronics.com>
Message-ID: <CAKbxfG8HoigFN64JiUChGuj9X2r5ydNtcHRdrP0Mm5o-xSVeHA@mail.gmail.com>

It took me like an hour to compose this explanation.....I hope it at least
gives you an idea.

"domain" is a core variable its used as a fallback in certain situations
when its defined and no other value is specified for domain.
It's also used in the default configuration as a pre-processor substitution
with the $${domain} notation which is substituted into the sofia profiles
and conference etc.


"domain_name" is a variable that only exists in the scope of the demo
dialplan.  Its set initially as a global in vars.xml to whatever domain is
set to as a base value.
>From there its only set on a per-call basis to override and used in per
call dialplan logic.

Its is not true that domain must be an IP.  It defaults to the same value
as local_ip_v4 to make sure the system works on any box without any
configuration.
If you have a working FQHN that points to your box, you should set that
value in your domain var instead of the ip if you want things to reference
the actual domain name.


I think once you learn that way the config files are setup in the default
distribution are purely arbitrary the more sense it may begin to make.


The thing that made your head hurt is trying to explain that in SIP there
is a separation from the host portion of a URI and the IP its sending the
packet to.
Because of PROXIES etc you can easily send a SIP packet to a specific
address that contains a request URI or To: header with a different host
name or ip.


think of domain as referring to the host portion of a user at host concept.
 The same notion of a REALM in radius.  In SIP and in FS that can be an IP
or a FQHN and it need not exist in the TLD dns on the internet or it can
match one that exists and function on its own because it's not directly
tied to dns.  (For the sake of sanity do not use a real domain name that is
not yours even if it works).

Basically the path of least resistance is to keep all these names aligned..
This is what the default config does for you because you set the domain
variable once and its pre-processed into all the key places in the config
that have domains so they are the same everywhere.

Say you set it to example.com

* The default directory will have a domain defined called example.com with
the default 1000-1019 extensions.
* The internal sip profile will associate the internal profile with
example.com and force any registrations etc to find that domain for auth.
* The conferences will use example.com for presence etc.

Then it uses domain_name in the extensions etc as a clone of that
example.com domain that you can change at runtime on a per call basis.
 Most likely in the default config with no changes this value will remain
example.com the whole time and there is no difference.

Why?

Because people love multi-home stuff and they want to configure all of
these things separately and once they start doing this, the config still is
usable.

So:

FS never resolves domains to the IP and vice versa for you like it might in
your unix environment or in a web browser.  IP and FQHN have the same
significance in sip packets as a unique realm string.  It's only because
you are binding the SIP UA to ${local_ip_v4} that your sip works on the IP.

The way SIP was designed to work, you would define a SRV record for the
host name used in your sip uri and then that IP would be used in place of a
default A record lookup.  We can just as easily override that in a client
by forcing the fields for the proxy address and domains.


Say you used example.com like I mentioned above and your valid IP was
200.8.8.8

You could change domain=example.com and restart FS and it would still come
up.

Then you could go take a sip soft phone and play with the advanced settings
and set the proxy addr to 200.8.8.8 and set the host or domain field to
example.com and configure it to always send calls to the proxy addr.

Now you can use example.com in your phones all day long even though its not
really your domain because its just saying that in the packets and you
explicitly showed it what server to use.

Next, instead of example.com you could use 1.2.3.4 (or some real ip) as the
domain even though it's obviously silly.

If you turn on the siptrace you'll see the packets hit your box and using
those values like 1.2.3.4 and example.com in the host portion and talk to
your client over the actual IP of the server.
















































On Fri, Aug 23, 2013 at 4:55 PM, Karl Schmidt <karl at xtronics.com> wrote:

> On 08/23/2013 03:28 PM, Stanislav Sinyagin wrote:
> > What is the difference between ${domain} and ${domain_name} variables?
> Both are used in the source
> > code, and example configs set them to the same value.
> >
> > What is the design consideration behind?
> >
> > Why not merging them into one variable?
> >
>
> This is an issue that has me a bit puzzled as well.
>
> domain in vars.xml is supposed to be an IP address ( IMO Should have been
> called domain_ip )
> domain_name is supposed to be a FQDN I think. grepping the configs tells
> me domain_name is never
> used in the default config so you can probably ignore it.
>
> What is also not clear is the multiple uses of the term domain:
>
>
>  From the wiki -
>
> https://wiki.freeswitch.org/wiki/Sofia.conf.xml#The_Relationship_Between_SIP_Profiles_and_Domains:
>
>  > The domains inside the XML registry are completely different from the
> domains on the internet and
>  > again completely different from domains in sip packets. The profiles
> are again entirely different
>  > from any of the above. Its up to you to align them if you so choose.
>
> OK that made my head hurt.
>
> In some places in FS 'domain' = domain_IP_address
>
> Here it says that SIP profile domain != IP-domain != SIP packet domain.
>
> So I know what it isn't, but it might also help if I knew what it IS!
>
> So, I think the domain definition in sofia MIGHT be a grouping of
> profiles? Then as you read on, it
> tells us that a domain is a domain.
>
> " FreeSWITCH works off the concept of users and domains just like email."
>
> The books nor the wiki nor googling the mailing list makes this clear.
>
> Both the internal and external profiles set the name of the sofia-domain
> to all and $ sofia status
> gives the names as internal and external (after the directory? ).
>
>
>
>
>
> --------------------------------------------------------------------------------
> Karl Schmidt                                  EMail Karl at xtronics.com
> Transtronics, Inc.                              WEB
> http://secure.transtronics.com
> 3209 West 9th Street                             Ph (785) 841-3089
> Lawrence, KS 66049                              FAX (785) 841-0434
>
> Reality isn't fair, but that it is where I live. -kps
>
> --------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From bdfoster at davri.com  Sat Aug 24 05:12:35 2013
From: bdfoster at davri.com (Brian Foster)
Date: Fri, 23 Aug 2013 21:12:35 -0400
Subject: [Freeswitch-users] Tuning DTMF
In-Reply-To: <FB7D97A214987F458242ACBDF87614073B1BD19CD6@clubvirtual40.ClubMessage.local>
References: <FB7D97A214987F458242ACBDF87614073B1BD19CD6@clubvirtual40.ClubMessage.local>
Message-ID: <CAHgk4BWrESZ=cnZonBHTPQr0r3aUzOYZHJpPGOmzsA3xMoc-Rg@mail.gmail.com>

Have you tried using Xlite on an IVR or something? Call your bank or your
cable company and see if you can get through the menus fine. I'd bet that
you can. Post back and let us know. They sound short but that should be ok.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 23, 2013 3:21 AM, "Grant Bagdasarian" <GB at cm.nl> wrote:

> Hello,****
>
> ** **
>
> Setup is as following:****
>
> ** **
>
> X-Lite (Test phone) ------>  SIP Proxy ------> FS (as SBC) ------> Carrier
> ------> End-user phone****
>
> ** **
>
> Whenever I place a call using X-Lite to the End-user phone, in this case
> my own mobile phone or office phone, and start sending DTMF tones from the
> remote telephone, the DMTF tones arrive with a crackling sound at the end
> of the tone.****
>
> ** **
>
> Console debug:****
>
> 2013-08-23 09:00:13.813207 [DEBUG] switch_rtp.c:3829 Send start packet for
> [2] ts=3778476449 dur=160/160/1280 seq=5098 lw=-516490847****
>
> 2013-08-23 09:00:13.833208 [DEBUG] switch_rtp.c:3729 Send middle packet
> for [2] ts=3778476449 dur=320/320/1280 seq=5099 lw=161****
>
> 2013-08-23 09:00:13.853208 [DEBUG] switch_rtp.c:3729 Send middle packet
> for [2] ts=3778476449 dur=480/480/1280 seq=5100 lw=161****
>
> 2013-08-23 09:00:13.873208 [DEBUG] switch_rtp.c:3729 Send middle packet
> for [2] ts=3778476449 dur=640/640/1280 seq=5101 lw=161****
>
> 2013-08-23 09:00:13.893208 [DEBUG] switch_rtp.c:3729 Send middle packet
> for [2] ts=3778476449 dur=800/800/1280 seq=5102 lw=161****
>
> 2013-08-23 09:00:13.913208 [DEBUG] switch_rtp.c:3729 Send middle packet
> for [2] ts=3778476449 dur=960/960/1280 seq=5103 lw=161****
>
> 2013-08-23 09:00:13.933208 [DEBUG] switch_rtp.c:3729 Send middle packet
> for [2] ts=3778476449 dur=1120/1120/1280 seq=5104 lw=161****
>
> 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for
> [2] ts=3778476449 dur=1280/1280/1280 seq=5105 lw=1****
>
> 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for
> [2] ts=3778476449 dur=1280/1280/1280 seq=5106 lw=1****
>
> 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for
> [2] ts=3778476449 dur=1280/1280/1280 seq=5107 lw=1****
>
> 2013-08-23 09:00:14.813206 [DEBUG] switch_rtp.c:5117 RTP RECV DTMF 1:1280*
> ***
>
> 2013-08-23 09:00:14.813206 [DEBUG] switch_channel.c:471 RECV DTMF 1:1280**
> **
>
> ** **
>
> Both internal and external sip profiles have the following configuration
> for DTMF:****
>
> <param name="liberal-dtmf" value="true"/>****
>
> <param name="rfc2833-pt" value="101"/>****
>
> <param name="dtmf-duration" value="2000"/>****
>
> <param name="inbound-codec-negotiation" value="generous"/>****
>
> <param name="inbound-late-negotiation" value="true"/>****
>
> ** **
>
> Internal.xml (Dialplan)****
>
>     <extension name="internal_to_external">****
>
>       <condition field="${sip_req_host}" expression="^10\.0\.0\.1$">****
>
>         <action application="set" data="hangup_after_bridge=false"/>****
>
>         <action application="set" data="continue_on_fail=true"/>****
>
>         <action application="bridge" data="sofia/external/${sip_req_uri}"/>
> ****
>
>       </condition>****
>
>     </extension>****
>
> ** **
>
> Is there a way to tune the DTMF tones? If so, which parameters do I need
> to add or edit?****
>
> ** **
>
> Thanks!****
>
> ** **
>
> Grant****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Sat Aug 24 06:49:58 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Sat, 24 Aug 2013 08:19:58 +0530
Subject: [Freeswitch-users] Cluecon presentations - archived / available ?
Message-ID: <CAHDL-swipeXhTGtE50g0dj24bwWoPok2uEPSm8=BxLeyPrsFnw@mail.gmail.com>

Not sure if this is the right list for asking this question, but I wasn't
aware of a better one.

Are the presentations done in past Cluecons archived and available for
download from some location ? Scribd / Slideshare etc. ?
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From william.king at quentustech.com  Sat Aug 24 06:59:30 2013
From: william.king at quentustech.com (William King)
Date: Fri, 23 Aug 2013 19:59:30 -0700
Subject: [Freeswitch-users] Cluecon presentations - archived / available
 ?
In-Reply-To: <CAHDL-swipeXhTGtE50g0dj24bwWoPok2uEPSm8=BxLeyPrsFnw@mail.gmail.com>
References: <CAHDL-swipeXhTGtE50g0dj24bwWoPok2uEPSm8=BxLeyPrsFnw@mail.gmail.com>
Message-ID: <52182192.8070605@quentustech.com>

http://torrents.freeswitch.org/

At least the 2011 ClueCon torrent is there.

William King
Senior Engineer
Quentus Technologies, INC
1037 NE 65th St Suite 273
Seattle, WA 98115
Main:   (877) 211-9337
Office: (206) 388-4772
Cell:   (253) 686-5518
william.king at quentustech.com

On 08/23/2013 07:49 PM, Jayanth Acharya wrote:
> Not sure if this is the right list for asking this question, but I
> wasn't aware of a better one.
> 
> Are the presentations done in past Cluecons archived and available for
> download from some location ? Scribd / Slideshare etc. ?
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Sat Aug 24 08:42:29 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Sat, 24 Aug 2013 10:12:29 +0530
Subject: [Freeswitch-users] Cluecon presentations - archived / available
	?
In-Reply-To: <52182192.8070605@quentustech.com>
References: <CAHDL-swipeXhTGtE50g0dj24bwWoPok2uEPSm8=BxLeyPrsFnw@mail.gmail.com>
	<52182192.8070605@quentustech.com>
Message-ID: <CAHDL-swWULXmUVghAN7xEvF=1Qao48F=xOuyfmbh-qfemQ+UiQ@mail.gmail.com>

Many thanks. Wonder if other list members are aware of Cluecon 2013
decks... there are 2-3 I am quite interested in seeing.

Just as a hint to organizers... if recorded videos of sessions with
presentations are made available for reasonable fee per session, I think
many of us might lap it up !


On Sat, Aug 24, 2013 at 8:29 AM, William King
<william.king at quentustech.com>wrote:

> http://torrents.freeswitch.org/
>
> At least the 2011 ClueCon torrent is there.
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
> On 08/23/2013 07:49 PM, Jayanth Acharya wrote:
> > Not sure if this is the right list for asking this question, but I
> > wasn't aware of a better one.
> >
> > Are the presentations done in past Cluecons archived and available for
> > download from some location ? Scribd / Slideshare etc. ?
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Sat Aug 24 09:41:22 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Sat, 24 Aug 2013 11:11:22 +0530
Subject: [Freeswitch-users] Cluecon presentations - archived / available
	?
In-Reply-To: <CAHDL-swWULXmUVghAN7xEvF=1Qao48F=xOuyfmbh-qfemQ+UiQ@mail.gmail.com>
References: <CAHDL-swipeXhTGtE50g0dj24bwWoPok2uEPSm8=BxLeyPrsFnw@mail.gmail.com>
	<52182192.8070605@quentustech.com>
	<CAHDL-swWULXmUVghAN7xEvF=1Qao48F=xOuyfmbh-qfemQ+UiQ@mail.gmail.com>
Message-ID: <CAHDL-syaxkctmu6a_vnLB-cGqOuFcuUVrYRB_NgJo8_deh_GMg@mail.gmail.com>

The torrent is no good, as no one is seeding it !


On Sat, Aug 24, 2013 at 10:12 AM, Jayanth Acharya <jayachar88 at gmail.com>wrote:

> Many thanks. Wonder if other list members are aware of Cluecon 2013
> decks... there are 2-3 I am quite interested in seeing.
>
> Just as a hint to organizers... if recorded videos of sessions with
> presentations are made available for reasonable fee per session, I think
> many of us might lap it up !
>
>
> On Sat, Aug 24, 2013 at 8:29 AM, William King <
> william.king at quentustech.com> wrote:
>
>> http://torrents.freeswitch.org/
>>
>> At least the 2011 ClueCon torrent is there.
>>
>> William King
>> Senior Engineer
>> Quentus Technologies, INC
>> 1037 NE 65th St Suite 273
>> Seattle, WA 98115
>> Main:   (877) 211-9337
>> Office: (206) 388-4772
>> Cell:   (253) 686-5518
>> william.king at quentustech.com
>>
>> On 08/23/2013 07:49 PM, Jayanth Acharya wrote:
>> > Not sure if this is the right list for asking this question, but I
>> > wasn't aware of a better one.
>> >
>> > Are the presentations done in past Cluecons archived and available for
>> > download from some location ? Scribd / Slideshare etc. ?
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.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
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 anton.vazir at gmail.com  Sat Aug 24 13:31:28 2013
From: anton.vazir at gmail.com (Anton VG)
Date: Sat, 24 Aug 2013 13:31:28 +0400
Subject: [Freeswitch-users] freeswitch HANGING (process exists but stops
	responding)
In-Reply-To: <CE2EA466.78CDE%krice@freeswitch.org>
References: <CAE2+g+KR-kZkrhxH3NekjsCk5+CiKG2j6Q5UTiR6TWa-QpGU9Q@mail.gmail.com>
	<CE2EA466.78CDE%krice@freeswitch.org>
Message-ID: <CAGX+XzvuYewzSoTjREMWJ7D+=ZPvKDt_KrwJjx71MC=TM39OUw@mail.gmail.com>

Hi Nino, I do experience very similar problem with latest stable: FS
1.2.12. Please let me know if you have managed resolving that.

So, eventually my FS stops responding, generating events, requesting
xml_curl_auth, etc. I can start fs_cli but when I do want to have "show
bridged_calls" and enter sh and press tab "sho" appears and fs_cli totally
hangs - i have falled back to 1.1beta1 where I do not saw that problem. But
I do not see "Create event dispatch thread" messages in the log.
In my setup I have quite busy installation with around 1000 active SIP
registrations with ~100 simultaneous calls, and ESL controller over the
network, so EVENT activity is quite intensive.

Regards, Anton.

2013/8/12 Ken Rice <krice at freeswitch.org>

>  You might want to update to lastest and test then it could have already
> been resolved...
>
>
> On 8/12/13 2:35 PM, "Nuno Reis" <nreis at wavecom.pt> wrote:
>
> Another eventually important tip:
>
> This is happening at least with freeswitch versions 1.2.3 and 1.2.8.
>
> Thanks.
> --
>
> *Nuno Miguel Reis* | *Unified Communication Systems
> *
> M. +351 913907481 | nreis at wavecom.pt <mailto:nreis at wavecom.pt<nreis at wavecom.pt>>
>
> WAVECOM-Solu??es R?dio, S.A.
> Cacia Park | Rua do Progresso, Lote 15
> 3800-639 AVEIRO | Portugal
> T. +351 309 700 225 | F. +351 234 919 191
> *GPS** <
> http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>
>  | www.wavecom.pt <http://www.wavecom.pt/> *
> * <http://www.wavecom.pt/>
> *
>
>  <http://www.wavecom.pt/pt/wavecom/premios.php>
>
>  <http://www.wavecom.pt/pt/mail_eventos.php>
>
>
>
>
> On Mon, Aug 12, 2013 at 8:29 PM, Nuno Reis <nreis at wavecom.pt> wrote:
>
> Hello all.
>
> I've been experiencing a problem with freeswitch where it stops responding
> (doesn't answer any SIP REQUESTS) and produces the following output in the
> log file every time the symptom happens:
>
> 2013-08-12 15:18:31.488906 [WARNING] switch_event.c:552 Create event
> dispatch thread 1
> 2013-08-12 15:18:31.578906 [WARNING] switch_event.c:554 Create additional
> event dispatch thread 2
> 2013-08-12 15:18:31.598907 [WARNING] switch_event.c:554 Create additional
> event dispatch thread 3
> 2013-08-12 15:18:31.648904 [WARNING] switch_event.c:554 Create additional
> event dispatch thread 4
> 2013-08-12 15:18:31.728905 [WARNING] switch_event.c:554 Create additional
> event dispatch thread 5
> 2013-08-12 15:19:04.348914 [CONSOLE] sofia.c:1602 MSG Thread 4 Started
>
> After this, i can see that the process exists and i can get to the fs_cli
> prompt, although none of the normal command like: show calls work. The
> command simply hangs without returning back a response.
> I think this could be memory related, although can't find a reasonable
> explanation to that. I had 4GB of physical RAM available + 512MB for SWAP
> on a CentOS x86_64 machine and the OS was using like about 800MB but if i
> considered  the cached amount of memory, it was like a total of 3.8GB which
> from what i understand is normal in linux, since the kernel tries to cache
> all the available memory over time.
> Nevertheless i've decided to give the machine more 4GB RAM and SWAP is now
> 16GB, so now i've 8GB total for physical RAM and 16GB SWAP which solved the
> problem at least temporarily and made freeswitch stop getting back into the
> symptom described above.
> I can see that the cached memory is again growing over time. I still don't
> know if I'll end up with the same symptom again when the total used memory
> (used+ cached) gets near the 8GB.
>
> Anyone has had this symptom before too?
> Some feedback is appreciated.
>
> Thanks.
> --
>
> *Nuno Miguel Reis* | *Unified Communication Systems
> *
> M. +351 913907481 | nreis at wavecom.pt <mailto:nreis at wavecom.pt<nreis at wavecom.pt>>
>
> WAVECOM-Solu??es R?dio, S.A.
> Cacia Park | Rua do Progresso, Lote 15
> 3800-639 AVEIRO | Portugal
> T. +351 309 700 225 | F. +351 234 919 191
> *GPS** <
> http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a6144f0df88>
>  | www.wavecom.pt <http://www.wavecom.pt/> *
> * <http://www.wavecom.pt/>
> *
>
>  <http://www.wavecom.pt/pt/wavecom/premios.php>
>
>  <http://www.wavecom.pt/pt/mail_eventos.php>
>
>
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 cjbujold at accra.ca  Sat Aug 24 15:01:16 2013
From: cjbujold at accra.ca (cjb)
Date: Sat, 24 Aug 2013 04:01:16 -0700 (PDT)
Subject: [Freeswitch-users] make current mod-spandsl error - 'bool' unknown
	type
Message-ID: <1377342076207-7594259.post@n2.nabble.com>

How can I fix?  Using Ubuntu 12.04 and trying to do an update by using make
current.  The process starts and then I get this error:  

make[2]: Entering directory `/usr/src/freeswitch'
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:53:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/logging.h:85:1: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:57:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/queue.h:67:1: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:78:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/crc.h:65:1: error: unknown type
name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/crc.h:98:1: error: unknown type
name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:79:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/async.h:175:48: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/async.h:209:48: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:80:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:96:46: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:97:46: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:180:46: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:182:46: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:89:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/bell_r2_mf.h:181:69: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/bell_r2_mf.h:254:48: error:
unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:94:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v8.h:138:30: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/v8.h:150:36: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:95:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v42.h:89:38: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/v42.h:90:38: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:97:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v29rx.h:159:67: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:98:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v29tx.h:121:77: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:98:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v29tx.h:129:67: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:99:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v17rx.h:244:67: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:100:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v17tx.h:108:77: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:100:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v17tx.h:117:67: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/v17tx.h:117:77: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:104:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v22bis.h:160:61: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/v22bis.h:181:44: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:105:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v27ter_rx.h:79:73: error:
unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:106:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v27ter_tx.h:90:86: error:
unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:106:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v27ter_tx.h:98:73: error:
unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:110:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v18.h:121:38: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:113:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:364:70: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:123:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t30.h:193:47: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:124:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.h:367:73: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.h:430:53: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.hmake[2]: Entering
directory `/usr/src/freeswitch'
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:53:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/logging.h:85:1: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:57:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/queue.h:67:1: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:78:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/crc.h:65:1: error: unknown type
name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/crc.h:98:1: error: unknown type
name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:79:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/async.h:175:48: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/async.h:209:48: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:80:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:96:46: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:97:46: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:180:46: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:182:46: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:89:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/bell_r2_mf.h:181:69: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/bell_r2_mf.h:254:48: error:
unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:94:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v8.h:138:30: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/v8.h:150:36: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:95:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v42.h:89:38: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/v42.h:90:38: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:97:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v29rx.h:159:67: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:98:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v29tx.h:121:77: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:98:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v29tx.h:129:67: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:99:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v17rx.h:244:67: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:100:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v17tx.h:108:77: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:100:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v17tx.h:117:67: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/v17tx.h:117:77: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:104:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v22bis.h:160:61: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/v22bis.h:181:44: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:105:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v27ter_rx.h:79:73: error:
unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:106:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v27ter_tx.h:90:86: error:
unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:106:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v27ter_tx.h:98:73: error:
unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:110:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/v18.h:121:38: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:113:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:364:70: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:123:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t30.h:193:47: error: unknown
type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:124:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.h:367:73: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.h:430:53: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.h:437:58: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.h:544:68: error:
unknown type name ?t30_real_time_frame_handler_t?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:127:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t35.h:88:1: error: unknown type
name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:128:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:119:5: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:121:5: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:125:5: error:
unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:131:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:317:66: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:323:65: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:329:66: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:369:74: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:375:62: error:
unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:132:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_non_ecm_buffer.h:89:99:
error: unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:132:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_non_ecm_buffer.h:99:79:
error: unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:133:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:54:55: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:66:5: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:129:75: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:138:77: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:170:69: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:177:77: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:208:60: error:
unknown type name ?t38_gateway_real_time_frame_handler_t?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:134:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:77:71: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:84:79: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:115:40: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:125:56: error:
unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:96:61: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:103:53: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:111:55: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:118:49: error: unknown
type name ?bool?
make[2]: *** [freeswitch-switch.o] Error 1
make[2]: Leaving directory `/usr/src/freeswitch'
make[1]: *** [all] Error 2
make[1]: Leaving directory `/usr/src/freeswitch'
make: *** [current] Error 2
root at sip:/usr/src/freeswitch# 
:437:58: error: unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.h:544:68: error:
unknown type name ?t30_real_time_frame_handler_t?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:127:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t35.h:88:1: error: unknown type
name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:128:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:119:5: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:121:5: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:125:5: error:
unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:131:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:317:66: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:323:65: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:329:66: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:369:74: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:375:62: error:
unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:132:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_non_ecm_buffer.h:89:99:
error: unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:132:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_non_ecm_buffer.h:99:79:
error: unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:133:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:54:55: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:66:5: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:129:75: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:138:77: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:170:69: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:177:77: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:208:60: error:
unknown type name ?t38_gateway_real_time_frame_handler_t?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:134:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:77:71: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:84:79: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:115:40: error:
unknown type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:125:56: error:
unknown type name ?bool?
In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135:0,
                 from ./src/include/private/switch_core_pvt.h:35,
                 from src/switch.c:53:
/usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:96:61: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:103:53: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:111:55: error: unknown
type name ?bool?
/usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:118:49: error: unknown
type name ?bool?
make[2]: *** [freeswitch-switch.o] Error 1
make[2]: Leaving directory `/usr/src/freeswitch'
make[1]: *** [all] Error 2
make[1]: Leaving directory `/usr/src/freeswitch'
make: *** [current] Error 2
root at sip:/usr/src/freeswitch# 




--
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From steveu at coppice.org  Sat Aug 24 15:20:53 2013
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 24 Aug 2013 19:20:53 +0800
Subject: [Freeswitch-users] make current mod-spandsl error - 'bool'
 unknown type
In-Reply-To: <1377342076207-7594259.post@n2.nabble.com>
References: <1377342076207-7594259.post@n2.nabble.com>
Message-ID: <52189715.4090706@coppice.org>

My guess is you've done a "git pull" for an existing directory, and you 
haven't done a "make spandsp-reconf".

Regards,
Steve

On 08/24/2013 07:01 PM, cjb wrote:
> How can I fix?  Using Ubuntu 12.04 and trying to do an update by using make
> current.  The process starts and then I get this error:
>
> make[2]: Entering directory `/usr/src/freeswitch'
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:53:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/logging.h:85:1: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:57:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/queue.h:67:1: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:78:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/crc.h:65:1: error: unknown type
> name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/crc.h:98:1: error: unknown type
> name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:79:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/async.h:175:48: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/async.h:209:48: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:80:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:96:46: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:97:46: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:180:46: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:182:46: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:89:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/bell_r2_mf.h:181:69: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/bell_r2_mf.h:254:48: error:
> unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:94:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v8.h:138:30: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v8.h:150:36: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:95:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v42.h:89:38: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v42.h:90:38: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:97:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v29rx.h:159:67: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:98:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v29tx.h:121:77: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:98:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v29tx.h:129:67: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:99:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v17rx.h:244:67: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:100:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v17tx.h:108:77: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:100:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v17tx.h:117:67: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v17tx.h:117:77: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:104:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v22bis.h:160:61: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v22bis.h:181:44: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:105:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v27ter_rx.h:79:73: error:
> unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:106:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v27ter_tx.h:90:86: error:
> unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:106:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v27ter_tx.h:98:73: error:
> unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:110:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v18.h:121:38: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:113:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:364:70: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:123:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t30.h:193:47: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:124:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.h:367:73: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.h:430:53: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.hmake[2]: Entering
> directory `/usr/src/freeswitch'
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:53:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/logging.h:85:1: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:57:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/queue.h:67:1: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:78:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/crc.h:65:1: error: unknown type
> name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/crc.h:98:1: error: unknown type
> name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:79:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/async.h:175:48: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/async.h:209:48: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:80:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:96:46: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:97:46: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:180:46: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/hdlc.h:182:46: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:89:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/bell_r2_mf.h:181:69: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/bell_r2_mf.h:254:48: error:
> unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:94:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v8.h:138:30: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v8.h:150:36: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:95:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v42.h:89:38: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v42.h:90:38: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:97:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v29rx.h:159:67: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:98:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v29tx.h:121:77: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:98:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v29tx.h:129:67: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:99:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v17rx.h:244:67: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:100:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v17tx.h:108:77: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:100:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v17tx.h:117:67: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v17tx.h:117:77: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:104:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v22bis.h:160:61: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v22bis.h:181:44: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:105:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v27ter_rx.h:79:73: error:
> unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:106:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v27ter_tx.h:90:86: error:
> unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:106:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v27ter_tx.h:98:73: error:
> unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:110:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/v18.h:121:38: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:113:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:364:70: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:123:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t30.h:193:47: error: unknown
> type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:124:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.h:367:73: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.h:430:53: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.h:437:58: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.h:544:68: error:
> unknown type name ?t30_real_time_frame_handler_t?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:127:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t35.h:88:1: error: unknown type
> name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:128:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:119:5: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:121:5: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:125:5: error:
> unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:131:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:317:66: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:323:65: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:329:66: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:369:74: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:375:62: error:
> unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:132:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_non_ecm_buffer.h:89:99:
> error: unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:132:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_non_ecm_buffer.h:99:79:
> error: unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:133:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:54:55: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:66:5: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:129:75: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:138:77: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:170:69: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:177:77: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:208:60: error:
> unknown type name ?t38_gateway_real_time_frame_handler_t?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:134:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:77:71: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:84:79: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:115:40: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:125:56: error:
> unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:96:61: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:103:53: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:111:55: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:118:49: error: unknown
> type name ?bool?
> make[2]: *** [freeswitch-switch.o] Error 1
> make[2]: Leaving directory `/usr/src/freeswitch'
> make[1]: *** [all] Error 2
> make[1]: Leaving directory `/usr/src/freeswitch'
> make: *** [current] Error 2
> root at sip:/usr/src/freeswitch#
> :437:58: error: unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t30_api.h:544:68: error:
> unknown type name ?t30_real_time_frame_handler_t?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:127:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t35.h:88:1: error: unknown type
> name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:128:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:119:5: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:121:5: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/at_interpreter.h:125:5: error:
> unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:131:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:317:66: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:323:65: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:329:66: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:369:74: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_core.h:375:62: error:
> unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:132:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_non_ecm_buffer.h:89:99:
> error: unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:132:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_non_ecm_buffer.h:99:79:
> error: unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:133:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:54:55: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:66:5: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:129:75: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:138:77: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:170:69: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:177:77: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_gateway.h:208:60: error:
> unknown type name ?t38_gateway_real_time_frame_handler_t?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:134:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:77:71: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:84:79: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:115:40: error:
> unknown type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t38_terminal.h:125:56: error:
> unknown type name ?bool?
> In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135:0,
>                   from ./src/include/private/switch_core_pvt.h:35,
>                   from src/switch.c:53:
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:96:61: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:103:53: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:111:55: error: unknown
> type name ?bool?
> /usr/src/freeswitch/libs/spandsp/src/spandsp/t31.h:118:49: error: unknown
> type name ?bool?
> make[2]: *** [freeswitch-switch.o] Error 1
> make[2]: Leaving directory `/usr/src/freeswitch'
> make[1]: *** [all] Error 2
> make[1]: Leaving directory `/usr/src/freeswitch'
> make: *** [current] Error 2
> root at sip:/usr/src/freeswitch#
>



From babak.freeswitch at gmail.com  Sat Aug 24 18:02:14 2013
From: babak.freeswitch at gmail.com (Babak Yakhchali)
Date: Sat, 24 Aug 2013 18:32:14 +0430
Subject: [Freeswitch-users] freeswitch 1.2.12 build fails on centos 6.2 32bit
Message-ID: <CAEVfcaYHR88=sJPCHtm+kpKPvd6qk56rdxUc8JJcmxpr+XqrOg@mail.gmail.com>

Hi
I'm trying to build freeswitch 1.2.12 in centos 6.2 but I get:

 gcc -I/usr/local/src/freeswitch-1.2.12/libs/curl/include
-I/usr/local/src/freeswitch-1.2.12/src/include
-I/usr/local/src/freeswitch-1.2.12/src/include
-I/usr/local/src/freeswitch-1.2.12/libs/libteletone/src
-I/usr/local/src/freeswitch-1.2.12/libs/stfu -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g
-O2 -o .libs/freeswitch freeswitch-switch.o -Wl,-lodbc -Wl,-z -Wl,relro
 -lm ./.libs/libfreeswitch.so
-L/usr/local/src/freeswitch-1.2.12/libs/apr-util/xml/expat/lib -lpq
/usr/local/src/freeswitch-1.2.12/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/usr/local/src/freeswitch-1.2.12/libs/apr/.libs/libapr-1.a
-L/usr/local/src/freeswitch-1.2.12/libs/srtp libs/apr/.libs/libapr-1.a -lrt
-lcrypt -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -ldl -lz
-lncurses -ljpeg -lodbc  -Wl,--rpath -Wl,/usr/local/freeswitch/lib
/usr/lib/libnss3.so: undefined reference to `PR_FindSymbol'
/usr/lib/libnss3.so: undefined reference to `PR_RWLock_Rlock'
/usr/lib/libssl3.so: undefined reference to `PR_OpenAnonFileMap'
/usr/lib/libssl3.so: undefined reference to `PR_UnloadLibrary'
/usr/lib/libnss3.so: undefined reference to `PL_InitArenaPool'
/usr/lib/libnss3.so: undefined reference to `PR_NewRWLock'
/usr/lib/libnss3.so: undefined reference to `PR_RWLock_Wlock'
/usr/lib/libnss3.so: undefined reference to `PR_LoadLibrary'
/lib/libldap_r-2.4.so.2: undefined reference to `PR_GetEnv'
/usr/lib/libssl3.so: undefined reference to `PR_LoadLibraryWithFlags'
/usr/lib/libnssutil3.so: undefined reference to `PL_ClearArenaPool'
/usr/lib/libnss3.so: undefined reference to `PR_DestroyRWLock'
/usr/lib/libnss3.so: undefined reference to `PR_NewTCPSocket'
/lib/libldap_r-2.4.so.2: undefined reference to `PR_SetEnv'
/lib/libldap_r-2.4.so.2: undefined reference to `PR_GetLibraryName'
/usr/lib/libssl3.so: undefined reference to `PR_ErrorInstallTable'
/usr/lib/libssl3.so: undefined reference to `PR_ExportFileMapAsString'
/usr/lib/libssl3.so: undefined reference to `PR_GetLibraryFilePathname'
/usr/lib/libssl3.so: undefined reference to `PR_FindFunctionSymbol'
/usr/lib/libsmime3.so: undefined reference to `PL_NewHashTable'
/lib/libldap_r-2.4.so.2: undefined reference to `PR_ErrorToString'
/usr/lib/libnss3.so: undefined reference to `PR_RWLock_Unlock'
/usr/lib/libssl3.so: undefined reference to `PR_ImportFileMapFromString'
/lib/libldap_r-2.4.so.2: undefined reference to `PR_GetDirectorySeparator'
collect2: ld returned 1 exit status

I tried ./configure --without-libcurl --without-pgsql but no success
thanks
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From krice at freeswitch.org  Sat Aug 24 18:56:37 2013
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 24 Aug 2013 09:56:37 -0500
Subject: [Freeswitch-users] Cluecon presentations - archived / available
	?
In-Reply-To: <CAHDL-swWULXmUVghAN7xEvF=1Qao48F=xOuyfmbh-qfemQ+UiQ@mail.gmail.com>
Message-ID: <CE3E33D5.7938D%krice@freeswitch.org>

The 2013 videos are being loaded to youtube... There will be links published
shortly via cluecon.com with notices going out via social media and the
mailing lists


On 8/23/13 11:42 PM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:

> Many thanks. Wonder if other list members are aware of Cluecon 2013 decks...
> there are 2-3 I am quite interested in seeing.
> 
> Just as a hint to organizers... if recorded videos of sessions with
> presentations are made available for reasonable fee per session, I think many
> of us might lap it up !
> 
> 
> On Sat, Aug 24, 2013 at 8:29 AM, William King <william.king at quentustech.com>
> wrote:
>> http://torrents.freeswitch.org/
>> 
>> At least the 2011 ClueCon torrent is there.
>> 
>> William King
>> Senior Engineer
>> Quentus Technologies, INC
>> 1037 NE 65th St Suite 273
>> Seattle, WA 98115
>> Main: ? (877) 211-9337
>> Office: (206) 388-4772
>> Cell: ? (253) 686-5518
>> william.king at quentustech.com
>> 
>> On 08/23/2013 07:49 PM, Jayanth Acharya wrote:
>>> > Not sure if this is the right list for asking this question, but I
>>> > wasn't aware of a better one.
>>> >
>>> > Are the presentations done in past Cluecons archived and available for
>>> > download from some location ? Scribd / Slideshare etc. ?
>>> >
>>> >
>>> > _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.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
>>> >
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From mehroz.ashraf85 at gmail.com  Sat Aug 24 19:08:13 2013
From: mehroz.ashraf85 at gmail.com (Mehroz Ashraf)
Date: Sat, 24 Aug 2013 20:08:13 +0500
Subject: [Freeswitch-users] Early media with proxy mode
In-Reply-To: <CAKbxfG-tBxAJLfJxUfeFA9Dkk+7wUb0BWXDbhs_HS512iQmQbg@mail.gmail.com>
References: <1377270826705-7594240.post@n2.nabble.com>
	<CAKbxfG-tBxAJLfJxUfeFA9Dkk+7wUb0BWXDbhs_HS512iQmQbg@mail.gmail.com>
Message-ID: <CAC0j6rdH6Hn3j_PYtVrO=TiVY_R8puXfWhsO+BjDxxHUaRKPXw@mail.gmail.com>

Thanks Anthony,

I meant Ring back by RBT.  I understand proxy media cannot process media
and manage media streams.
Is it possible to initiate with default mode , send ring with early media
and then took media into proxy after bridging.
OR, only 180 Ringing Message is sent and client itself has to generate
ringing?


On Sat, Aug 24, 2013 at 4:58 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I don't know what RBT means but you can't send any media from FS itself if
> you want to do successful proxy media.
> Attempting to send media results in a sdp being generated and we do not
> support transitioning from regular media to proxy media once the call has
> reached that state.
>
>
>
>
> On Fri, Aug 23, 2013 at 10:13 AM, mehroz <mehroz.ashraf85 at gmail.com>wrote:
>
>> Hi,
>>
>>
>> Is it somehow possible to send RBT as early media with proxy media mode?
>> I have only observed early media in default media mode. I am with ZRTP,
>> thus
>> cannot switch to default mode.
>>
>> Please advice!
>>
>>
>>
>>
>> --
>> View this message in context:
>> http://freeswitch-users.2379917.n2.nabble.com/Early-media-with-proxy-mode-tp7594240.html
>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 cal.leeming at simplicitymedialtd.co.uk  Sat Aug 24 20:15:03 2013
From: cal.leeming at simplicitymedialtd.co.uk (Cal Leeming [Simplicity Media Ltd])
Date: Sat, 24 Aug 2013 17:15:03 +0100
Subject: [Freeswitch-users] Cluecon presentations - archived / available
	?
In-Reply-To: <CE3E33D5.7938D%krice@freeswitch.org>
References: <CAHDL-swWULXmUVghAN7xEvF=1Qao48F=xOuyfmbh-qfemQ+UiQ@mail.gmail.com>
	<CE3E33D5.7938D%krice@freeswitch.org>
Message-ID: <CAHKQagEY9XjbEv5=p8=XpAH-kTsYt4vUwvDhUGKC7W93MuXk4A@mail.gmail.com>

Thanks Ken!

Cal

On Sat, Aug 24, 2013 at 3:56 PM, Ken Rice <krice at freeswitch.org> wrote:

>  The 2013 videos are being loaded to youtube... There will be links
> published shortly via cluecon.com with notices going out via social media
> and the mailing lists
>
>
>
> On 8/23/13 11:42 PM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:
>
> Many thanks. Wonder if other list members are aware of Cluecon 2013
> decks... there are 2-3 I am quite interested in seeing.
>
> Just as a hint to organizers... if recorded videos of sessions with
> presentations are made available for reasonable fee per session, I think
> many of us might lap it up !
>
>
> On Sat, Aug 24, 2013 at 8:29 AM, William King <
> william.king at quentustech.com> wrote:
>
> http://torrents.freeswitch.org/
>
> At least the 2011 ClueCon torrent is there.
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
> On 08/23/2013 07:49 PM, Jayanth Acharya wrote:
> > Not sure if this is the right list for asking this question, but I
> > wasn't aware of a better one.
> >
> > Are the presentations done in past Cluecons archived and available for
> > download from some location ? Scribd / Slideshare etc. ?
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Sat Aug 24 20:15:56 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 24 Aug 2013 11:15:56 -0500
Subject: [Freeswitch-users] Early media with proxy mode
In-Reply-To: <CAC0j6rdH6Hn3j_PYtVrO=TiVY_R8puXfWhsO+BjDxxHUaRKPXw@mail.gmail.com>
References: <1377270826705-7594240.post@n2.nabble.com>
	<CAKbxfG-tBxAJLfJxUfeFA9Dkk+7wUb0BWXDbhs_HS512iQmQbg@mail.gmail.com>
	<CAC0j6rdH6Hn3j_PYtVrO=TiVY_R8puXfWhsO+BjDxxHUaRKPXw@mail.gmail.com>
Message-ID: <CAKbxfG_W_XeyKR8gQ4tw39Xg3=z8YS8Os-J3+sDygtf1Nr=w0w@mail.gmail.com>

Run the ring_ready app to send 180 that is all you can do.  You can never
send media if you intend to use proxy media.
On Aug 24, 2013 10:29 AM, "Mehroz Ashraf" <mehroz.ashraf85 at gmail.com> wrote:

> Thanks Anthony,
>
> I meant Ring back by RBT.  I understand proxy media cannot process media
> and manage media streams.
> Is it possible to initiate with default mode , send ring with early media
> and then took media into proxy after bridging.
> OR, only 180 Ringing Message is sent and client itself has to generate
> ringing?
>
>
> On Sat, Aug 24, 2013 at 4:58 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> I don't know what RBT means but you can't send any media from FS itself
>> if you want to do successful proxy media.
>> Attempting to send media results in a sdp being generated and we do not
>> support transitioning from regular media to proxy media once the call has
>> reached that state.
>>
>>
>>
>>
>> On Fri, Aug 23, 2013 at 10:13 AM, mehroz <mehroz.ashraf85 at gmail.com>wrote:
>>
>>> Hi,
>>>
>>>
>>> Is it somehow possible to send RBT as early media with proxy media mode?
>>> I have only observed early media in default media mode. I am with ZRTP,
>>> thus
>>> cannot switch to default mode.
>>>
>>> Please advice!
>>>
>>>
>>>
>>>
>>> --
>>> View this message in context:
>>> http://freeswitch-users.2379917.n2.nabble.com/Early-media-with-proxy-mode-tp7594240.html
>>> Sent from the freeswitch-users mailing list archive at Nabble.com.
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Sat Aug 24 22:09:37 2013
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 24 Aug 2013 11:09:37 -0700
Subject: [Freeswitch-users] Avoiding download of sounds/moh files (reuse
 from another build)
In-Reply-To: <CAHDL-syuUiDPUST1tn5FYR8MWfU1+Dbm7usjwtXPyRtYGNGt3g@mail.gmail.com>
References: <CAHDL-syuUiDPUST1tn5FYR8MWfU1+Dbm7usjwtXPyRtYGNGt3g@mail.gmail.com>
Message-ID: <CAKzWOxWRhZJe1O3bdr=cRJH=MS7NDaJOUJXtNhYQMzP5KaXZow@mail.gmail.com>

Try copying the freeswitch-sounds*gz files from the old directory into the
new one. If the installer sees that the files are already there it won't
try and re-download them.

-MC


On Fri, Aug 23, 2013 at 7:47 AM, Jayanth Acharya <jayachar88 at gmail.com>wrote:

> Hi,
>
> Already cloned Git HEAD (1.5b.5) a day back, and had successfully build FS.
>
> I wish to try latest stable, due to some issues I am facing with the Git
> HEAD version.
>
> The longest download for me were the sound/MOH files triggered as a result
> of :
>
> make sounds-install
> make moh-install
>
> Anyway to reuse the version of the already downloaded sound/MOH files
> while rebuilding stable ?
>
> thanks,
>
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From jayachar88 at gmail.com  Sat Aug 24 22:29:35 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Sat, 24 Aug 2013 23:59:35 +0530
Subject: [Freeswitch-users] mod_gsmopen: some success - hear crackly
	unintelligible announcement
Message-ID: <CAHDL-szwednREpc+ZpTvVXRTRrudozbTjhhmYd-oaj4iLyOgXg@mail.gmail.com>

Finally, found success with mod_gsmopen. Have to say that there's a tonne
of information in the Wiki, and it takes a newbie some time to get to
understand even the elementary things. Jumping right in, assuming defaults
to take care of everything, doesn't work. I believe that was thanks to
being connected to the conference bridge 888 at sip.freeswitch.net (if I noted
the SIP URI correctly).Next tried another inbound call, and this time, I
heard only silence.

Then I tried sending an SMS. That went out fine and was received
instantaneously by the destination number. After that, I replied back to
the dongle's number, and had a telnet session open listening on the events
(none arrived), but that telnet session was closed, and turning to the FS
console, I saw that it too had closed with a segmentation fault. So seems
like receiving the SMS precipitated a core.

Anyone seen similar behaviour ? Should I go ahead an file another JIRA
bugreport ? File one yesterday regarding another segmentation fault. Note,
I am on Git HEAD (or was, day before, when I cloned it).

thanks,
Jay
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From jayachar88 at gmail.com  Sat Aug 24 22:31:06 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Sun, 25 Aug 2013 00:01:06 +0530
Subject: [Freeswitch-users] Avoiding download of sounds/moh files (reuse
 from another build)
In-Reply-To: <CAKzWOxWRhZJe1O3bdr=cRJH=MS7NDaJOUJXtNhYQMzP5KaXZow@mail.gmail.com>
References: <CAHDL-syuUiDPUST1tn5FYR8MWfU1+Dbm7usjwtXPyRtYGNGt3g@mail.gmail.com>
	<CAKzWOxWRhZJe1O3bdr=cRJH=MS7NDaJOUJXtNhYQMzP5KaXZow@mail.gmail.com>
Message-ID: <CAHDL-sxJFGwVQT8+Oiv=GaD4rSCe=MGms+6JajkNm4qNZEvXXQ@mail.gmail.com>

Okay - thanks Michael. Will try that. Getting a bit bugged by the
segfaults, so need to try the stable version.


On Sat, Aug 24, 2013 at 11:39 PM, Michael Collins <msc at freeswitch.org>wrote:

> Try copying the freeswitch-sounds*gz files from the old directory into the
> new one. If the installer sees that the files are already there it won't
> try and re-download them.
>
> -MC
>
>
> On Fri, Aug 23, 2013 at 7:47 AM, Jayanth Acharya <jayachar88 at gmail.com>wrote:
>
>> Hi,
>>
>> Already cloned Git HEAD (1.5b.5) a day back, and had successfully build
>> FS.
>>
>> I wish to try latest stable, due to some issues I am facing with the Git
>> HEAD version.
>>
>> The longest download for me were the sound/MOH files triggered as a
>> result of :
>>
>> make sounds-install
>> make moh-install
>>
>> Anyway to reuse the version of the already downloaded sound/MOH files
>> while rebuilding stable ?
>>
>> thanks,
>>
>> Jay
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 intralanman at freeswitch.org  Sat Aug 24 22:53:31 2013
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Sat, 24 Aug 2013 14:53:31 -0400
Subject: [Freeswitch-users] mod_gsmopen: some success - hear crackly
 unintelligible announcement
In-Reply-To: <CAHDL-szwednREpc+ZpTvVXRTRrudozbTjhhmYd-oaj4iLyOgXg@mail.gmail.com>
References: <CAHDL-szwednREpc+ZpTvVXRTRrudozbTjhhmYd-oaj4iLyOgXg@mail.gmail.com>
Message-ID: <CAB+Kzh5rxjkwcd-cFb9cEOzfmsU7jH9oHBcZf+Gh6=5WUyRK1w@mail.gmail.com>

Any time FreeSWITCH segfaults, you should file a JIRA.  It would be nice if
you could look at the backtrace and see if it happens in the same place or
not, but if you're unable to read it, nobody will get mad about you filing
a duplicate.  It will just be noted as such and linked to the other.  JIRAs
are free, so when in doubt, fill it out :-)

-Ray
On Aug 24, 2013 2:33 PM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:

> Finally, found success with mod_gsmopen. Have to say that there's a tonne
> of information in the Wiki, and it takes a newbie some time to get to
> understand even the elementary things. Jumping right in, assuming defaults
> to take care of everything, doesn't work. I believe that was thanks to
> being connected to the conference bridge 888 at sip.freeswitch.net (if I
> noted the SIP URI correctly).Next tried another inbound call, and this
> time, I heard only silence.
>
> Then I tried sending an SMS. That went out fine and was received
> instantaneously by the destination number. After that, I replied back to
> the dongle's number, and had a telnet session open listening on the events
> (none arrived), but that telnet session was closed, and turning to the FS
> console, I saw that it too had closed with a segmentation fault. So seems
> like receiving the SMS precipitated a core.
>
> Anyone seen similar behaviour ? Should I go ahead an file another JIRA
> bugreport ? File one yesterday regarding another segmentation fault. Note,
> I am on Git HEAD (or was, day before, when I cloned it).
>
> thanks,
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Sat Aug 24 23:01:19 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 25 Aug 2013 00:31:19 +0530
Subject: [Freeswitch-users] mod_gsmopen: some success - hear crackly
 unintelligible announcement
In-Reply-To: <CAHDL-szwednREpc+ZpTvVXRTRrudozbTjhhmYd-oaj4iLyOgXg@mail.gmail.com>
References: <CAHDL-szwednREpc+ZpTvVXRTRrudozbTjhhmYd-oaj4iLyOgXg@mail.gmail.com>
Message-ID: <CAMvhjJty+9hV75Wu=iQcs2mHhfNQ4qB9QaPN3z4Kr=X-5+mubQ@mail.gmail.com>

Hi jaynath
May i ask you what what was was the dialplan which you had used to call
<any indian phone number> to <any indian phone number> in your mod_gsmopen.
Also what did you do with error u received day before yesterday....???

Regards
Ashish Mishra
On Aug 25, 2013 12:05 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:

> Finally, found success with mod_gsmopen. Have to say that there's a tonne
> of information in the Wiki, and it takes a newbie some time to get to
> understand even the elementary things. Jumping right in, assuming defaults
> to take care of everything, doesn't work. I believe that was thanks to
> being connected to the conference bridge 888 at sip.freeswitch.net (if I
> noted the SIP URI correctly).Next tried another inbound call, and this
> time, I heard only silence.
>
> Then I tried sending an SMS. That went out fine and was received
> instantaneously by the destination number. After that, I replied back to
> the dongle's number, and had a telnet session open listening on the events
> (none arrived), but that telnet session was closed, and turning to the FS
> console, I saw that it too had closed with a segmentation fault. So seems
> like receiving the SMS precipitated a core.
>
> Anyone seen similar behaviour ? Should I go ahead an file another JIRA
> bugreport ? File one yesterday regarding another segmentation fault. Note,
> I am on Git HEAD (or was, day before, when I cloned it).
>
> thanks,
> Jay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Sat Aug 24 23:46:59 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Sun, 25 Aug 2013 01:16:59 +0530
Subject: [Freeswitch-users] mod_gsmopen: some success - hear crackly
 unintelligible announcement
In-Reply-To: <CAMvhjJty+9hV75Wu=iQcs2mHhfNQ4qB9QaPN3z4Kr=X-5+mubQ@mail.gmail.com>
References: <CAHDL-szwednREpc+ZpTvVXRTRrudozbTjhhmYd-oaj4iLyOgXg@mail.gmail.com>
	<CAMvhjJty+9hV75Wu=iQcs2mHhfNQ4qB9QaPN3z4Kr=X-5+mubQ@mail.gmail.com>
Message-ID: <CAHDL-sxnDwO53sRkg0wPrf2QFbyPu1hWE3bGvWmPS9+x+GxxUA@mail.gmail.com>

Using the default dialplan. I have only 1 dongle at the moment, so doing a
telephone -> telephone mapping won't work (well, at least, based on what
I've understood so far -- I could be wrong). Tomorrow, I plan to walk
through the dialplan to understand it somewhat better. My eyes don't take
XML too kindly !!

As such no significant changes. The only few hiccups to build, and the
previous segfault I got, I filed a JIRA for, and that has the steps I took
and my preliminary findings.
http://jira.freeswitch.org/browse/FS-5721
http://jira.freeswitch.org/browse/FS-5727

I used a trick I found in the mail archives to use "wvdial" to identify the
control port. After that, set the usb ports for control/audio correctly in
the gsmopen.conf.xml file. After this the module started getting loaded
properly, but I noticed that the signal strength was too low. Post that, I
had to use a long USB extension cable to keep the dongle outside the
window, and ensured that signal strength (RSSI) was good enough. After
this, the rig started working.



On Sun, Aug 25, 2013 at 12:31 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Hi jaynath
> May i ask you what what was was the dialplan which you had used to call
> <any indian phone number> to <any indian phone number> in your mod_gsmopen.
> Also what did you do with error u received day before yesterday....???
>
> Regards
> Ashish Mishra
> On Aug 25, 2013 12:05 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:
>
>> Finally, found success with mod_gsmopen. Have to say that there's a tonne
>> of information in the Wiki, and it takes a newbie some time to get to
>> understand even the elementary things. Jumping right in, assuming defaults
>> to take care of everything, doesn't work. I believe that was thanks to
>> being connected to the conference bridge 888 at sip.freeswitch.net (if I
>> noted the SIP URI correctly).Next tried another inbound call, and this
>> time, I heard only silence.
>>
>> Then I tried sending an SMS. That went out fine and was received
>> instantaneously by the destination number. After that, I replied back to
>> the dongle's number, and had a telnet session open listening on the events
>> (none arrived), but that telnet session was closed, and turning to the FS
>> console, I saw that it too had closed with a segmentation fault. So seems
>> like receiving the SMS precipitated a core.
>>
>> Anyone seen similar behaviour ? Should I go ahead an file another JIRA
>> bugreport ? File one yesterday regarding another segmentation fault. Note,
>> I am on Git HEAD (or was, day before, when I cloned it).
>>
>> thanks,
>> Jay
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 at cassidywebservices.co.uk  Sun Aug 25 00:01:34 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sat, 24 Aug 2013 21:01:34 +0100
Subject: [Freeswitch-users] Pacemaker Resource Agent
Message-ID: <CAMbW9oJ3F_K3nMz0qKj9KFfYq_ZLB-buLkYAOu9s6xNooq6f1Q@mail.gmail.com>

Hi all,

I know this has come up before but I can't find a pacemaker resource agent
for FreeSWITCH. I'm running a master/slave config and wondering if anyone
has something they're willing to share?

I know there's mod_ha_cluster, however I already have corosync/pacemaker
for other services, and would like to keep things uniform.

-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From itsme.kunnu at gmail.com  Sun Aug 25 00:32:27 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 25 Aug 2013 02:02:27 +0530
Subject: [Freeswitch-users] mod_gsmopen: some success - hear crackly
 unintelligible announcement
In-Reply-To: <CAHDL-sxnDwO53sRkg0wPrf2QFbyPu1hWE3bGvWmPS9+x+GxxUA@mail.gmail.com>
References: <CAHDL-szwednREpc+ZpTvVXRTRrudozbTjhhmYd-oaj4iLyOgXg@mail.gmail.com>
	<CAMvhjJty+9hV75Wu=iQcs2mHhfNQ4qB9QaPN3z4Kr=X-5+mubQ@mail.gmail.com>
	<CAHDL-sxnDwO53sRkg0wPrf2QFbyPu1hWE3bGvWmPS9+x+GxxUA@mail.gmail.com>
Message-ID: <CAMvhjJuxmeUdgZBr3ZrxPBaNQO=Q=2ntkRoXsxEbuEnoaK5NcA@mail.gmail.com>

Thanks jayanath
But i mean how to enable the sim u had used in the dongle for making calls
through mod_gsmopen make and receive calls ?

Regards
Ashish Mishra
On Aug 25, 2013 1:21 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:

> Using the default dialplan. I have only 1 dongle at the moment, so doing a
> telephone -> telephone mapping won't work (well, at least, based on what
> I've understood so far -- I could be wrong). Tomorrow, I plan to walk
> through the dialplan to understand it somewhat better. My eyes don't take
> XML too kindly !!
>
> As such no significant changes. The only few hiccups to build, and the
> previous segfault I got, I filed a JIRA for, and that has the steps I took
> and my preliminary findings.
> http://jira.freeswitch.org/browse/FS-5721
> http://jira.freeswitch.org/browse/FS-5727
>
> I used a trick I found in the mail archives to use "wvdial" to identify
> the control port. After that, set the usb ports for control/audio correctly
> in the gsmopen.conf.xml file. After this the module started getting loaded
> properly, but I noticed that the signal strength was too low. Post that, I
> had to use a long USB extension cable to keep the dongle outside the
> window, and ensured that signal strength (RSSI) was good enough. After
> this, the rig started working.
>
>
>
> On Sun, Aug 25, 2013 at 12:31 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Hi jaynath
>> May i ask you what what was was the dialplan which you had used to call
>> <any indian phone number> to <any indian phone number> in your mod_gsmopen.
>> Also what did you do with error u received day before yesterday....???
>>
>> Regards
>> Ashish Mishra
>> On Aug 25, 2013 12:05 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:
>>
>>> Finally, found success with mod_gsmopen. Have to say that there's a
>>> tonne of information in the Wiki, and it takes a newbie some time to get to
>>> understand even the elementary things. Jumping right in, assuming defaults
>>> to take care of everything, doesn't work. I believe that was thanks to
>>> being connected to the conference bridge 888 at sip.freeswitch.net (if I
>>> noted the SIP URI correctly).Next tried another inbound call, and this
>>> time, I heard only silence.
>>>
>>> Then I tried sending an SMS. That went out fine and was received
>>> instantaneously by the destination number. After that, I replied back to
>>> the dongle's number, and had a telnet session open listening on the events
>>> (none arrived), but that telnet session was closed, and turning to the FS
>>> console, I saw that it too had closed with a segmentation fault. So seems
>>> like receiving the SMS precipitated a core.
>>>
>>> Anyone seen similar behaviour ? Should I go ahead an file another JIRA
>>> bugreport ? File one yesterday regarding another segmentation fault. Note,
>>> I am on Git HEAD (or was, day before, when I cloned it).
>>>
>>> thanks,
>>> Jay
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Sun Aug 25 00:36:50 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 25 Aug 2013 02:06:50 +0530
Subject: [Freeswitch-users] mod_gsmopen: some success - hear crackly
 unintelligible announcement
In-Reply-To: <CAMvhjJuxmeUdgZBr3ZrxPBaNQO=Q=2ntkRoXsxEbuEnoaK5NcA@mail.gmail.com>
References: <CAHDL-szwednREpc+ZpTvVXRTRrudozbTjhhmYd-oaj4iLyOgXg@mail.gmail.com>
	<CAMvhjJty+9hV75Wu=iQcs2mHhfNQ4qB9QaPN3z4Kr=X-5+mubQ@mail.gmail.com>
	<CAHDL-sxnDwO53sRkg0wPrf2QFbyPu1hWE3bGvWmPS9+x+GxxUA@mail.gmail.com>
	<CAMvhjJuxmeUdgZBr3ZrxPBaNQO=Q=2ntkRoXsxEbuEnoaK5NcA@mail.gmail.com>
Message-ID: <CAMvhjJtqagMtoevO6MnO5om=pHsmdfwYvxMOV5tmr3suidRb9w@mail.gmail.com>

I hope my question was clear...i meant that how should i enable the sim
,that i had put in dongle, make and receive calls through freeswitch.??
On Aug 25, 2013 2:02 AM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:

> Thanks jayanath
> But i mean how to enable the sim u had used in the dongle for making calls
> through mod_gsmopen make and receive calls ?
>
> Regards
> Ashish Mishra
> On Aug 25, 2013 1:21 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:
>
>> Using the default dialplan. I have only 1 dongle at the moment, so doing
>> a telephone -> telephone mapping won't work (well, at least, based on what
>> I've understood so far -- I could be wrong). Tomorrow, I plan to walk
>> through the dialplan to understand it somewhat better. My eyes don't take
>> XML too kindly !!
>>
>> As such no significant changes. The only few hiccups to build, and the
>> previous segfault I got, I filed a JIRA for, and that has the steps I took
>> and my preliminary findings.
>> http://jira.freeswitch.org/browse/FS-5721
>> http://jira.freeswitch.org/browse/FS-5727
>>
>> I used a trick I found in the mail archives to use "wvdial" to identify
>> the control port. After that, set the usb ports for control/audio correctly
>> in the gsmopen.conf.xml file. After this the module started getting loaded
>> properly, but I noticed that the signal strength was too low. Post that, I
>> had to use a long USB extension cable to keep the dongle outside the
>> window, and ensured that signal strength (RSSI) was good enough. After
>> this, the rig started working.
>>
>>
>>
>> On Sun, Aug 25, 2013 at 12:31 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> Hi jaynath
>>> May i ask you what what was was the dialplan which you had used to call
>>> <any indian phone number> to <any indian phone number> in your mod_gsmopen.
>>> Also what did you do with error u received day before yesterday....???
>>>
>>> Regards
>>> Ashish Mishra
>>> On Aug 25, 2013 12:05 AM, "Jayanth Acharya" <jayachar88 at gmail.com>
>>> wrote:
>>>
>>>> Finally, found success with mod_gsmopen. Have to say that there's a
>>>> tonne of information in the Wiki, and it takes a newbie some time to get to
>>>> understand even the elementary things. Jumping right in, assuming defaults
>>>> to take care of everything, doesn't work. I believe that was thanks to
>>>> being connected to the conference bridge 888 at sip.freeswitch.net (if I
>>>> noted the SIP URI correctly).Next tried another inbound call, and this
>>>> time, I heard only silence.
>>>>
>>>> Then I tried sending an SMS. That went out fine and was received
>>>> instantaneously by the destination number. After that, I replied back to
>>>> the dongle's number, and had a telnet session open listening on the events
>>>> (none arrived), but that telnet session was closed, and turning to the FS
>>>> console, I saw that it too had closed with a segmentation fault. So seems
>>>> like receiving the SMS precipitated a core.
>>>>
>>>> Anyone seen similar behaviour ? Should I go ahead an file another JIRA
>>>> bugreport ? File one yesterday regarding another segmentation fault. Note,
>>>> I am on Git HEAD (or was, day before, when I cloned it).
>>>>
>>>> thanks,
>>>> Jay
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 avi at avimarcus.net  Sun Aug 25 02:00:40 2013
From: avi at avimarcus.net (Avi Marcus)
Date: Sat, 24 Aug 2013 22:00:40 +0000
Subject: [Freeswitch-users] Pacemaker Resource Agent
In-Reply-To: <CAMbW9oJ3F_K3nMz0qKj9KFfYq_ZLB-buLkYAOu9s6xNooq6f1Q@mail.gmail.com>
References: <CAMbW9oJ3F_K3nMz0qKj9KFfYq_ZLB-buLkYAOu9s6xNooq6f1Q@mail.gmail.com>
Message-ID: <00000140b257ea79-cfb47be2-f8c6-4466-86a0-c6bde0cc4ec9-000000@email.amazonses.com>

Did you look at
http://wiki.freeswitch.org/wiki/Enterprise_deployment_IP_Failover and
http://wiki.freeswitch.org/wiki/Enterprise_deployment_with_Corosync?

-Avi


On Sat, Aug 24, 2013 at 11:01 PM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> Hi all,
>
> I know this has come up before but I can't find a pacemaker resource agent
> for FreeSWITCH. I'm running a master/slave config and wondering if anyone
> has something they're willing to share?
>
> I know there's mod_ha_cluster, however I already have corosync/pacemaker
> for other services, and would like to keep things uniform.
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
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From andrew at cassidywebservices.co.uk  Sun Aug 25 03:31:42 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sun, 25 Aug 2013 00:31:42 +0100
Subject: [Freeswitch-users] Pacemaker Resource Agent
In-Reply-To: <00000140b257ea79-cfb47be2-f8c6-4466-86a0-c6bde0cc4ec9-000000@email.amazonses.com>
References: <CAMbW9oJ3F_K3nMz0qKj9KFfYq_ZLB-buLkYAOu9s6xNooq6f1Q@mail.gmail.com>
	<00000140b257ea79-cfb47be2-f8c6-4466-86a0-c6bde0cc4ec9-000000@email.amazonses.com>
Message-ID: <CAMbW9o+hQYHDNVCtTbB-3RSDBSDU8FA-9+SAf3Vs6qbvYA-EMw@mail.gmail.com>

I did not as for some reason Google does not deem the relevant to the term
"freeswitch pacemaker resource agent"

Thanks Avi.


On 24 August 2013 23:00, Avi Marcus <avi at avimarcus.net> wrote:

> Did you look at
> http://wiki.freeswitch.org/wiki/Enterprise_deployment_IP_Failover and
> http://wiki.freeswitch.org/wiki/Enterprise_deployment_with_Corosync?
>
> -Avi
>
>
> On Sat, Aug 24, 2013 at 11:01 PM, Andrew Cassidy <
> andrew at cassidywebservices.co.uk> wrote:
>
>> Hi all,
>>
>> I know this has come up before but I can't find a pacemaker resource
>> agent for FreeSWITCH. I'm running a master/slave config and wondering if
>> anyone has something they're willing to share?
>>
>> I know there's mod_ha_cluster, however I already have corosync/pacemaker
>> for other services, and would like to keep things uniform.
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>>
>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>  *03300 100 961
>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From ssinyagin at yahoo.com  Sun Aug 25 04:20:39 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Sat, 24 Aug 2013 17:20:39 -0700 (PDT)
Subject: [Freeswitch-users] domain vs. domain_name variables
In-Reply-To: <CAKbxfG8HoigFN64JiUChGuj9X2r5ydNtcHRdrP0Mm5o-xSVeHA@mail.gmail.com>
References: <1377289699.93283.YahooMailNeo@web126203.mail.ne1.yahoo.com>	<5217DA3F.70205@xtronics.com>
	<CAKbxfG8HoigFN64JiUChGuj9X2r5ydNtcHRdrP0Mm5o-xSVeHA@mail.gmail.com>
Message-ID: <1377390039.74422.YahooMailNeo@web126202.mail.ne1.yahoo.com>

Anthony, thanks a lot for your explanations. I will try to put this together into a Wiki article.

I also searched through the C code, and indeed, many modules resort to the "domain" global variable if they don't find the domain in the command arguments.

switch_ivr_set_user() parses the user at domain string, finds the user, and sets two channel variables, "user_name" and "domain_name".
Some modules use this when they create a channel (such as incoming INVITE in mod_sofia).
But I could not find any C code which uses "domain_name" channel variable. I guess the intent is to use the variable in matching conditions in the dialplan.

Also, I saw both "domain" and "domain_name" event headers in event creation and processing, but didn't yet have time to figure out where they are used. 

So, it looks like $${domain} global variable is required in vars.xml for general FreeSWITCH stability (it would probably coredump in some occasions if it's undefined). But $${domain_name} is obsolete and can be skipped if no other parts in the dialplan refer to it.

Then, the channel variable ${domain_name} is the right place to indicate the user's realm. It would be used to look up the user in the directory and for the realm part in SIP messages.

am I right in my considerations? If so, I will prepare a Wiki text for approval.

thanks
stan






>________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Saturday, August 24, 2013 3:03 AM
>Subject: Re: [Freeswitch-users] domain vs. domain_name variables
> 
>
>
>It took me like an hour to compose this explanation.....I hope it at least gives you an idea.
>
>"domain" is a core variable its used as a fallback in certain situations when its defined and no other value is specified for domain.
>It's also used in the default configuration as a pre-processor substitution with the $${domain} notation which is substituted into the sofia profiles and conference etc.
>
>
>
>
>
>"domain_name" is a variable that only exists in the scope of the demo dialplan. ?Its set initially as a global in vars.xml to whatever domain is set to as a base value.
>From there its only set on a per-call basis to override and used in per call dialplan logic.
>
>
>Its is not true that domain must be an IP. ?It defaults to the same value as local_ip_v4 to make sure the system works on any box without any configuration.
>If you have a working FQHN that points to your box, you should set that value in your domain var instead of the ip if you want things to reference the actual domain name.
>
>
>
>
>I think once you learn that way the config files are setup in the default distribution are purely arbitrary the more sense it may begin to make.
>
>
>
>
>The thing that made your head hurt is trying to explain that in SIP there is a separation from the host portion of a URI and the IP its sending the packet to.
>Because of PROXIES etc you can easily send a SIP packet to a specific address that contains a request URI or To: header with a different host name or ip.
>
>
>
>
>think of domain as referring to the host portion of a user at host concept. ?The same notion of a REALM in radius. ?In SIP and in FS that can be an IP or a FQHN and it need not exist in the TLD dns on the internet or it can match one that exists and function on its own because it's not directly tied to dns. ?(For the sake of sanity do not use a real domain name that is not yours even if it works).
>
>
>Basically the path of least resistance is to keep all these names aligned.. This is what the default config does for you because you set the domain variable once and its pre-processed into all the key places in the config that have domains so they are the same everywhere. ?
>
>
>Say you set it to example.com
>
>
>* The default directory will have a domain defined called example.com with the default 1000-1019 extensions.
>* The internal sip profile will associate the internal profile with example.com and force any registrations etc to find that domain for auth.
>* The conferences will use example.com for presence etc.
>
>
>Then it uses domain_name in the extensions etc as a clone of that example.com domain that you can change at runtime on a per call basis. ?Most likely in the default config with no changes this value will remain example.com the whole time and there is no difference.
>
>
>Why? ?
>
>
>Because people love multi-home stuff and they want to configure all of these things separately and once they start doing this, the config still is usable.
>
>
>So:
>
>
>FS never resolves domains to the IP and vice versa for you like it might in your unix environment or in a web browser. ?IP and FQHN have the same significance in sip packets as a unique realm string. ?It's only because you are binding the SIP UA to ${local_ip_v4} that your sip works on the IP.
>
>
>The way SIP was designed to work, you would define a SRV record for the host name used in your sip uri and then that IP would be used in place of a default A record lookup. ?We can just as easily override that in a client by forcing the fields for the proxy address and domains.
>
>
>
>
>Say you used example.com like I mentioned above and your valid IP was 200.8.8.8
>
>
>You could change domain=example.com and restart FS and it would still come up.
>
>
>Then you could go take a sip soft phone and play with the advanced settings and set the proxy addr to 200.8.8.8 and set the host or domain field to example.com and configure it to always send calls to the proxy addr.
>
>
>Now you can use example.com in your phones all day long even though its not really your domain because its just saying that in the packets and you explicitly showed it what server to use.
>
>
>Next, instead of example.com you could use 1.2.3.4 (or some real ip) as the domain even though it's obviously silly.
>
>
>If you turn on the siptrace you'll see the packets hit your box and using those values like 1.2.3.4 and example.com in the host portion and talk to your client over the actual IP of the server.
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>?
>
>
>
>
>
>
>
>
>
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>
>
>
>
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>
>
>
>
>
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>On Fri, Aug 23, 2013 at 4:55 PM, Karl Schmidt <karl at xtronics.com> wrote:
>
>On 08/23/2013 03:28 PM, Stanislav Sinyagin wrote:
>>> What is the difference between ${domain} and ${domain_name} variables? Both are used in the source
>>> code, and example configs set them to the same value.
>>>
>>> What is the design consideration behind?
>>>
>>> Why not merging them into one variable?
>>>
>>
>>This is an issue that has me a bit puzzled as well.
>>
>>domain in vars.xml is supposed to be an IP address ( IMO Should have been called domain_ip )
>>domain_name is supposed to be a FQDN I think. grepping the configs tells me domain_name is never
>>used in the default config so you can probably ignore it.
>>
>>What is also not clear is the multiple uses of the term domain:
>>
>>
>>?From the wiki -
>>https://wiki.freeswitch.org/wiki/Sofia.conf.xml#The_Relationship_Between_SIP_Profiles_and_Domains :
>>
>>?> The domains inside the XML registry are completely different from the domains on the internet and
>>?> again completely different from domains in sip packets. The profiles are again entirely different
>>?> from any of the above. Its up to you to align them if you so choose.
>>
>>OK that made my head hurt.
>>
>>In some places in FS 'domain' = domain_IP_address
>>
>>Here it says that SIP profile domain != IP-domain != SIP packet domain.
>>
>>So I know what it isn't, but it might also help if I knew what it IS!
>>
>>So, I think the domain definition in sofia MIGHT be a grouping of profiles? Then as you read on, it
>>tells us that a domain is a domain.
>>
>>" FreeSWITCH works off the concept of users and domains just like email."
>>
>>The books nor the wiki nor googling the mailing list makes this clear.
>>
>>Both the internal and external profiles set the name of the sofia-domain to all and $ sofia status
>>gives the names as internal and external (after the directory? ).
>>
>>
>>
>>
>>--------------------------------------------------------------------------------
>>Karl Schmidt ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?EMail Karl at xtronics.com
>>Transtronics, Inc. ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?WEB http://secure.transtronics.com
>>3209 West 9th Street ? ? ? ? ? ? ? ? ? ? ? ? ? ? Ph (785) 841-3089
>>Lawrence, KS 66049 ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?FAX (785) 841-0434
>>
>>Reality isn't fair, but that it is where I live. -kps
>>--------------------------------------------------------------------------------
>>
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
>>consulting at freeswitch.org
>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
>>http://www.freeswitch.org
>>http://wiki.freeswitch.org
>>http://www.cluecon.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 
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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  Sun Aug 25 06:11:56 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 24 Aug 2013 21:11:56 -0500
Subject: [Freeswitch-users] domain vs. domain_name variables
In-Reply-To: <1377390039.74422.YahooMailNeo@web126202.mail.ne1.yahoo.com>
References: <1377289699.93283.YahooMailNeo@web126203.mail.ne1.yahoo.com>
	<5217DA3F.70205@xtronics.com>
	<CAKbxfG8HoigFN64JiUChGuj9X2r5ydNtcHRdrP0Mm5o-xSVeHA@mail.gmail.com>
	<1377390039.74422.YahooMailNeo@web126202.mail.ne1.yahoo.com>
Message-ID: <CAKbxfG9T+kCsx=3s1tiTietv2mon=2hTpgEP4LodrYNciJTcTw@mail.gmail.com>

I would be disappointed if not setting it resulted in any segs.  Hopefully
if so, someone would open jiras.
 On Aug 24, 2013 7:25 PM, "Stanislav Sinyagin" <ssinyagin at yahoo.com> wrote:

> Anthony, thanks a lot for your explanations. I will try to put this
> together into a Wiki article.
>
> I also searched through the C code, and indeed, many modules resort to the
> "domain" global variable if they don't find the domain in the command
> arguments.
>
> switch_ivr_set_user() parses the user at domain string, finds the user, and
> sets two channel variables, "user_name" and "domain_name".
> Some modules use this when they create a channel (such as incoming INVITE
> in mod_sofia).
> But I could not find any C code which uses "domain_name" channel variable.
> I guess the intent is to use the variable in matching conditions in the
> dialplan.
>
> Also, I saw both "domain" and "domain_name" event headers in event
> creation and processing, but didn't yet have time to figure out where they
> are used.
>
> So, it looks like $${domain} global variable is required in vars.xml for
> general FreeSWITCH stability (it would probably coredump in some occasions
> if it's undefined). But $${domain_name} is obsolete and can be skipped if
> no other parts in the dialplan refer to it.
>
> Then, the channel variable ${domain_name} is the right place to indicate
> the user's realm. It would be used to look up the user in the directory and
> for the realm part in SIP messages.
>
> am I right in my considerations? If so, I will prepare a Wiki text for
> approval.
>
> thanks
> stan
>
>
>
>   ------------------------------
>  *From:* Anthony Minessale <anthony.minessale at gmail.com>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Saturday, August 24, 2013 3:03 AM
> *Subject:* Re: [Freeswitch-users] domain vs. domain_name variables
>
> It took me like an hour to compose this explanation.....I hope it at least
> gives you an idea.
>
> "domain" is a core variable its used as a fallback in certain situations
> when its defined and no other value is specified for domain.
> It's also used in the default configuration as a pre-processor
> substitution with the $${domain} notation which is substituted into the
> sofia profiles and conference etc.
>
>
> "domain_name" is a variable that only exists in the scope of the demo
> dialplan.  Its set initially as a global in vars.xml to whatever domain is
> set to as a base value.
> >From there its only set on a per-call basis to override and used in per
> call dialplan logic.
>
> Its is not true that domain must be an IP.  It defaults to the same value
> as local_ip_v4 to make sure the system works on any box without any
> configuration.
> If you have a working FQHN that points to your box, you should set that
> value in your domain var instead of the ip if you want things to reference
> the actual domain name.
>
>
> I think once you learn that way the config files are setup in the default
> distribution are purely arbitrary the more sense it may begin to make.
>
>
> The thing that made your head hurt is trying to explain that in SIP there
> is a separation from the host portion of a URI and the IP its sending the
> packet to.
> Because of PROXIES etc you can easily send a SIP packet to a specific
> address that contains a request URI or To: header with a different host
> name or ip.
>
>
> think of domain as referring to the host portion of a user at host concept.
>  The same notion of a REALM in radius.  In SIP and in FS that can be an IP
> or a FQHN and it need not exist in the TLD dns on the internet or it can
> match one that exists and function on its own because it's not directly
> tied to dns.  (For the sake of sanity do not use a real domain name that is
> not yours even if it works).
>
> Basically the path of least resistance is to keep all these names
> aligned.. This is what the default config does for you because you set the
> domain variable once and its pre-processed into all the key places in the
> config that have domains so they are the same everywhere.
>
> Say you set it to example.com
>
> * The default directory will have a domain defined called example.comwith the default 1000-1019 extensions.
> * The internal sip profile will associate the internal profile with
> example.com and force any registrations etc to find that domain for auth.
> * The conferences will use example.com for presence etc.
>
> Then it uses domain_name in the extensions etc as a clone of that
> example.com domain that you can change at runtime on a per call basis.
>  Most likely in the default config with no changes this value will remain
> example.com the whole time and there is no difference.
>
> Why?
>
> Because people love multi-home stuff and they want to configure all of
> these things separately and once they start doing this, the config still is
> usable.
>
> So:
>
> FS never resolves domains to the IP and vice versa for you like it might
> in your unix environment or in a web browser.  IP and FQHN have the same
> significance in sip packets as a unique realm string.  It's only because
> you are binding the SIP UA to ${local_ip_v4} that your sip works on the IP.
>
> The way SIP was designed to work, you would define a SRV record for the
> host name used in your sip uri and then that IP would be used in place of a
> default A record lookup.  We can just as easily override that in a client
> by forcing the fields for the proxy address and domains.
>
>
> Say you used example.com like I mentioned above and your valid IP was
> 200.8.8.8
>
> You could change domain=example.com and restart FS and it would still
> come up.
>
> Then you could go take a sip soft phone and play with the advanced
> settings and set the proxy addr to 200.8.8.8 and set the host or domain
> field to example.com and configure it to always send calls to the proxy
> addr.
>
> Now you can use example.com in your phones all day long even though its
> not really your domain because its just saying that in the packets and you
> explicitly showed it what server to use.
>
> Next, instead of example.com you could use 1.2.3.4 (or some real ip) as
> the domain even though it's obviously silly.
>
> If you turn on the siptrace you'll see the packets hit your box and using
> those values like 1.2.3.4 and example.com in the host portion and talk to
> your client over the actual IP of the server.
>
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>
> On Fri, Aug 23, 2013 at 4:55 PM, Karl Schmidt <karl at xtronics.com> wrote:
>
> On 08/23/2013 03:28 PM, Stanislav Sinyagin wrote:
> > What is the difference between ${domain} and ${domain_name} variables?
> Both are used in the source
> > code, and example configs set them to the same value.
> >
> > What is the design consideration behind?
> >
> > Why not merging them into one variable?
> >
>
> This is an issue that has me a bit puzzled as well.
>
> domain in vars.xml is supposed to be an IP address ( IMO Should have been
> called domain_ip )
> domain_name is supposed to be a FQDN I think. grepping the configs tells
> me domain_name is never
> used in the default config so you can probably ignore it.
>
> What is also not clear is the multiple uses of the term domain:
>
>
>  From the wiki -
>
> https://wiki.freeswitch.org/wiki/Sofia.conf.xml#The_Relationship_Between_SIP_Profiles_and_Domains:
>
>  > The domains inside the XML registry are completely different from the
> domains on the internet and
>  > again completely different from domains in sip packets. The profiles
> are again entirely different
>  > from any of the above. Its up to you to align them if you so choose.
>
> OK that made my head hurt.
>
> In some places in FS 'domain' = domain_IP_address
>
> Here it says that SIP profile domain != IP-domain != SIP packet domain.
>
> So I know what it isn't, but it might also help if I knew what it IS!
>
> So, I think the domain definition in sofia MIGHT be a grouping of
> profiles? Then as you read on, it
> tells us that a domain is a domain.
>
> " FreeSWITCH works off the concept of users and domains just like email."
>
> The books nor the wiki nor googling the mailing list makes this clear.
>
> Both the internal and external profiles set the name of the sofia-domain
> to all and $ sofia status
> gives the names as internal and external (after the directory? ).
>
>
>
>
>
> --------------------------------------------------------------------------------
> Karl Schmidt                                  EMail Karl at xtronics.com
> Transtronics, Inc.                              WEB
> http://secure.transtronics.com
> 3209 West 9th Street                             Ph (785) 841-3089
> Lawrence, KS 66049                              FAX (785) 841-0434
>
> Reality isn't fair, but that it is where I live. -kps
>
> --------------------------------------------------------------------------------
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Sun Aug 25 08:44:28 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Sun, 25 Aug 2013 10:14:28 +0530
Subject: [Freeswitch-users] mod_gsmopen: some success - hear crackly
 unintelligible announcement
In-Reply-To: <CAMvhjJuxmeUdgZBr3ZrxPBaNQO=Q=2ntkRoXsxEbuEnoaK5NcA@mail.gmail.com>
References: <CAHDL-szwednREpc+ZpTvVXRTRrudozbTjhhmYd-oaj4iLyOgXg@mail.gmail.com>
	<CAMvhjJty+9hV75Wu=iQcs2mHhfNQ4qB9QaPN3z4Kr=X-5+mubQ@mail.gmail.com>
	<CAHDL-sxnDwO53sRkg0wPrf2QFbyPu1hWE3bGvWmPS9+x+GxxUA@mail.gmail.com>
	<CAMvhjJuxmeUdgZBr3ZrxPBaNQO=Q=2ntkRoXsxEbuEnoaK5NcA@mail.gmail.com>
Message-ID: <CAHDL-swM-XySi_GakNu7uw_Jh+oDWo6TJFRp+RANphLWTsx57g@mail.gmail.com>

Sorry, but I don't think I understood the question. The Airtel SIM card I
am using is a 32K standard issues SIM card. Already enabled for everything.
I knew that it doesn't have a PIN enabled, which I believe is the only
requirement. The SIM has the IMSI associated with it. The dongle has the
IMEI associated with it. I just set up the standard gsmopen.conf.xml to set
the right /dev/ttyUSBx's -- figuring that out correctly, took a while...
for which I refer to a mail on this list around use of wvdial (search the
archives, as I've not kept a bookmark on it). Rest everything was taken
care of by FS.


On Sun, Aug 25, 2013 at 2:02 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Thanks jayanath
> But i mean how to enable the sim u had used in the dongle for making calls
> through mod_gsmopen make and receive calls ?
>
> Regards
> Ashish Mishra
> On Aug 25, 2013 1:21 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:
>
>> Using the default dialplan. I have only 1 dongle at the moment, so doing
>> a telephone -> telephone mapping won't work (well, at least, based on what
>> I've understood so far -- I could be wrong). Tomorrow, I plan to walk
>> through the dialplan to understand it somewhat better. My eyes don't take
>> XML too kindly !!
>>
>> As such no significant changes. The only few hiccups to build, and the
>> previous segfault I got, I filed a JIRA for, and that has the steps I took
>> and my preliminary findings.
>> http://jira.freeswitch.org/browse/FS-5721
>> http://jira.freeswitch.org/browse/FS-5727
>>
>> I used a trick I found in the mail archives to use "wvdial" to identify
>> the control port. After that, set the usb ports for control/audio correctly
>> in the gsmopen.conf.xml file. After this the module started getting loaded
>> properly, but I noticed that the signal strength was too low. Post that, I
>> had to use a long USB extension cable to keep the dongle outside the
>> window, and ensured that signal strength (RSSI) was good enough. After
>> this, the rig started working.
>>
>>
>>
>> On Sun, Aug 25, 2013 at 12:31 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> Hi jaynath
>>> May i ask you what what was was the dialplan which you had used to call
>>> <any indian phone number> to <any indian phone number> in your mod_gsmopen.
>>> Also what did you do with error u received day before yesterday....???
>>>
>>> Regards
>>> Ashish Mishra
>>> On Aug 25, 2013 12:05 AM, "Jayanth Acharya" <jayachar88 at gmail.com>
>>> wrote:
>>>
>>>> Finally, found success with mod_gsmopen. Have to say that there's a
>>>> tonne of information in the Wiki, and it takes a newbie some time to get to
>>>> understand even the elementary things. Jumping right in, assuming defaults
>>>> to take care of everything, doesn't work. I believe that was thanks to
>>>> being connected to the conference bridge 888 at sip.freeswitch.net (if I
>>>> noted the SIP URI correctly).Next tried another inbound call, and this
>>>> time, I heard only silence.
>>>>
>>>> Then I tried sending an SMS. That went out fine and was received
>>>> instantaneously by the destination number. After that, I replied back to
>>>> the dongle's number, and had a telnet session open listening on the events
>>>> (none arrived), but that telnet session was closed, and turning to the FS
>>>> console, I saw that it too had closed with a segmentation fault. So seems
>>>> like receiving the SMS precipitated a core.
>>>>
>>>> Anyone seen similar behaviour ? Should I go ahead an file another JIRA
>>>> bugreport ? File one yesterday regarding another segmentation fault. Note,
>>>> I am on Git HEAD (or was, day before, when I cloned it).
>>>>
>>>> thanks,
>>>> Jay
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Sun Aug 25 08:47:27 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Sun, 25 Aug 2013 10:17:27 +0530
Subject: [Freeswitch-users] mod_gsmopen: some success - hear crackly
 unintelligible announcement
In-Reply-To: <CAHDL-swM-XySi_GakNu7uw_Jh+oDWo6TJFRp+RANphLWTsx57g@mail.gmail.com>
References: <CAHDL-szwednREpc+ZpTvVXRTRrudozbTjhhmYd-oaj4iLyOgXg@mail.gmail.com>
	<CAMvhjJty+9hV75Wu=iQcs2mHhfNQ4qB9QaPN3z4Kr=X-5+mubQ@mail.gmail.com>
	<CAHDL-sxnDwO53sRkg0wPrf2QFbyPu1hWE3bGvWmPS9+x+GxxUA@mail.gmail.com>
	<CAMvhjJuxmeUdgZBr3ZrxPBaNQO=Q=2ntkRoXsxEbuEnoaK5NcA@mail.gmail.com>
	<CAHDL-swM-XySi_GakNu7uw_Jh+oDWo6TJFRp+RANphLWTsx57g@mail.gmail.com>
Message-ID: <CAHDL-swyWyQhC+Z_TfbSK3F4e4GMTfwO2HvrfcHbTNaAvgZ81w@mail.gmail.com>

Maybe, if you share what exactly you have tried, what worked/didn't work,
with the relevant FS log messages... like I mentioned earlier, I could try
to take a look.
To be honest, I don't think I have encountered a very difficult problem
with FS so far, so I don't think I had to do much... of course, apart from
the 2 different instances of segmentation faults.

PS> I created a 3rd JIRA as well, for the 2nd segmentation fault.. but I
don't think you are facing seg faults.



On Sun, Aug 25, 2013 at 10:14 AM, Jayanth Acharya <jayachar88 at gmail.com>wrote:

> Sorry, but I don't think I understood the question. The Airtel SIM card I
> am using is a 32K standard issues SIM card. Already enabled for everything.
> I knew that it doesn't have a PIN enabled, which I believe is the only
> requirement. The SIM has the IMSI associated with it. The dongle has the
> IMEI associated with it. I just set up the standard gsmopen.conf.xml to set
> the right /dev/ttyUSBx's -- figuring that out correctly, took a while...
> for which I refer to a mail on this list around use of wvdial (search the
> archives, as I've not kept a bookmark on it). Rest everything was taken
> care of by FS.
>
>
> On Sun, Aug 25, 2013 at 2:02 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Thanks jayanath
>> But i mean how to enable the sim u had used in the dongle for making
>> calls through mod_gsmopen make and receive calls ?
>>
>> Regards
>> Ashish Mishra
>> On Aug 25, 2013 1:21 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:
>>
>>> Using the default dialplan. I have only 1 dongle at the moment, so doing
>>> a telephone -> telephone mapping won't work (well, at least, based on what
>>> I've understood so far -- I could be wrong). Tomorrow, I plan to walk
>>> through the dialplan to understand it somewhat better. My eyes don't take
>>> XML too kindly !!
>>>
>>> As such no significant changes. The only few hiccups to build, and the
>>> previous segfault I got, I filed a JIRA for, and that has the steps I took
>>> and my preliminary findings.
>>> http://jira.freeswitch.org/browse/FS-5721
>>> http://jira.freeswitch.org/browse/FS-5727
>>>
>>> I used a trick I found in the mail archives to use "wvdial" to identify
>>> the control port. After that, set the usb ports for control/audio correctly
>>> in the gsmopen.conf.xml file. After this the module started getting loaded
>>> properly, but I noticed that the signal strength was too low. Post that, I
>>> had to use a long USB extension cable to keep the dongle outside the
>>> window, and ensured that signal strength (RSSI) was good enough. After
>>> this, the rig started working.
>>>
>>>
>>>
>>> On Sun, Aug 25, 2013 at 12:31 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> Hi jaynath
>>>> May i ask you what what was was the dialplan which you had used to call
>>>> <any indian phone number> to <any indian phone number> in your mod_gsmopen.
>>>> Also what did you do with error u received day before yesterday....???
>>>>
>>>> Regards
>>>> Ashish Mishra
>>>> On Aug 25, 2013 12:05 AM, "Jayanth Acharya" <jayachar88 at gmail.com>
>>>> wrote:
>>>>
>>>>> Finally, found success with mod_gsmopen. Have to say that there's a
>>>>> tonne of information in the Wiki, and it takes a newbie some time to get to
>>>>> understand even the elementary things. Jumping right in, assuming defaults
>>>>> to take care of everything, doesn't work. I believe that was thanks to
>>>>> being connected to the conference bridge 888 at sip.freeswitch.net (if I
>>>>> noted the SIP URI correctly).Next tried another inbound call, and this
>>>>> time, I heard only silence.
>>>>>
>>>>> Then I tried sending an SMS. That went out fine and was received
>>>>> instantaneously by the destination number. After that, I replied back to
>>>>> the dongle's number, and had a telnet session open listening on the events
>>>>> (none arrived), but that telnet session was closed, and turning to the FS
>>>>> console, I saw that it too had closed with a segmentation fault. So seems
>>>>> like receiving the SMS precipitated a core.
>>>>>
>>>>> Anyone seen similar behaviour ? Should I go ahead an file another JIRA
>>>>> bugreport ? File one yesterday regarding another segmentation fault. Note,
>>>>> I am on Git HEAD (or was, day before, when I cloned it).
>>>>>
>>>>> thanks,
>>>>> Jay
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 jayachar88 at gmail.com  Sun Aug 25 08:49:01 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Sun, 25 Aug 2013 10:19:01 +0530
Subject: [Freeswitch-users] Cluecon presentations - archived / available
	?
In-Reply-To: <CE3E33D5.7938D%krice@freeswitch.org>
References: <CAHDL-swWULXmUVghAN7xEvF=1Qao48F=xOuyfmbh-qfemQ+UiQ@mail.gmail.com>
	<CE3E33D5.7938D%krice@freeswitch.org>
Message-ID: <CAHDL-swioAC-AMFN=JDGHHcWryw8FtXwoJS8D3mQX3yuc5biXg@mail.gmail.com>

Looking forward to it Ken. I presume this list is included in the "mailing
lists" the notice is going to be sent out to, right ?


On Sat, Aug 24, 2013 at 8:26 PM, Ken Rice <krice at freeswitch.org> wrote:

>  The 2013 videos are being loaded to youtube... There will be links
> published shortly via cluecon.com with notices going out via social media
> and the mailing lists
>
>
>
> On 8/23/13 11:42 PM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:
>
> Many thanks. Wonder if other list members are aware of Cluecon 2013
> decks... there are 2-3 I am quite interested in seeing.
>
> Just as a hint to organizers... if recorded videos of sessions with
> presentations are made available for reasonable fee per session, I think
> many of us might lap it up !
>
>
> On Sat, Aug 24, 2013 at 8:29 AM, William King <
> william.king at quentustech.com> wrote:
>
> http://torrents.freeswitch.org/
>
> At least the 2011 ClueCon torrent is there.
>
> William King
> Senior Engineer
> Quentus Technologies, INC
> 1037 NE 65th St Suite 273
> Seattle, WA 98115
> Main:   (877) 211-9337
> Office: (206) 388-4772
> Cell:   (253) 686-5518
> william.king at quentustech.com
>
> On 08/23/2013 07:49 PM, Jayanth Acharya wrote:
> > Not sure if this is the right list for asking this question, but I
> > wasn't aware of a better one.
> >
> > Are the presentations done in past Cluecons archived and available for
> > download from some location ? Scribd / Slideshare etc. ?
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> ------------------------------
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Sun Aug 25 11:14:01 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 25 Aug 2013 12:44:01 +0530
Subject: [Freeswitch-users] mod_gsmopen: some success - hear crackly
 unintelligible announcement
In-Reply-To: <CAHDL-swyWyQhC+Z_TfbSK3F4e4GMTfwO2HvrfcHbTNaAvgZ81w@mail.gmail.com>
References: <CAHDL-szwednREpc+ZpTvVXRTRrudozbTjhhmYd-oaj4iLyOgXg@mail.gmail.com>
	<CAMvhjJty+9hV75Wu=iQcs2mHhfNQ4qB9QaPN3z4Kr=X-5+mubQ@mail.gmail.com>
	<CAHDL-sxnDwO53sRkg0wPrf2QFbyPu1hWE3bGvWmPS9+x+GxxUA@mail.gmail.com>
	<CAMvhjJuxmeUdgZBr3ZrxPBaNQO=Q=2ntkRoXsxEbuEnoaK5NcA@mail.gmail.com>
	<CAHDL-swM-XySi_GakNu7uw_Jh+oDWo6TJFRp+RANphLWTsx57g@mail.gmail.com>
	<CAHDL-swyWyQhC+Z_TfbSK3F4e4GMTfwO2HvrfcHbTNaAvgZ81w@mail.gmail.com>
Message-ID: <CAMvhjJsCtOd4NUHwK1xM8ZAK8o8-s7a0n2bRVKsEL_sB55R-ZA@mail.gmail.com>

hi jayanath
i am attaching the error as well as the console log which i see when i load
mod_gsmopen in freeswitch.
i typed the commands:
console loglevel 9
fsctl loglevel 9
load mod_gsmopen

after which the log that i received is attached in log file and the error
which was in red is attached in error file respectively.

thanks
Ashish Mishra


On Sun, Aug 25, 2013 at 10:17 AM, Jayanth Acharya <jayachar88 at gmail.com>wrote:

> Maybe, if you share what exactly you have tried, what worked/didn't work,
> with the relevant FS log messages... like I mentioned earlier, I could try
> to take a look.
> To be honest, I don't think I have encountered a very difficult problem
> with FS so far, so I don't think I had to do much... of course, apart from
> the 2 different instances of segmentation faults.
>
> PS> I created a 3rd JIRA as well, for the 2nd segmentation fault.. but I
> don't think you are facing seg faults.
>
>
>
> On Sun, Aug 25, 2013 at 10:14 AM, Jayanth Acharya <jayachar88 at gmail.com>wrote:
>
>> Sorry, but I don't think I understood the question. The Airtel SIM card I
>> am using is a 32K standard issues SIM card. Already enabled for everything.
>> I knew that it doesn't have a PIN enabled, which I believe is the only
>> requirement. The SIM has the IMSI associated with it. The dongle has the
>> IMEI associated with it. I just set up the standard gsmopen.conf.xml to set
>> the right /dev/ttyUSBx's -- figuring that out correctly, took a while...
>> for which I refer to a mail on this list around use of wvdial (search the
>> archives, as I've not kept a bookmark on it). Rest everything was taken
>> care of by FS.
>>
>>
>> On Sun, Aug 25, 2013 at 2:02 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> Thanks jayanath
>>> But i mean how to enable the sim u had used in the dongle for making
>>> calls through mod_gsmopen make and receive calls ?
>>>
>>> Regards
>>> Ashish Mishra
>>> On Aug 25, 2013 1:21 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:
>>>
>>>> Using the default dialplan. I have only 1 dongle at the moment, so
>>>> doing a telephone -> telephone mapping won't work (well, at least, based on
>>>> what I've understood so far -- I could be wrong). Tomorrow, I plan to walk
>>>> through the dialplan to understand it somewhat better. My eyes don't take
>>>> XML too kindly !!
>>>>
>>>> As such no significant changes. The only few hiccups to build, and the
>>>> previous segfault I got, I filed a JIRA for, and that has the steps I took
>>>> and my preliminary findings.
>>>> http://jira.freeswitch.org/browse/FS-5721
>>>> http://jira.freeswitch.org/browse/FS-5727
>>>>
>>>> I used a trick I found in the mail archives to use "wvdial" to identify
>>>> the control port. After that, set the usb ports for control/audio correctly
>>>> in the gsmopen.conf.xml file. After this the module started getting loaded
>>>> properly, but I noticed that the signal strength was too low. Post that, I
>>>> had to use a long USB extension cable to keep the dongle outside the
>>>> window, and ensured that signal strength (RSSI) was good enough. After
>>>> this, the rig started working.
>>>>
>>>>
>>>>
>>>> On Sun, Aug 25, 2013 at 12:31 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>
>>>>> Hi jaynath
>>>>> May i ask you what what was was the dialplan which you had used to
>>>>> call <any indian phone number> to <any indian phone number> in your
>>>>> mod_gsmopen. Also what did you do with error u received day before
>>>>> yesterday....???
>>>>>
>>>>> Regards
>>>>> Ashish Mishra
>>>>> On Aug 25, 2013 12:05 AM, "Jayanth Acharya" <jayachar88 at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Finally, found success with mod_gsmopen. Have to say that there's a
>>>>>> tonne of information in the Wiki, and it takes a newbie some time to get to
>>>>>> understand even the elementary things. Jumping right in, assuming defaults
>>>>>> to take care of everything, doesn't work. I believe that was thanks to
>>>>>> being connected to the conference bridge 888 at sip.freeswitch.net (if
>>>>>> I noted the SIP URI correctly).Next tried another inbound call, and this
>>>>>> time, I heard only silence.
>>>>>>
>>>>>> Then I tried sending an SMS. That went out fine and was received
>>>>>> instantaneously by the destination number. After that, I replied back to
>>>>>> the dongle's number, and had a telnet session open listening on the events
>>>>>> (none arrived), but that telnet session was closed, and turning to the FS
>>>>>> console, I saw that it too had closed with a segmentation fault. So seems
>>>>>> like receiving the SMS precipitated a core.
>>>>>>
>>>>>> Anyone seen similar behaviour ? Should I go ahead an file another
>>>>>> JIRA bugreport ? File one yesterday regarding another segmentation fault.
>>>>>> Note, I am on Git HEAD (or was, day before, when I cloned it).
>>>>>>
>>>>>> thanks,
>>>>>> Jay
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Sun Aug 25 11:43:11 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Sun, 25 Aug 2013 13:13:11 +0530
Subject: [Freeswitch-users] mod_gsmopen: some success - hear crackly
 unintelligible announcement
In-Reply-To: <CAMvhjJsCtOd4NUHwK1xM8ZAK8o8-s7a0n2bRVKsEL_sB55R-ZA@mail.gmail.com>
References: <CAHDL-szwednREpc+ZpTvVXRTRrudozbTjhhmYd-oaj4iLyOgXg@mail.gmail.com>
	<CAMvhjJty+9hV75Wu=iQcs2mHhfNQ4qB9QaPN3z4Kr=X-5+mubQ@mail.gmail.com>
	<CAHDL-sxnDwO53sRkg0wPrf2QFbyPu1hWE3bGvWmPS9+x+GxxUA@mail.gmail.com>
	<CAMvhjJuxmeUdgZBr3ZrxPBaNQO=Q=2ntkRoXsxEbuEnoaK5NcA@mail.gmail.com>
	<CAHDL-swM-XySi_GakNu7uw_Jh+oDWo6TJFRp+RANphLWTsx57g@mail.gmail.com>
	<CAHDL-swyWyQhC+Z_TfbSK3F4e4GMTfwO2HvrfcHbTNaAvgZ81w@mail.gmail.com>
	<CAMvhjJsCtOd4NUHwK1xM8ZAK8o8-s7a0n2bRVKsEL_sB55R-ZA@mail.gmail.com>
Message-ID: <CAHDL-sydnPG0tCzwgv1wqkgLRa78iAMTGR6eBgV8UH76F9bj_Q@mail.gmail.com>

In future pls use pastebin to upload logs, instead of attaching them to
emails! These days people check mails on tablets etc., so large attachments
become a problem for them :-) !  Also, might be better to run this in an
independent thread and not use the existing threads.

I see this:
2013-08-25 12:27:03.563390 [DEBUG] mod_gsmopen.cpp:1598 rev
4a3d1a0|4a3d1a0[(nil)|37     ][DEBUG_GSMOPEN  1598 ][gsm01     ][-1, 0, 0]
controldevice_name=/dev/ttyUSB3
2013-08-25 12:27:03.563390 [DEBUG] mod_gsmopen.cpp:1599 rev
4a3d1a0|4a3d1a0[(nil)|37     ][DEBUG_GSMOPEN  1599 ][gsm01     ][-1, 0, 0]
controldevice_audio_name=/dev/ttyUSB2
2013-08-25 12:27:03.563390 [DEBUG] mod_gsmopen.cpp:1601 rev
4a3d1a0|4a3d1a0[(nil)|37     ][DEBUG_GSMOPEN  1601 ][gsm01     ][-1, 0, 0]
gsmopen_serial_sync_period=300
2013-08-25 12:27:03.563390 [ERR] gsmopen_protocol.cpp:137 rev
4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
/dev/ttyUSB3, NOT open

What does: "ls -l /dev/ttyUSB*" tell you ?
I presume that you are running FreeSwitch as a root, or as a user that has
been added to sudoers or similar logic to achieve root privileges at run
time ?
Also, did you try "wvdialconf /etc/wvdial.conf" as root ? It'll tell you
which ports are modem ports (i.e. control ports... as against voice ports
-- USB bulk transfer) ?



On Sun, Aug 25, 2013 at 12:44 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> hi jayanath
> i am attaching the error as well as the console log which i see when i
> load mod_gsmopen in freeswitch.
> i typed the commands:
> console loglevel 9
> fsctl loglevel 9
> load mod_gsmopen
>
> after which the log that i received is attached in log file and the error
> which was in red is attached in error file respectively.
>
> thanks
> Ashish Mishra
>
>
> On Sun, Aug 25, 2013 at 10:17 AM, Jayanth Acharya <jayachar88 at gmail.com>wrote:
>
>> Maybe, if you share what exactly you have tried, what worked/didn't work,
>> with the relevant FS log messages... like I mentioned earlier, I could try
>> to take a look.
>> To be honest, I don't think I have encountered a very difficult problem
>> with FS so far, so I don't think I had to do much... of course, apart from
>> the 2 different instances of segmentation faults.
>>
>> PS> I created a 3rd JIRA as well, for the 2nd segmentation fault.. but I
>> don't think you are facing seg faults.
>>
>>
>>
>> On Sun, Aug 25, 2013 at 10:14 AM, Jayanth Acharya <jayachar88 at gmail.com>wrote:
>>
>>> Sorry, but I don't think I understood the question. The Airtel SIM card
>>> I am using is a 32K standard issues SIM card. Already enabled for
>>> everything. I knew that it doesn't have a PIN enabled, which I believe is
>>> the only requirement. The SIM has the IMSI associated with it. The dongle
>>> has the IMEI associated with it. I just set up the standard
>>> gsmopen.conf.xml to set the right /dev/ttyUSBx's -- figuring that out
>>> correctly, took a while... for which I refer to a mail on this list around
>>> use of wvdial (search the archives, as I've not kept a bookmark on it).
>>> Rest everything was taken care of by FS.
>>>
>>>
>>> On Sun, Aug 25, 2013 at 2:02 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> Thanks jayanath
>>>> But i mean how to enable the sim u had used in the dongle for making
>>>> calls through mod_gsmopen make and receive calls ?
>>>>
>>>> Regards
>>>> Ashish Mishra
>>>> On Aug 25, 2013 1:21 AM, "Jayanth Acharya" <jayachar88 at gmail.com>
>>>> wrote:
>>>>
>>>>> Using the default dialplan. I have only 1 dongle at the moment, so
>>>>> doing a telephone -> telephone mapping won't work (well, at least, based on
>>>>> what I've understood so far -- I could be wrong). Tomorrow, I plan to walk
>>>>> through the dialplan to understand it somewhat better. My eyes don't take
>>>>> XML too kindly !!
>>>>>
>>>>> As such no significant changes. The only few hiccups to build, and the
>>>>> previous segfault I got, I filed a JIRA for, and that has the steps I took
>>>>> and my preliminary findings.
>>>>> http://jira.freeswitch.org/browse/FS-5721
>>>>> http://jira.freeswitch.org/browse/FS-5727
>>>>>
>>>>> I used a trick I found in the mail archives to use "wvdial" to
>>>>> identify the control port. After that, set the usb ports for control/audio
>>>>> correctly in the gsmopen.conf.xml file. After this the module started
>>>>> getting loaded properly, but I noticed that the signal strength was too
>>>>> low. Post that, I had to use a long USB extension cable to keep the dongle
>>>>> outside the window, and ensured that signal strength (RSSI) was good
>>>>> enough. After this, the rig started working.
>>>>>
>>>>>
>>>>>
>>>>> On Sun, Aug 25, 2013 at 12:31 AM, Ashish Mishra <itsme.kunnu at gmail.com
>>>>> > wrote:
>>>>>
>>>>>> Hi jaynath
>>>>>> May i ask you what what was was the dialplan which you had used to
>>>>>> call <any indian phone number> to <any indian phone number> in your
>>>>>> mod_gsmopen. Also what did you do with error u received day before
>>>>>> yesterday....???
>>>>>>
>>>>>> Regards
>>>>>> Ashish Mishra
>>>>>> On Aug 25, 2013 12:05 AM, "Jayanth Acharya" <jayachar88 at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Finally, found success with mod_gsmopen. Have to say that there's a
>>>>>>> tonne of information in the Wiki, and it takes a newbie some time to get to
>>>>>>> understand even the elementary things. Jumping right in, assuming defaults
>>>>>>> to take care of everything, doesn't work. I believe that was thanks to
>>>>>>> being connected to the conference bridge 888 at sip.freeswitch.net (if
>>>>>>> I noted the SIP URI correctly).Next tried another inbound call, and this
>>>>>>> time, I heard only silence.
>>>>>>>
>>>>>>> Then I tried sending an SMS. That went out fine and was received
>>>>>>> instantaneously by the destination number. After that, I replied back to
>>>>>>> the dongle's number, and had a telnet session open listening on the events
>>>>>>> (none arrived), but that telnet session was closed, and turning to the FS
>>>>>>> console, I saw that it too had closed with a segmentation fault. So seems
>>>>>>> like receiving the SMS precipitated a core.
>>>>>>>
>>>>>>> Anyone seen similar behaviour ? Should I go ahead an file another
>>>>>>> JIRA bugreport ? File one yesterday regarding another segmentation fault.
>>>>>>> Note, I am on Git HEAD (or was, day before, when I cloned it).
>>>>>>>
>>>>>>> thanks,
>>>>>>> Jay
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Sun Aug 25 11:45:36 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Sun, 25 Aug 2013 13:15:36 +0530
Subject: [Freeswitch-users] mod_gsmopen: some success - hear crackly
 unintelligible announcement
In-Reply-To: <CAHDL-sydnPG0tCzwgv1wqkgLRa78iAMTGR6eBgV8UH76F9bj_Q@mail.gmail.com>
References: <CAHDL-szwednREpc+ZpTvVXRTRrudozbTjhhmYd-oaj4iLyOgXg@mail.gmail.com>
	<CAMvhjJty+9hV75Wu=iQcs2mHhfNQ4qB9QaPN3z4Kr=X-5+mubQ@mail.gmail.com>
	<CAHDL-sxnDwO53sRkg0wPrf2QFbyPu1hWE3bGvWmPS9+x+GxxUA@mail.gmail.com>
	<CAMvhjJuxmeUdgZBr3ZrxPBaNQO=Q=2ntkRoXsxEbuEnoaK5NcA@mail.gmail.com>
	<CAHDL-swM-XySi_GakNu7uw_Jh+oDWo6TJFRp+RANphLWTsx57g@mail.gmail.com>
	<CAHDL-swyWyQhC+Z_TfbSK3F4e4GMTfwO2HvrfcHbTNaAvgZ81w@mail.gmail.com>
	<CAMvhjJsCtOd4NUHwK1xM8ZAK8o8-s7a0n2bRVKsEL_sB55R-ZA@mail.gmail.com>
	<CAHDL-sydnPG0tCzwgv1wqkgLRa78iAMTGR6eBgV8UH76F9bj_Q@mail.gmail.com>
Message-ID: <CAHDL-syc5c_GtFkO8uYAXcFcj3h0-4j0JvehR62BSRADrviKCQ@mail.gmail.com>

Sorry, I did you mention the dongle model name/number earlier ?
And, did you ensure that it works fine for outgoing/incoming voice-calls,
MO/MT SMS's, MO/MT USSD via Huawei Mobile-Partner software in Windows ??
Don't just assume your device model to be supported... there are known
issues with certain firmware versions on the device, and certain chipsets.
Also, what Linux distro, release are you using ?


On Sun, Aug 25, 2013 at 1:13 PM, Jayanth Acharya <jayachar88 at gmail.com>wrote:

> In future pls use pastebin to upload logs, instead of attaching them to
> emails! These days people check mails on tablets etc., so large attachments
> become a problem for them :-) !  Also, might be better to run this in an
> independent thread and not use the existing threads.
>
> I see this:
> 2013-08-25 12:27:03.563390 [DEBUG] mod_gsmopen.cpp:1598 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][DEBUG_GSMOPEN  1598 ][gsm01     ][-1, 0, 0]
> controldevice_name=/dev/ttyUSB3
> 2013-08-25 12:27:03.563390 [DEBUG] mod_gsmopen.cpp:1599 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][DEBUG_GSMOPEN  1599 ][gsm01     ][-1, 0, 0]
> controldevice_audio_name=/dev/ttyUSB2
> 2013-08-25 12:27:03.563390 [DEBUG] mod_gsmopen.cpp:1601 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][DEBUG_GSMOPEN  1601 ][gsm01     ][-1, 0, 0]
> gsmopen_serial_sync_period=300
> 2013-08-25 12:27:03.563390 [ERR] gsmopen_protocol.cpp:137 rev
> 4a3d1a0|4a3d1a0[(nil)|37     ][ERRORA  137  ][gsm01     ][-1, 0, 0] port
> /dev/ttyUSB3, NOT open
>
> What does: "ls -l /dev/ttyUSB*" tell you ?
> I presume that you are running FreeSwitch as a root, or as a user that has
> been added to sudoers or similar logic to achieve root privileges at run
> time ?
> Also, did you try "wvdialconf /etc/wvdial.conf" as root ? It'll tell you
> which ports are modem ports (i.e. control ports... as against voice ports
> -- USB bulk transfer) ?
>
>
>
> On Sun, Aug 25, 2013 at 12:44 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> hi jayanath
>> i am attaching the error as well as the console log which i see when i
>> load mod_gsmopen in freeswitch.
>> i typed the commands:
>> console loglevel 9
>> fsctl loglevel 9
>> load mod_gsmopen
>>
>> after which the log that i received is attached in log file and the error
>> which was in red is attached in error file respectively.
>>
>> thanks
>> Ashish Mishra
>>
>>
>> On Sun, Aug 25, 2013 at 10:17 AM, Jayanth Acharya <jayachar88 at gmail.com>wrote:
>>
>>> Maybe, if you share what exactly you have tried, what worked/didn't
>>> work, with the relevant FS log messages... like I mentioned earlier, I
>>> could try to take a look.
>>> To be honest, I don't think I have encountered a very difficult problem
>>> with FS so far, so I don't think I had to do much... of course, apart from
>>> the 2 different instances of segmentation faults.
>>>
>>> PS> I created a 3rd JIRA as well, for the 2nd segmentation fault.. but I
>>> don't think you are facing seg faults.
>>>
>>>
>>>
>>> On Sun, Aug 25, 2013 at 10:14 AM, Jayanth Acharya <jayachar88 at gmail.com>wrote:
>>>
>>>> Sorry, but I don't think I understood the question. The Airtel SIM card
>>>> I am using is a 32K standard issues SIM card. Already enabled for
>>>> everything. I knew that it doesn't have a PIN enabled, which I believe is
>>>> the only requirement. The SIM has the IMSI associated with it. The dongle
>>>> has the IMEI associated with it. I just set up the standard
>>>> gsmopen.conf.xml to set the right /dev/ttyUSBx's -- figuring that out
>>>> correctly, took a while... for which I refer to a mail on this list around
>>>> use of wvdial (search the archives, as I've not kept a bookmark on it).
>>>> Rest everything was taken care of by FS.
>>>>
>>>>
>>>> On Sun, Aug 25, 2013 at 2:02 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>
>>>>> Thanks jayanath
>>>>> But i mean how to enable the sim u had used in the dongle for making
>>>>> calls through mod_gsmopen make and receive calls ?
>>>>>
>>>>> Regards
>>>>> Ashish Mishra
>>>>> On Aug 25, 2013 1:21 AM, "Jayanth Acharya" <jayachar88 at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Using the default dialplan. I have only 1 dongle at the moment, so
>>>>>> doing a telephone -> telephone mapping won't work (well, at least, based on
>>>>>> what I've understood so far -- I could be wrong). Tomorrow, I plan to walk
>>>>>> through the dialplan to understand it somewhat better. My eyes don't take
>>>>>> XML too kindly !!
>>>>>>
>>>>>> As such no significant changes. The only few hiccups to build, and
>>>>>> the previous segfault I got, I filed a JIRA for, and that has the steps I
>>>>>> took and my preliminary findings.
>>>>>> http://jira.freeswitch.org/browse/FS-5721
>>>>>> http://jira.freeswitch.org/browse/FS-5727
>>>>>>
>>>>>> I used a trick I found in the mail archives to use "wvdial" to
>>>>>> identify the control port. After that, set the usb ports for control/audio
>>>>>> correctly in the gsmopen.conf.xml file. After this the module started
>>>>>> getting loaded properly, but I noticed that the signal strength was too
>>>>>> low. Post that, I had to use a long USB extension cable to keep the dongle
>>>>>> outside the window, and ensured that signal strength (RSSI) was good
>>>>>> enough. After this, the rig started working.
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Sun, Aug 25, 2013 at 12:31 AM, Ashish Mishra <
>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>
>>>>>>> Hi jaynath
>>>>>>> May i ask you what what was was the dialplan which you had used to
>>>>>>> call <any indian phone number> to <any indian phone number> in your
>>>>>>> mod_gsmopen. Also what did you do with error u received day before
>>>>>>> yesterday....???
>>>>>>>
>>>>>>> Regards
>>>>>>> Ashish Mishra
>>>>>>> On Aug 25, 2013 12:05 AM, "Jayanth Acharya" <jayachar88 at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Finally, found success with mod_gsmopen. Have to say that there's a
>>>>>>>> tonne of information in the Wiki, and it takes a newbie some time to get to
>>>>>>>> understand even the elementary things. Jumping right in, assuming defaults
>>>>>>>> to take care of everything, doesn't work. I believe that was thanks to
>>>>>>>> being connected to the conference bridge 888 at sip.freeswitch.net(if I noted the SIP URI correctly).Next tried another inbound call, and
>>>>>>>> this time, I heard only silence.
>>>>>>>>
>>>>>>>> Then I tried sending an SMS. That went out fine and was received
>>>>>>>> instantaneously by the destination number. After that, I replied back to
>>>>>>>> the dongle's number, and had a telnet session open listening on the events
>>>>>>>> (none arrived), but that telnet session was closed, and turning to the FS
>>>>>>>> console, I saw that it too had closed with a segmentation fault. So seems
>>>>>>>> like receiving the SMS precipitated a core.
>>>>>>>>
>>>>>>>> Anyone seen similar behaviour ? Should I go ahead an file another
>>>>>>>> JIRA bugreport ? File one yesterday regarding another segmentation fault.
>>>>>>>> Note, I am on Git HEAD (or was, day before, when I cloned it).
>>>>>>>>
>>>>>>>> thanks,
>>>>>>>> Jay
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 itsme.kunnu at gmail.com  Sun Aug 25 12:00:43 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 25 Aug 2013 13:30:43 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
Message-ID: <CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>

Hi jayanth,
This is the new thread that i am starting. I am using the dongle E1732
huawei which works well as far as receiving and dialing the calls is
concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
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From itsme.kunnu at gmail.com  Sun Aug 25 12:09:46 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 25 Aug 2013 13:39:46 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
Message-ID: <CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>

Also ls -l /dev/ttyUSB* gives me no such file or directory.
On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:

> Hi jayanth,
> This is the new thread that i am starting. I am using the dongle E1732
> huawei which works well as far as receiving and dialing the calls is
> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>
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From gmaruzz at gmail.com  Sun Aug 25 12:56:33 2013
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Sun, 25 Aug 2013 10:56:33 +0200
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
Message-ID: <CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>

you have probably not installed all the prerequisites as per the wiki page.
Your linux does not see the dongle.
Please follow the wiki page step by step



On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Also ls -l /dev/ttyUSB* gives me no such file or directory.
> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>
>> Hi jayanth,
>> This is the new thread that i am starting. I am using the dongle E1732
>> huawei which works well as far as receiving and dialing the calls is
>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From jayachar88 at gmail.com  Sun Aug 25 13:04:02 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Sun, 25 Aug 2013 14:34:02 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
Message-ID: <CAHDL-szOMu0QqXoMMOcB0tRAQcJXax338R0QA8WkA1ccewLa5g@mail.gmail.com>

That is the first problem to solve, assuming that you followed all the
instructions on Wiki page meticulously... something I said earlier :-) !

Things work well if we follow Wiki instructions carefully.... but of
course, sometimes you may still face issues.

Does "lsusb -v" as root, show you some device from "Huawei" ?? If you have
filesystem autolaunch enabled, does plugging in the dongle automatically
mount it as a storage drive ? That might indicate taht you don't have
usbmodeswitch installed or installed properly !


On Sun, Aug 25, 2013 at 1:39 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Also ls -l /dev/ttyUSB* gives me no such file or directory.
> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>
>> Hi jayanth,
>> This is the new thread that i am starting. I am using the dongle E1732
>> huawei which works well as far as receiving and dialing the calls is
>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ssinyagin at yahoo.com  Sun Aug 25 14:21:02 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Sun, 25 Aug 2013 03:21:02 -0700 (PDT)
Subject: [Freeswitch-users] domain vs. domain_name variables
In-Reply-To: <CAKbxfG9T+kCsx=3s1tiTietv2mon=2hTpgEP4LodrYNciJTcTw@mail.gmail.com>
References: <1377289699.93283.YahooMailNeo@web126203.mail.ne1.yahoo.com>	<5217DA3F.70205@xtronics.com>	<CAKbxfG8HoigFN64JiUChGuj9X2r5ydNtcHRdrP0Mm5o-xSVeHA@mail.gmail.com>	<1377390039.74422.YahooMailNeo@web126202.mail.ne1.yahoo.com>
	<CAKbxfG9T+kCsx=3s1tiTietv2mon=2hTpgEP4LodrYNciJTcTw@mail.gmail.com>
Message-ID: <1377426062.77984.YahooMailNeo@web126203.mail.ne1.yahoo.com>

here's the first potential coredump that I found by grepping through the sources:

"mod/applications/mod_commands/mod_commands.c" line 470
??????? if (zstr(domain)) {
??????????????? dup_domain = switch_core_get_variable_dup("domain");
??????????????? domain = dup_domain;
??????? }


further on, the domain value is used directly in SQL statement without checking if it's NULL.







>________________________________
> From: Anthony Minessale <anthony.minessale at gmail.com>
>To: Freeswitch-users <freeswitch-users at lists.freeswitch.org> 
>Sent: Sunday, August 25, 2013 4:11 AM
>Subject: Re: [Freeswitch-users] domain vs. domain_name variables
> 
>
>
>I would be disappointed if not setting it resulted in any segs.? Hopefully if so, someone would open jiras.
>
>On Aug 24, 2013 7:25 PM, "Stanislav Sinyagin" <ssinyagin at yahoo.com> wrote:
>
>Anthony, thanks a lot for your explanations. I will try to put this together into a Wiki article.
>>
>>I also searched through the C code, and indeed, many modules resort to the "domain" global variable if they don't find the domain in the command arguments.
>>
>>switch_ivr_set_user() parses the user at domain string, finds the user, and sets two channel variables, "user_name" and "domain_name".
>>Some modules use this when they create a channel (such as incoming INVITE in mod_sofia).
>>But I could not find any C code which uses "domain_name" channel variable. I guess the intent is to use the variable in matching conditions in the dialplan.
>>
>>Also, I saw both "domain" and "domain_name" event headers in event creation and processing, but didn't yet have time to figure out where they are used. 
>>
>>So, it looks like $${domain}
 global variable is required in vars.xml for general FreeSWITCH stability (it would probably coredump in some occasions if it's undefined). But $${domain_name} is obsolete and can be skipped if no other parts in the dialplan refer to it.
>>
>>Then, the channel variable ${domain_name} is the right place to indicate the user's realm. It would be used to look up the user in the directory and for the realm part in SIP messages.
>>
>>am I right in my considerations? If so, I will prepare a Wiki text for approval.
>>
>>thanks
>>stan
>>
>>
>>
>>
>>
>>
>>
>>>________________________________
>>> From: Anthony Minessale <anthony.minessale at gmail.com>
>>>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>>>Sent: Saturday, August 24, 2013 3:03 AM
>>>Subject: Re: [Freeswitch-users] domain vs. domain_name variables
>>> 
>>>
>>>
>>>It took me like an hour to compose this explanation.....I hope it at least gives you an idea.
>>>
>>>"domain" is a core variable its used as a fallback in certain situations when its defined and no other value is specified for domain.
>>>It's also used in the default configuration as a pre-processor substitution with the $${domain} notation which is substituted into the sofia profiles and conference etc.
>>>
>>>
>>>
>>>
>>>
>>>"domain_name" is a variable that only exists in the scope of the demo dialplan. ?Its set initially as a global in vars.xml to whatever domain is set to as a base value.
>>>>From there its only set on a per-call basis to override and used in per call dialplan logic.
>>>
>>>
>>>Its is not true that domain must be an IP. ?It defaults to the same value as local_ip_v4 to make sure the system works on any box without any configuration.
>>>If you have a working FQHN that points to your box, you should set that value in your domain var instead of the ip if you want things to reference the actual domain name.
>>>
>>>
>>>
>>>
>>>I think once you learn that way the config files are setup in the default distribution are purely arbitrary the more sense it may begin to make.
>>>
>>>
>>>
>>>
>>>The thing that made your head hurt is trying to explain that in SIP there is a separation from the host portion of a URI and the IP its sending the packet to.
>>>Because of PROXIES etc you can easily send a SIP packet to a specific address that contains a request URI or To: header with a different host name or ip.
>>>
>>>
>>>
>>>
>>>think of domain as referring to the host portion of a user at host concept. ?The same notion of a REALM in radius. ?In SIP and in FS that can be an IP or a FQHN and it need not exist in the TLD dns on the internet or it can match one that exists and function on its own because it's not directly tied to dns. ?(For the sake of sanity do not use a real domain name that is not yours even if it works).
>>>
>>>
>>>Basically the path of least resistance is to keep all these names aligned.. This is what the default config does for you because you set the domain variable once and its pre-processed into all the key places in the config that have domains so they are the same everywhere. ?
>>>
>>>
>>>Say you set it to example.com
>>>
>>>
>>>* The default directory will have a domain defined called example.com with the default 1000-1019 extensions.
>>>* The internal sip profile will associate the internal profile with example.com and force any registrations etc to find that domain for auth.
>>>* The conferences will use example.com for presence etc.
>>>
>>>
>>>Then it uses domain_name in the extensions etc as a clone of that example.com domain that you can change at runtime on a per call basis. ?Most likely in the default config with no changes this value will remain example.com the whole time and there is no difference.
>>>
>>>
>>>Why? ?
>>>
>>>
>>>Because people love multi-home stuff and they want to configure all of these things separately and once they start doing this, the config still is usable.
>>>
>>>
>>>So:
>>>
>>>
>>>FS never resolves domains to the IP and vice versa for you like it might in your unix environment or in a web browser. ?IP and FQHN have the same significance in sip packets as a unique realm string. ?It's only because you are binding the SIP UA to ${local_ip_v4} that your sip works on the IP.
>>>
>>>
>>>The way SIP was designed to work, you would define a SRV record for the host name used in your sip uri and then that IP would be used in place of a default A record lookup. ?We can just as easily override that in a client by forcing the fields for the proxy address and domains.
>>>
>>>
>>>
>>>
>>>Say you used example.com like I mentioned above and your valid IP was 200.8.8.8
>>>
>>>
>>>You could change domain=example.com and restart FS and it would still come up.
>>>
>>>
>>>Then you could go take a sip soft phone and play with the advanced settings and set the proxy addr to 200.8.8.8 and set the host or domain field to example.com and configure it to always send calls to the proxy addr.
>>>
>>>
>>>Now you can use example.com in your phones all day long even though its not really your domain because its just saying that in the packets and you explicitly showed it what server to use.
>>>
>>>
>>>Next, instead of example.com you could use 1.2.3.4 (or some real ip) as the domain even though it's obviously silly.
>>>
>>>
>>>If you turn on the siptrace you'll see the packets hit your box and using those values like 1.2.3.4 and example.com in the host portion and talk to your client over the actual IP of the server.
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>?
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>?
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>
>>>On Fri, Aug 23, 2013 at 4:55 PM, Karl Schmidt <karl at xtronics.com> wrote:
>>>
>>>On 08/23/2013 03:28 PM, Stanislav Sinyagin wrote:
>>>>> What is the difference between ${domain} and ${domain_name} variables? Both are used in the source
>>>>> code, and example configs set them to the same value.
>>>>>
>>>>> What is the design consideration behind?
>>>>>
>>>>> Why not merging them into one variable?
>>>>>
>>>>
>>>>This is an issue that has me a bit puzzled as well.
>>>>
>>>>domain in vars.xml is supposed to be an IP address ( IMO Should have been called domain_ip )
>>>>domain_name is supposed to be a FQDN I think. grepping the configs tells me domain_name is never
>>>>used in the default config so you can probably ignore it.
>>>>
>>>>What is also not clear is the multiple uses of the term domain:
>>>>
>>>>
>>>>?From the wiki -
>>>>https://wiki.freeswitch.org/wiki/Sofia.conf.xml#The_Relationship_Between_SIP_Profiles_and_Domains :
>>>>
>>>>?> The domains inside the XML registry are completely different from the domains on the internet and
>>>>?> again completely different from domains in sip packets. The profiles are again entirely different
>>>>?> from any of the above. Its up to you to align them if you so choose.
>>>>
>>>>OK that made my head hurt.
>>>>
>>>>In some places in FS 'domain' = domain_IP_address
>>>>
>>>>Here it says that SIP profile domain != IP-domain != SIP packet domain.
>>>>
>>>>So I know what it isn't, but it might also help if I knew what it IS!
>>>>
>>>>So, I think the domain definition in sofia MIGHT be a grouping of profiles? Then as you read on, it
>>>>tells us that a domain is a domain.
>>>>
>>>>" FreeSWITCH works off the concept of users and domains just like email."
>>>>
>>>>The books nor the wiki nor googling the mailing list makes this clear.
>>>>
>>>>Both the internal and external profiles set the name of the sofia-domain to all and $ sofia status
>>>>gives the names as internal and external (after the directory? ).
>>>>
>>>>
>>>>
>>>>
>>>>--------------------------------------------------------------------------------
>>>>Karl Schmidt ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?EMail Karl at xtronics.com
>>>>Transtronics, Inc. ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?WEB http://secure.transtronics.com
>>>>3209 West 9th Street ? ? ? ? ? ? ? ? ? ? ? ? ? ? Ph (785) 841-3089
>>>>Lawrence, KS 66049 ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?FAX (785) 841-0434
>>>>
>>>>Reality isn't fair, but that it is where I live. -kps
>>>>--------------------------------------------------------------------------------
>>>>
>>>>_________________________________________________________________________
>>>>Professional FreeSWITCH Consulting Services:
>>>>consulting at freeswitch.org
>>>>http://www.freeswitchsolutions.com
>>>>
>>>>
>>>>
>>>>
>>>>Official FreeSWITCH Sites
>>>>http://www.freeswitch.org
>>>>http://wiki.freeswitch.org
>>>>http://www.cluecon.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 
>>>_________________________________________________________________________
>>>Professional FreeSWITCH Consulting Services:
>>>consulting at freeswitch.org
>>>http://www.freeswitchsolutions.com
>>>
>>>
>>>
>>>
>>>Official FreeSWITCH Sites
>>>http://www.freeswitch.org
>>>http://wiki.freeswitch.org
>>>http://www.cluecon.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
>>>
>>>
>>>
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
>>consulting at freeswitch.org
>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
>>http://www.freeswitch.org
>>http://wiki.freeswitch.org
>>http://www.cluecon.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
>>
>>
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 itsme.kunnu at gmail.com  Sun Aug 25 14:26:19 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 25 Aug 2013 15:56:19 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
Message-ID: <CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>

Thank you,
Which prerequisites are you talking about...??? The prerequisites for
mod_gsmopen? I have followed the wiki page that entails the prerequisites
and perhaps to some extent i feel that i have installed all the
prerequisites. But if you guys i will give it another shot.

Regards
Ashish Mishra
On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:

> you have probably not installed all the prerequisites as per the wiki page.
> Your linux does not see the dongle.
> Please follow the wiki page step by step
>
>
>
> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>>
>>> Hi jayanth,
>>> This is the new thread that i am starting. I am using the dongle E1732
>>> huawei which works well as far as receiving and dialing the calls is
>>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Sun Aug 25 14:29:57 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 25 Aug 2013 15:59:57 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
	<CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
Message-ID: <CAMvhjJuDZRsEHsoppOPKimfXLBHxhvnRZqs4DbWTKjZ7S5WyYQ@mail.gmail.com>

Yes jayanth as soon as i plug in the dongle it mounts as a storage device.
On Aug 25, 2013 3:56 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:

> Thank you,
> Which prerequisites are you talking about...??? The prerequisites for
> mod_gsmopen? I have followed the wiki page that entails the prerequisites
> and perhaps to some extent i feel that i have installed all the
> prerequisites. But if you guys i will give it another shot.
>
> Regards
> Ashish Mishra
> On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>
>> you have probably not installed all the prerequisites as per the wiki
>> page.
>> Your linux does not see the dongle.
>> Please follow the wiki page step by step
>>
>>
>>
>> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>>>
>>>> Hi jayanth,
>>>> This is the new thread that i am starting. I am using the dongle E1732
>>>> huawei which works well as far as receiving and dialing the calls is
>>>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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  Sun Aug 25 14:32:19 2013
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 25 Aug 2013 12:32:19 +0200
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
	<CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
Message-ID: <CALXCt0rb7A8wXHFwNmMOLEYiEwer234NNsTaNGfjw5hneQUdFQ@mail.gmail.com>

apt-get install usb-modeswitch-data usb-modeswitch



On Sun, Aug 25, 2013 at 12:26 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Thank you,
> Which prerequisites are you talking about...??? The prerequisites for
> mod_gsmopen? I have followed the wiki page that entails the prerequisites
> and perhaps to some extent i feel that i have installed all the
> prerequisites. But if you guys i will give it another shot.
>
> Regards
> Ashish Mishra
> On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>
>> you have probably not installed all the prerequisites as per the wiki
>> page.
>> Your linux does not see the dongle.
>> Please follow the wiki page step by step
>>
>>
>>
>> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>>>
>>>> Hi jayanth,
>>>> This is the new thread that i am starting. I am using the dongle E1732
>>>> huawei which works well as far as receiving and dialing the calls is
>>>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From itsme.kunnu at gmail.com  Sun Aug 25 14:33:39 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 25 Aug 2013 16:03:39 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAHDL-szOMu0QqXoMMOcB0tRAQcJXax338R0QA8WkA1ccewLa5g@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CAHDL-szOMu0QqXoMMOcB0tRAQcJXax338R0QA8WkA1ccewLa5g@mail.gmail.com>
Message-ID: <CAMvhjJs4xoz6DmvOV39ovk454T1y0f8xjG0A+RMcCvwaKaYefg@mail.gmail.com>

Also lsusb -v shows me no huawei device.
On Aug 25, 2013 2:37 PM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:

> That is the first problem to solve, assuming that you followed all the
> instructions on Wiki page meticulously... something I said earlier :-) !
>
> Things work well if we follow Wiki instructions carefully.... but of
> course, sometimes you may still face issues.
>
> Does "lsusb -v" as root, show you some device from "Huawei" ?? If you have
> filesystem autolaunch enabled, does plugging in the dongle automatically
> mount it as a storage drive ? That might indicate taht you don't have
> usbmodeswitch installed or installed properly !
>
>
> On Sun, Aug 25, 2013 at 1:39 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>>
>>> Hi jayanth,
>>> This is the new thread that i am starting. I am using the dongle E1732
>>> huawei which works well as far as receiving and dialing the calls is
>>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Sun Aug 25 14:40:54 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Sun, 25 Aug 2013 16:10:54 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CALXCt0rb7A8wXHFwNmMOLEYiEwer234NNsTaNGfjw5hneQUdFQ@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
	<CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
	<CALXCt0rb7A8wXHFwNmMOLEYiEwer234NNsTaNGfjw5hneQUdFQ@mail.gmail.com>
Message-ID: <CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>

Thank you so much..after this will my dongle be easily detected by FS..???
On Aug 25, 2013 4:09 PM, "Giovanni Maruzzelli" <gmaruzz at celliax.org> wrote:

> apt-get install usb-modeswitch-data usb-modeswitch
>
>
>
> On Sun, Aug 25, 2013 at 12:26 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Thank you,
>> Which prerequisites are you talking about...??? The prerequisites for
>> mod_gsmopen? I have followed the wiki page that entails the prerequisites
>> and perhaps to some extent i feel that i have installed all the
>> prerequisites. But if you guys i will give it another shot.
>>
>> Regards
>> Ashish Mishra
>> On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>>
>>> you have probably not installed all the prerequisites as per the wiki
>>> page.
>>> Your linux does not see the dongle.
>>> Please follow the wiki page step by step
>>>
>>>
>>>
>>> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>>>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>>>>
>>>>> Hi jayanth,
>>>>> This is the new thread that i am starting. I am using the dongle E1732
>>>>> huawei which works well as far as receiving and dialing the calls is
>>>>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 a.afzali2003 at gmail.com  Sun Aug 25 16:29:12 2013
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Sun, 25 Aug 2013 16:59:12 +0430
Subject: [Freeswitch-users] freeswitch 1.2.12 build fails on centos 6.2
	32bit
In-Reply-To: <CAEVfcaYHR88=sJPCHtm+kpKPvd6qk56rdxUc8JJcmxpr+XqrOg@mail.gmail.com>
References: <CAEVfcaYHR88=sJPCHtm+kpKPvd6qk56rdxUc8JJcmxpr+XqrOg@mail.gmail.com>
Message-ID: <CABK0fC6oWsTMQ9-AtBgV06Li9pRoyXY6On2R1+igcWcriydQng@mail.gmail.com>

Babak,

check this:
http://jira.freeswitch.org/browse/FS-3384


On Sat, Aug 24, 2013 at 6:32 PM, Babak Yakhchali <babak.freeswitch at gmail.com
> wrote:

> Hi
> I'm trying to build freeswitch 1.2.12 in centos 6.2 but I get:
>
>  gcc -I/usr/local/src/freeswitch-1.2.12/libs/curl/include
> -I/usr/local/src/freeswitch-1.2.12/src/include
> -I/usr/local/src/freeswitch-1.2.12/src/include
> -I/usr/local/src/freeswitch-1.2.12/libs/libteletone/src
> -I/usr/local/src/freeswitch-1.2.12/libs/stfu -fPIC -Werror
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
> -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g
> -O2 -o .libs/freeswitch freeswitch-switch.o -Wl,-lodbc -Wl,-z -Wl,relro
>  -lm ./.libs/libfreeswitch.so
> -L/usr/local/src/freeswitch-1.2.12/libs/apr-util/xml/expat/lib -lpq
> /usr/local/src/freeswitch-1.2.12/libs/apr-util/xml/expat/lib/.libs/libexpat.a
> /usr/local/src/freeswitch-1.2.12/libs/apr/.libs/libapr-1.a
> -L/usr/local/src/freeswitch-1.2.12/libs/srtp libs/apr/.libs/libapr-1.a -lrt
> -lcrypt -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -ldl -lz
> -lncurses -ljpeg -lodbc  -Wl,--rpath -Wl,/usr/local/freeswitch/lib
> /usr/lib/libnss3.so: undefined reference to `PR_FindSymbol'
> /usr/lib/libnss3.so: undefined reference to `PR_RWLock_Rlock'
> /usr/lib/libssl3.so: undefined reference to `PR_OpenAnonFileMap'
> /usr/lib/libssl3.so: undefined reference to `PR_UnloadLibrary'
> /usr/lib/libnss3.so: undefined reference to `PL_InitArenaPool'
> /usr/lib/libnss3.so: undefined reference to `PR_NewRWLock'
> /usr/lib/libnss3.so: undefined reference to `PR_RWLock_Wlock'
> /usr/lib/libnss3.so: undefined reference to `PR_LoadLibrary'
> /lib/libldap_r-2.4.so.2: undefined reference to `PR_GetEnv'
> /usr/lib/libssl3.so: undefined reference to `PR_LoadLibraryWithFlags'
> /usr/lib/libnssutil3.so: undefined reference to `PL_ClearArenaPool'
> /usr/lib/libnss3.so: undefined reference to `PR_DestroyRWLock'
> /usr/lib/libnss3.so: undefined reference to `PR_NewTCPSocket'
> /lib/libldap_r-2.4.so.2: undefined reference to `PR_SetEnv'
> /lib/libldap_r-2.4.so.2: undefined reference to `PR_GetLibraryName'
> /usr/lib/libssl3.so: undefined reference to `PR_ErrorInstallTable'
> /usr/lib/libssl3.so: undefined reference to `PR_ExportFileMapAsString'
> /usr/lib/libssl3.so: undefined reference to `PR_GetLibraryFilePathname'
> /usr/lib/libssl3.so: undefined reference to `PR_FindFunctionSymbol'
> /usr/lib/libsmime3.so: undefined reference to `PL_NewHashTable'
> /lib/libldap_r-2.4.so.2: undefined reference to `PR_ErrorToString'
> /usr/lib/libnss3.so: undefined reference to `PR_RWLock_Unlock'
> /usr/lib/libssl3.so: undefined reference to `PR_ImportFileMapFromString'
> /lib/libldap_r-2.4.so.2: undefined reference to `PR_GetDirectorySeparator'
> collect2: ld returned 1 exit status
>
> I tried ./configure --without-libcurl --without-pgsql but no success
> thanks
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jleung at v10networks.ca  Sun Aug 25 18:58:21 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Sun, 25 Aug 2013 07:58:21 -0700
Subject: [Freeswitch-users] freeswitch 1.2.12 build fails on centos
	6.2	32bit
In-Reply-To: <CABK0fC6oWsTMQ9-AtBgV06Li9pRoyXY6On2R1+igcWcriydQng@mail.gmail.com>
References: <CAEVfcaYHR88=sJPCHtm+kpKPvd6qk56rdxUc8JJcmxpr+XqrOg@mail.gmail.com>
	<CABK0fC6oWsTMQ9-AtBgV06Li9pRoyXY6On2R1+igcWcriydQng@mail.gmail.com>
Message-ID: <005701cea1a3$8b6dc090$a24941b0$@v10networks.ca>

Nuke any existing FreeSWITCH installations and then retry.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of afshin
afzali
Sent: Sunday, August 25, 2013 5:29 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] freeswitch 1.2.12 build fails on centos 6.2
32bit

 

Babak,

 

check this:

http://jira.freeswitch.org/browse/FS-3384

 

On Sat, Aug 24, 2013 at 6:32 PM, Babak Yakhchali
<babak.freeswitch at gmail.com> wrote:

Hi

I'm trying to build freeswitch 1.2.12 in centos 6.2 but I get:

 

 gcc -I/usr/local/src/freeswitch-1.2.12/libs/curl/include
-I/usr/local/src/freeswitch-1.2.12/src/include
-I/usr/local/src/freeswitch-1.2.12/src/include
-I/usr/local/src/freeswitch-1.2.12/libs/libteletone/src
-I/usr/local/src/freeswitch-1.2.12/libs/stfu -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2
-o .libs/freeswitch freeswitch-switch.o -Wl,-lodbc -Wl,-z -Wl,relro  -lm
./.libs/libfreeswitch.so
-L/usr/local/src/freeswitch-1.2.12/libs/apr-util/xml/expat/lib -lpq
/usr/local/src/freeswitch-1.2.12/libs/apr-util/xml/expat/lib/.libs/libexpat.
a /usr/local/src/freeswitch-1.2.12/libs/apr/.libs/libapr-1.a
-L/usr/local/src/freeswitch-1.2.12/libs/srtp libs/apr/.libs/libapr-1.a -lrt
-lcrypt -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -ldl -lz
-lncurses -ljpeg -lodbc  -Wl,--rpath -Wl,/usr/local/freeswitch/lib

/usr/lib/libnss3.so: undefined reference to `PR_FindSymbol'

/usr/lib/libnss3.so: undefined reference to `PR_RWLock_Rlock'

/usr/lib/libssl3.so: undefined reference to `PR_OpenAnonFileMap'

/usr/lib/libssl3.so: undefined reference to `PR_UnloadLibrary'

/usr/lib/libnss3.so: undefined reference to `PL_InitArenaPool'

/usr/lib/libnss3.so: undefined reference to `PR_NewRWLock'

/usr/lib/libnss3.so: undefined reference to `PR_RWLock_Wlock'

/usr/lib/libnss3.so: undefined reference to `PR_LoadLibrary'

/lib/libldap_r-2.4.so.2: undefined reference to `PR_GetEnv'

/usr/lib/libssl3.so: undefined reference to `PR_LoadLibraryWithFlags'

/usr/lib/libnssutil3.so: undefined reference to `PL_ClearArenaPool'

/usr/lib/libnss3.so: undefined reference to `PR_DestroyRWLock'

/usr/lib/libnss3.so: undefined reference to `PR_NewTCPSocket'

/lib/libldap_r-2.4.so.2: undefined reference to `PR_SetEnv'

/lib/libldap_r-2.4.so.2: undefined reference to `PR_GetLibraryName'

/usr/lib/libssl3.so: undefined reference to `PR_ErrorInstallTable'

/usr/lib/libssl3.so: undefined reference to `PR_ExportFileMapAsString'

/usr/lib/libssl3.so: undefined reference to `PR_GetLibraryFilePathname'

/usr/lib/libssl3.so: undefined reference to `PR_FindFunctionSymbol'

/usr/lib/libsmime3.so: undefined reference to `PL_NewHashTable'

/lib/libldap_r-2.4.so.2: undefined reference to `PR_ErrorToString'

/usr/lib/libnss3.so: undefined reference to `PR_RWLock_Unlock'

/usr/lib/libssl3.so: undefined reference to `PR_ImportFileMapFromString'

/lib/libldap_r-2.4.so.2: undefined reference to `PR_GetDirectorySeparator'

collect2: ld returned 1 exit status

 

I tried ./configure --without-libcurl --without-pgsql but no success

thanks

 


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 at cassidywebservices.co.uk  Sun Aug 25 20:07:03 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sun, 25 Aug 2013 17:07:03 +0100
Subject: [Freeswitch-users] Pacemaker Resource Agent
In-Reply-To: <CAMbW9o+hQYHDNVCtTbB-3RSDBSDU8FA-9+SAf3Vs6qbvYA-EMw@mail.gmail.com>
References: <CAMbW9oJ3F_K3nMz0qKj9KFfYq_ZLB-buLkYAOu9s6xNooq6f1Q@mail.gmail.com>
	<00000140b257ea79-cfb47be2-f8c6-4466-86a0-c6bde0cc4ec9-000000@email.amazonses.com>
	<CAMbW9o+hQYHDNVCtTbB-3RSDBSDU8FA-9+SAf3Vs6qbvYA-EMw@mail.gmail.com>
Message-ID: <CAMbW9oL7HYmRgxNBJz-DkDn5xF5X1cejz9LD5pYmusMYrrQKMA@mail.gmail.com>

Just a follow up on this (I will either update the wiki or add my own page
at some point)

Firstly, I'm using Debian Wheezy, using binary packages wherever possible.

Both init scripts are incorrect at this stage as they both return 3 if
status it is unable to query freeswitch profile status. According the LSB
an exit code of 3 is used to indicate that the service was stopped
normally. The current version of pacemaker therefore does not fail over the
resource.

However, fsctl crash properly crashes, leaving the PID file behind, where
as stop does not. The LSB exit code for such an event is 2, which does make
pacemaker fail the resource over. As such, I have added a few lines to
check the existence of the PID file if the status check fails and return
either 2 or 3 as required.


On 25 August 2013 00:31, Andrew Cassidy <andrew at cassidywebservices.co.uk>wrote:

> I did not as for some reason Google does not deem the relevant to the term
> "freeswitch pacemaker resource agent"
>
> Thanks Avi.
>
>
> On 24 August 2013 23:00, Avi Marcus <avi at avimarcus.net> wrote:
>
>> Did you look at
>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_IP_Failover and
>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_with_Corosync?
>>
>> -Avi
>>
>>
>> On Sat, Aug 24, 2013 at 11:01 PM, Andrew Cassidy <
>> andrew at cassidywebservices.co.uk> wrote:
>>
>>> Hi all,
>>>
>>> I know this has come up before but I can't find a pacemaker resource
>>> agent for FreeSWITCH. I'm running a master/slave config and wondering if
>>> anyone has something they're willing to share?
>>>
>>> I know there's mod_ha_cluster, however I already have corosync/pacemaker
>>> for other services, and would like to keep things uniform.
>>>
>>> --
>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>> Managing Director
>>>
>>>
>>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>>  *03300 100 961
>>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From andrew at cassidywebservices.co.uk  Sun Aug 25 20:24:08 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sun, 25 Aug 2013 17:24:08 +0100
Subject: [Freeswitch-users] Pacemaker Resource Agent
In-Reply-To: <CAMbW9oL7HYmRgxNBJz-DkDn5xF5X1cejz9LD5pYmusMYrrQKMA@mail.gmail.com>
References: <CAMbW9oJ3F_K3nMz0qKj9KFfYq_ZLB-buLkYAOu9s6xNooq6f1Q@mail.gmail.com>
	<00000140b257ea79-cfb47be2-f8c6-4466-86a0-c6bde0cc4ec9-000000@email.amazonses.com>
	<CAMbW9o+hQYHDNVCtTbB-3RSDBSDU8FA-9+SAf3Vs6qbvYA-EMw@mail.gmail.com>
	<CAMbW9oL7HYmRgxNBJz-DkDn5xF5X1cejz9LD5pYmusMYrrQKMA@mail.gmail.com>
Message-ID: <CAMbW9oLarp-soSycAtDXpo_eJuV0PMdRhgZ6JjSNcrDzCEtUDA@mail.gmail.com>

Scratch that now my script has stopped making it fail over too.


On 25 August 2013 17:07, Andrew Cassidy <andrew at cassidywebservices.co.uk>wrote:

> Just a follow up on this (I will either update the wiki or add my own page
> at some point)
>
> Firstly, I'm using Debian Wheezy, using binary packages wherever possible.
>
> Both init scripts are incorrect at this stage as they both return 3 if
> status it is unable to query freeswitch profile status. According the LSB
> an exit code of 3 is used to indicate that the service was stopped
> normally. The current version of pacemaker therefore does not fail over the
> resource.
>
> However, fsctl crash properly crashes, leaving the PID file behind, where
> as stop does not. The LSB exit code for such an event is 2, which does make
> pacemaker fail the resource over. As such, I have added a few lines to
> check the existence of the PID file if the status check fails and return
> either 2 or 3 as required.
>
>
> On 25 August 2013 00:31, Andrew Cassidy <andrew at cassidywebservices.co.uk>wrote:
>
>> I did not as for some reason Google does not deem the relevant to the
>> term "freeswitch pacemaker resource agent"
>>
>> Thanks Avi.
>>
>>
>> On 24 August 2013 23:00, Avi Marcus <avi at avimarcus.net> wrote:
>>
>>>  Did you look at
>>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_IP_Failover and
>>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_with_Corosync?
>>>
>>> -Avi
>>>
>>>
>>> On Sat, Aug 24, 2013 at 11:01 PM, Andrew Cassidy <
>>> andrew at cassidywebservices.co.uk> wrote:
>>>
>>>> Hi all,
>>>>
>>>> I know this has come up before but I can't find a pacemaker resource
>>>> agent for FreeSWITCH. I'm running a master/slave config and wondering if
>>>> anyone has something they're willing to share?
>>>>
>>>> I know there's mod_ha_cluster, however I already have
>>>> corosync/pacemaker for other services, and would like to keep things
>>>> uniform.
>>>>
>>>> --
>>>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>>>> Managing Director
>>>>
>>>>
>>>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>>>  *03300 100 961
>>>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>>>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>>
>> --
>> *Andrew Cassidy BSc (Hons) MBCS SSCA*
>> Managing Director
>>
>>
>> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>>  *03300 100 961
>> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
>> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>>
>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From john_platts at hotmail.com  Sun Aug 25 22:02:35 2013
From: john_platts at hotmail.com (John Platts)
Date: Sun, 25 Aug 2013 13:02:35 -0500
Subject: [Freeswitch-users] Call drops when attended transfer is being
	performed
Message-ID: <BLU174-W48D3A69832BBCDD51976C19D480@phx.gbl>

Whenever I do an attended transfer a call, the caller gets dropped instead of being transferred to the next party. Here are the steps to reproducing the problem:Make a call to a phone that is registered with the FreeSWITCH server.Answer the call.Hit the transfer button on the phone that the call rings on to initiate an attended transfer.Initiate the call to party #3, which is the party that the call needs to be transferred to.Once the call to party #3 is made, hit the transfer button again to complete the transfer.The caller should now be transferred to party #3, but the call drops instead.
I am running FreeSWITCH 1.2.12 on the switch where I am experiencing the attended transfer problem. How do I get the call to get transferred from the caller to party #3, and how do I prevent the caller from getting disconnected when a call is transferred?
 		 	   		  
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From andrew at cassidywebservices.co.uk  Mon Aug 26 00:50:29 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Sun, 25 Aug 2013 21:50:29 +0100
Subject: [Freeswitch-users] Application Failover
Message-ID: <CAMbW9oLCcRQY-5cMc1F3wDZSBjNO7z-yZtRiH=z=BJqu3t-Qzw@mail.gmail.com>

So, I have been successful in getting failover working for SIP calls. My
question here is are there any other applications that can fail over in
this way? For example mod_callcenter?

-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From mike at ultra.net.br  Sat Aug 24 18:44:17 2013
From: mike at ultra.net.br (Mike Tesliuk)
Date: Sat, 24 Aug 2013 10:44:17 -0400
Subject: [Freeswitch-users] Cluecon presentations - archived / available
	?
In-Reply-To: <CAHDL-swWULXmUVghAN7xEvF=1Qao48F=xOuyfmbh-qfemQ+UiQ@mail.gmail.com>
References: <CAHDL-swipeXhTGtE50g0dj24bwWoPok2uEPSm8=BxLeyPrsFnw@mail.gmail.com>
	<52182192.8070605@quentustech.com>
	<CAHDL-swWULXmUVghAN7xEvF=1Qao48F=xOuyfmbh-qfemQ+UiQ@mail.gmail.com>
Message-ID: <CAKNtoCEdqtYCEvhcun=oY989rgFXAYzzRjdjvLi2KHCUJWoCVg@mail.gmail.com>

you have the videos on youtube, search for cluecon 2013


2013/8/24 Jayanth Acharya <jayachar88 at gmail.com>

> Many thanks. Wonder if other list members are aware of Cluecon 2013
> decks... there are 2-3 I am quite interested in seeing.
>
> Just as a hint to organizers... if recorded videos of sessions with
> presentations are made available for reasonable fee per session, I think
> many of us might lap it up !
>
>
> On Sat, Aug 24, 2013 at 8:29 AM, William King <
> william.king at quentustech.com> wrote:
>
>> http://torrents.freeswitch.org/
>>
>> At least the 2011 ClueCon torrent is there.
>>
>> William King
>> Senior Engineer
>> Quentus Technologies, INC
>> 1037 NE 65th St Suite 273
>> Seattle, WA 98115
>> Main:   (877) 211-9337
>> Office: (206) 388-4772
>> Cell:   (253) 686-5518
>> william.king at quentustech.com
>>
>> On 08/23/2013 07:49 PM, Jayanth Acharya wrote:
>> > Not sure if this is the right list for asking this question, but I
>> > wasn't aware of a better one.
>> >
>> > Are the presentations done in past Cluecons archived and available for
>> > download from some location ? Scribd / Slideshare etc. ?
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.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
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mike at ultra.net.br  Sat Aug 24 18:45:20 2013
From: mike at ultra.net.br (Mike Tesliuk)
Date: Sat, 24 Aug 2013 10:45:20 -0400
Subject: [Freeswitch-users] Cluecon presentations - archived / available
	?
In-Reply-To: <CAKNtoCEdqtYCEvhcun=oY989rgFXAYzzRjdjvLi2KHCUJWoCVg@mail.gmail.com>
References: <CAHDL-swipeXhTGtE50g0dj24bwWoPok2uEPSm8=BxLeyPrsFnw@mail.gmail.com>
	<52182192.8070605@quentustech.com>
	<CAHDL-swWULXmUVghAN7xEvF=1Qao48F=xOuyfmbh-qfemQ+UiQ@mail.gmail.com>
	<CAKNtoCEdqtYCEvhcun=oY989rgFXAYzzRjdjvLi2KHCUJWoCVg@mail.gmail.com>
Message-ID: <CAKNtoCE9nbkgQwJW26ovsir6cdwTM+txTJUwO1f833Fb9k-idg@mail.gmail.com>

sorry, you say that you want the presentations before 2013


2013/8/24 Mike Tesliuk <mike at ultra.net.br>

> you have the videos on youtube, search for cluecon 2013
>
>
> 2013/8/24 Jayanth Acharya <jayachar88 at gmail.com>
>
>> Many thanks. Wonder if other list members are aware of Cluecon 2013
>> decks... there are 2-3 I am quite interested in seeing.
>>
>> Just as a hint to organizers... if recorded videos of sessions with
>> presentations are made available for reasonable fee per session, I think
>> many of us might lap it up !
>>
>>
>> On Sat, Aug 24, 2013 at 8:29 AM, William King <
>> william.king at quentustech.com> wrote:
>>
>>> http://torrents.freeswitch.org/
>>>
>>> At least the 2011 ClueCon torrent is there.
>>>
>>> William King
>>> Senior Engineer
>>> Quentus Technologies, INC
>>> 1037 NE 65th St Suite 273
>>> Seattle, WA 98115
>>> Main:   (877) 211-9337
>>> Office: (206) 388-4772
>>> Cell:   (253) 686-5518
>>> william.king at quentustech.com
>>>
>>> On 08/23/2013 07:49 PM, Jayanth Acharya wrote:
>>> > Not sure if this is the right list for asking this question, but I
>>> > wasn't aware of a better one.
>>> >
>>> > Are the presentations done in past Cluecons archived and available for
>>> > download from some location ? Scribd / Slideshare etc. ?
>>> >
>>> >
>>> >
>>> _________________________________________________________________________
>>> > Professional FreeSWITCH Consulting Services:
>>> > consulting at freeswitch.org
>>> > http://www.freeswitchsolutions.com
>>> >
>>> > 
>>> > 
>>> >
>>> > Official FreeSWITCH Sites
>>> > http://www.freeswitch.org
>>> > http://wiki.freeswitch.org
>>> > http://www.cluecon.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
>>> >
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 e.kozhuhovskiy at gmail.com  Sun Aug 25 19:26:48 2013
From: e.kozhuhovskiy at gmail.com (Evgeniy Kozhuhovskiy)
Date: Sun, 25 Aug 2013 18:26:48 +0300
Subject: [Freeswitch-users] mod_callcenter, ODBC, mysql
Message-ID: <CAHbVSn21qNYNX6gHBp0VR2qZOeGWtpz=JmUOA4un6qbL6_0m-w@mail.gmail.com>

I've switched from sqlite to mysql in mod_callcenter, and in fs log I
see a lot of SQL errors

On start of mod_callcenter

2013-08-25 18:19:06.739653 [ERR] switch_odbc.c:514 ERR: [update agents
set state = 'Waiting', uuid = '' where system = 'single_box';update
tiers set state = 'Ready' where agent IN (select name from agents
where system = 'single_box');update members set state = 'Abandoned',
session_uuid = '' where system = 'single_box';]
[STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
check the manual that corresponds to your MySQL server version for the
right syntax to use near 'update tiers set state = 'Ready' where agent
IN (select name from agents where s' at line 1
]
2013-08-25 18:19:06.739653 [ERR] switch_core_sqldb.c:585 ODBC SQL ERR
[STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
check the manual that corresponds to your MySQL server version for the
right syntax to use near 'update tiers set state = 'Ready' where agent
IN (select name from agents where s' at line 1
]
update agents set state = 'Waiting', uuid = '' where system =
'single_box';update tiers set state = 'Ready' where agent IN (select
name from agents where system = 'single_box');update members set state
= 'Abandoned', session_uuid = '' where system = 'single_box';

And during work:

2013-08-25 18:18:22.059648 [ERR] switch_odbc.c:514 ERR: [UPDATE tiers
SET state = 'Ready' WHERE agent = '715' AND queue = 'nal at default' AND
(state = 'Active Inbound' OR state = 'Standby' OR state =
'Offering');UPDATE tiers SET state = 'Ready' WHERE agent = '715' AND
NOT queue = 'nal at default' AND state = 'Standby']
[STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
check the manual that corresponds to your MySQL server version for the
right syntax to use near 'UPDATE tiers SET state = 'Ready' WHERE agent
= '715' AND NOT queue = 'nal at defaul' at line 1
]
2013-08-25 18:18:22.059648 [ERR] switch_core_sqldb.c:585 ODBC SQL ERR
[STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
check the manual that corresponds to your MySQL server version for the
right syntax to use near 'UPDATE tiers SET state = 'Ready' WHERE agent
= '715' AND NOT queue = 'nal at defaul' at line 1
]
UPDATE tiers SET state = 'Ready' WHERE agent = '715' AND queue =
'nal at default' AND (state = 'Active Inbound' OR state = 'Standby' OR
state = 'Offering');UPDATE tiers SET state = 'Ready' WHERE agent =
'715' AND NOT queue = 'nal at default' AND state = 'Standby'
2013-08-25 18:18:22.059648 [DEBUG] mod_callcenter.c:1046 Updated Agent
715 set state = Waiting
2013-08-25 18:18:27.099646 [DEBUG] mod_callcenter.c:1046 Updated Agent
715 set state = Receiving
2013-08-25 18:18:27.099646 [ERR] switch_odbc.c:514 ERR: [UPDATE tiers
SET state = 'Offering' WHERE agent = '715' AND queue =
'nal at default';UPDATE tiers SET state = 'Standby' WHERE agent = '715'
AND NOT queue = 'nal at default' AND state = 'Ready';]
[STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
check the manual that corresponds to your MySQL server version for the
right syntax to use near 'UPDATE tiers SET state = 'Standby' WHERE
agent = '715' AND NOT queue = 'nal at defa' at line 1
]
2013-08-25 18:18:27.099646 [ERR] switch_core_sqldb.c:585 ODBC SQL ERR
[STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
check the manual that corresponds to your MySQL server version for the
right syntax to use near 'UPDATE tiers SET state = 'Standby' WHERE
agent = '715' AND NOT queue = 'nal at defa' at line 1
]
UPDATE tiers SET state = 'Offering' WHERE agent = '715' AND queue =
'nal at default';UPDATE tiers SET state = 'Standby' WHERE agent = '715'
AND NOT queue = 'nal at default' AND state = 'Ready';


But if I'm executing that queries directly at mysql client, everything
is ok. Is that a problem with ODBC drivers? Or what? May be someone
had the same expirience.


P.S

root at callcenter:~# /usr/local/freeswitch/bin/freeswitch -version
FreeSWITCH version: 1.5.5b+git~20130809T135445Z~60e8ca1bcc (git
60e8ca1 2013-08-09 13:54:45Z)
root at callcenter:~# cat /etc/debian_version
6.0.6
Everything except fs installed from debian repo.

-- 
With best regards, Evgeniy Kozhuhovskiy


From krice at freeswitch.org  Mon Aug 26 02:09:29 2013
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 25 Aug 2013 17:09:29 -0500
Subject: [Freeswitch-users] mod_callcenter, ODBC, mysql
In-Reply-To: <CAHbVSn21qNYNX6gHBp0VR2qZOeGWtpz=JmUOA4un6qbL6_0m-w@mail.gmail.com>
Message-ID: <CE3FEAC9.793F8%krice@freeswitch.org>

That's a thing in ODBC that only allows 1 query at a time... You need to
look at your ODBC/MySQL settings to force it to enable > 1 SQL query per
statement

K



On 8/25/13 10:26 AM, "Evgeniy Kozhuhovskiy" <e.kozhuhovskiy at gmail.com>
wrote:

> I've switched from sqlite to mysql in mod_callcenter, and in fs log I
> see a lot of SQL errors
> 
> On start of mod_callcenter
> 
> 2013-08-25 18:19:06.739653 [ERR] switch_odbc.c:514 ERR: [update agents
> set state = 'Waiting', uuid = '' where system = 'single_box';update
> tiers set state = 'Ready' where agent IN (select name from agents
> where system = 'single_box');update members set state = 'Abandoned',
> session_uuid = '' where system = 'single_box';]
> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
> Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
> check the manual that corresponds to your MySQL server version for the
> right syntax to use near 'update tiers set state = 'Ready' where agent
> IN (select name from agents where s' at line 1
> ]
> 2013-08-25 18:19:06.739653 [ERR] switch_core_sqldb.c:585 ODBC SQL ERR
> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
> Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
> check the manual that corresponds to your MySQL server version for the
> right syntax to use near 'update tiers set state = 'Ready' where agent
> IN (select name from agents where s' at line 1
> ]
> update agents set state = 'Waiting', uuid = '' where system =
> 'single_box';update tiers set state = 'Ready' where agent IN (select
> name from agents where system = 'single_box');update members set state
> = 'Abandoned', session_uuid = '' where system = 'single_box';
> 
> And during work:
> 
> 2013-08-25 18:18:22.059648 [ERR] switch_odbc.c:514 ERR: [UPDATE tiers
> SET state = 'Ready' WHERE agent = '715' AND queue = 'nal at default' AND
> (state = 'Active Inbound' OR state = 'Standby' OR state =
> 'Offering');UPDATE tiers SET state = 'Ready' WHERE agent = '715' AND
> NOT queue = 'nal at default' AND state = 'Standby']
> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
> Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
> check the manual that corresponds to your MySQL server version for the
> right syntax to use near 'UPDATE tiers SET state = 'Ready' WHERE agent
> = '715' AND NOT queue = 'nal at defaul' at line 1
> ]
> 2013-08-25 18:18:22.059648 [ERR] switch_core_sqldb.c:585 ODBC SQL ERR
> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
> Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
> check the manual that corresponds to your MySQL server version for the
> right syntax to use near 'UPDATE tiers SET state = 'Ready' WHERE agent
> = '715' AND NOT queue = 'nal at defaul' at line 1
> ]
> UPDATE tiers SET state = 'Ready' WHERE agent = '715' AND queue =
> 'nal at default' AND (state = 'Active Inbound' OR state = 'Standby' OR
> state = 'Offering');UPDATE tiers SET state = 'Ready' WHERE agent =
> '715' AND NOT queue = 'nal at default' AND state = 'Standby'
> 2013-08-25 18:18:22.059648 [DEBUG] mod_callcenter.c:1046 Updated Agent
> 715 set state = Waiting
> 2013-08-25 18:18:27.099646 [DEBUG] mod_callcenter.c:1046 Updated Agent
> 715 set state = Receiving
> 2013-08-25 18:18:27.099646 [ERR] switch_odbc.c:514 ERR: [UPDATE tiers
> SET state = 'Offering' WHERE agent = '715' AND queue =
> 'nal at default';UPDATE tiers SET state = 'Standby' WHERE agent = '715'
> AND NOT queue = 'nal at default' AND state = 'Ready';]
> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
> Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
> check the manual that corresponds to your MySQL server version for the
> right syntax to use near 'UPDATE tiers SET state = 'Standby' WHERE
> agent = '715' AND NOT queue = 'nal at defa' at line 1
> ]
> 2013-08-25 18:18:27.099646 [ERR] switch_core_sqldb.c:585 ODBC SQL ERR
> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
> Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
> check the manual that corresponds to your MySQL server version for the
> right syntax to use near 'UPDATE tiers SET state = 'Standby' WHERE
> agent = '715' AND NOT queue = 'nal at defa' at line 1
> ]
> UPDATE tiers SET state = 'Offering' WHERE agent = '715' AND queue =
> 'nal at default';UPDATE tiers SET state = 'Standby' WHERE agent = '715'
> AND NOT queue = 'nal at default' AND state = 'Ready';
> 
> 
> But if I'm executing that queries directly at mysql client, everything
> is ok. Is that a problem with ODBC drivers? Or what? May be someone
> had the same expirience.
> 
> 
> P.S
> 
> root at callcenter:~# /usr/local/freeswitch/bin/freeswitch -version
> FreeSWITCH version: 1.5.5b+git~20130809T135445Z~60e8ca1bcc (git
> 60e8ca1 2013-08-09 13:54:45Z)
> root at callcenter:~# cat /etc/debian_version
> 6.0.6
> Everything except fs installed from debian repo.

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch




From krice at freeswitch.org  Mon Aug 26 02:11:10 2013
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 25 Aug 2013 17:11:10 -0500
Subject: [Freeswitch-users] Cluecon presentations - archived / available
	?
In-Reply-To: <CAHDL-swioAC-AMFN=JDGHHcWryw8FtXwoJS8D3mQX3yuc5biXg@mail.gmail.com>
Message-ID: <CE3FEB2E.793F9%krice@freeswitch.org>

Yes, this would be one of the lists...


On 8/24/13 11:49 PM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:

> Looking forward to it Ken. I presume this list is included in the "mailing
> lists" the notice is going to be sent out to, right ?
> 
> 
> On Sat, Aug 24, 2013 at 8:26 PM, Ken Rice <krice at freeswitch.org> wrote:
>> The 2013 videos are being loaded to youtube... There will be links published
>> shortly via cluecon.com <http://cluecon.com>  with notices going out via
>> social media and the mailing lists
>> 
>> 
>> 
>> On 8/23/13 11:42 PM, "Jayanth Acharya" <jayachar88 at gmail.com
>> <http://jayachar88 at gmail.com> > wrote:
>> 
>>> Many thanks. Wonder if other list members are aware of Cluecon 2013 decks...
>>> there are 2-3 I am quite interested in seeing.
>>> 
>>> Just as a hint to organizers... if recorded videos of sessions with
>>> presentations are made available for reasonable fee per session, I think
>>> many of us might lap it up !
>>> 
>>> 
>>> On Sat, Aug 24, 2013 at 8:29 AM, William King <william.king at quentustech.com
>>> <http://william.king at quentustech.com> > wrote:
>>>> http://torrents.freeswitch.org/
>>>> 
>>>> At least the 2011 ClueCon torrent is there.
>>>> 
>>>> William King
>>>> Senior Engineer
>>>> Quentus Technologies, INC
>>>> 1037 NE 65th St Suite 273
>>>> Seattle, WA 98115
>>>> Main: ? (877) 211-9337
>>>> Office: (206) 388-4772
>>>> Cell: ? (253) 686-5518
>>>> william.king at quentustech.com <http://william.king at quentustech.com>
>>>> 
>>>> On 08/23/2013 07:49 PM, Jayanth Acharya wrote:
>>>>> > Not sure if this is the right list for asking this question, but I
>>>>> > wasn't aware of a better one.
>>>>> >
>>>>> > Are the presentations done in past Cluecons archived and available for
>>>>> > download from some location ? Scribd / Slideshare etc. ?
>>>>> >
>>>>> >
>>>>> > 
>>>>> _________________________________________________________________________
>>>>> > Professional FreeSWITCH Consulting Services:
>>>>> > consulting at freeswitch.org <http://consulting at freeswitch.org>
>>>>> > http://www.freeswitchsolutions.com
>>>>> >
>>>>> > 
>>>>> > 
>>>>> >
>>>>> > Official FreeSWITCH Sites
>>>>> > http://www.freeswitch.org
>>>>> > http://wiki.freeswitch.org
>>>>> > http://www.cluecon.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
>>>>> >
>>>> 
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org <http://consulting at freeswitch.org>
>>>> http://www.freeswitchsolutions.com
>>>> 
>>>> 
>>>> 
>>>> 
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org <http://consulting at freeswitch.org>
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From e.kozhuhovskiy at gmail.com  Mon Aug 26 02:40:19 2013
From: e.kozhuhovskiy at gmail.com (Evgeniy Kozhuhovskiy)
Date: Mon, 26 Aug 2013 01:40:19 +0300
Subject: [Freeswitch-users] mod_callcenter, ODBC, mysql
In-Reply-To: <CE3FEAC9.793F8%krice@freeswitch.org>
References: <CAHbVSn21qNYNX6gHBp0VR2qZOeGWtpz=JmUOA4un6qbL6_0m-w@mail.gmail.com>
	<CE3FEAC9.793F8%krice@freeswitch.org>
Message-ID: <CAHbVSn3u2Ox25yx6CnQuZg=y0=-Yk-9oRVRYKBCVDfc4sUG9rA@mail.gmail.com>

Thanks, I've already solved.

This behavior can be enabled in odbc.ini by adding OPTION=67108864

On Mon, Aug 26, 2013 at 1:09 AM, Ken Rice <krice at freeswitch.org> wrote:
> That's a thing in ODBC that only allows 1 query at a time... You need to
> look at your ODBC/MySQL settings to force it to enable > 1 SQL query per
> statement
>
> K
>
>
>
> On 8/25/13 10:26 AM, "Evgeniy Kozhuhovskiy" <e.kozhuhovskiy at gmail.com>
> wrote:
>
>> I've switched from sqlite to mysql in mod_callcenter, and in fs log I
>> see a lot of SQL errors
>>
>> On start of mod_callcenter
>>
>> 2013-08-25 18:19:06.739653 [ERR] switch_odbc.c:514 ERR: [update agents
>> set state = 'Waiting', uuid = '' where system = 'single_box';update
>> tiers set state = 'Ready' where agent IN (select name from agents
>> where system = 'single_box');update members set state = 'Abandoned',
>> session_uuid = '' where system = 'single_box';]
>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
>> Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
>> check the manual that corresponds to your MySQL server version for the
>> right syntax to use near 'update tiers set state = 'Ready' where agent
>> IN (select name from agents where s' at line 1
>> ]
>> 2013-08-25 18:19:06.739653 [ERR] switch_core_sqldb.c:585 ODBC SQL ERR
>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
>> Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
>> check the manual that corresponds to your MySQL server version for the
>> right syntax to use near 'update tiers set state = 'Ready' where agent
>> IN (select name from agents where s' at line 1
>> ]
>> update agents set state = 'Waiting', uuid = '' where system =
>> 'single_box';update tiers set state = 'Ready' where agent IN (select
>> name from agents where system = 'single_box');update members set state
>> = 'Abandoned', session_uuid = '' where system = 'single_box';
>>
>> And during work:
>>
>> 2013-08-25 18:18:22.059648 [ERR] switch_odbc.c:514 ERR: [UPDATE tiers
>> SET state = 'Ready' WHERE agent = '715' AND queue = 'nal at default' AND
>> (state = 'Active Inbound' OR state = 'Standby' OR state =
>> 'Offering');UPDATE tiers SET state = 'Ready' WHERE agent = '715' AND
>> NOT queue = 'nal at default' AND state = 'Standby']
>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
>> Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
>> check the manual that corresponds to your MySQL server version for the
>> right syntax to use near 'UPDATE tiers SET state = 'Ready' WHERE agent
>> = '715' AND NOT queue = 'nal at defaul' at line 1
>> ]
>> 2013-08-25 18:18:22.059648 [ERR] switch_core_sqldb.c:585 ODBC SQL ERR
>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
>> Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
>> check the manual that corresponds to your MySQL server version for the
>> right syntax to use near 'UPDATE tiers SET state = 'Ready' WHERE agent
>> = '715' AND NOT queue = 'nal at defaul' at line 1
>> ]
>> UPDATE tiers SET state = 'Ready' WHERE agent = '715' AND queue =
>> 'nal at default' AND (state = 'Active Inbound' OR state = 'Standby' OR
>> state = 'Offering');UPDATE tiers SET state = 'Ready' WHERE agent =
>> '715' AND NOT queue = 'nal at default' AND state = 'Standby'
>> 2013-08-25 18:18:22.059648 [DEBUG] mod_callcenter.c:1046 Updated Agent
>> 715 set state = Waiting
>> 2013-08-25 18:18:27.099646 [DEBUG] mod_callcenter.c:1046 Updated Agent
>> 715 set state = Receiving
>> 2013-08-25 18:18:27.099646 [ERR] switch_odbc.c:514 ERR: [UPDATE tiers
>> SET state = 'Offering' WHERE agent = '715' AND queue =
>> 'nal at default';UPDATE tiers SET state = 'Standby' WHERE agent = '715'
>> AND NOT queue = 'nal at default' AND state = 'Ready';]
>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
>> Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
>> check the manual that corresponds to your MySQL server version for the
>> right syntax to use near 'UPDATE tiers SET state = 'Standby' WHERE
>> agent = '715' AND NOT queue = 'nal at defa' at line 1
>> ]
>> 2013-08-25 18:18:27.099646 [ERR] switch_core_sqldb.c:585 ODBC SQL ERR
>> [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1
>> Driver][mysqld-5.1.66-0+squeeze1]You have an error in your SQL syntax;
>> check the manual that corresponds to your MySQL server version for the
>> right syntax to use near 'UPDATE tiers SET state = 'Standby' WHERE
>> agent = '715' AND NOT queue = 'nal at defa' at line 1
>> ]
>> UPDATE tiers SET state = 'Offering' WHERE agent = '715' AND queue =
>> 'nal at default';UPDATE tiers SET state = 'Standby' WHERE agent = '715'
>> AND NOT queue = 'nal at default' AND state = 'Ready';
>>
>>
>> But if I'm executing that queries directly at mysql client, everything
>> is ok. Is that a problem with ODBC drivers? Or what? May be someone
>> had the same expirience.
>>
>>
>> P.S
>>
>> root at callcenter:~# /usr/local/freeswitch/bin/freeswitch -version
>> FreeSWITCH version: 1.5.5b+git~20130809T135445Z~60e8ca1bcc (git
>> 60e8ca1 2013-08-09 13:54:45Z)
>> root at callcenter:~# cat /etc/debian_version
>> 6.0.6
>> Everything except fs installed from debian repo.
>
> --
> Ken
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> irc.freenode.net #freeswitch
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 best regards, Evgeniy Kozhuhovskiy


From gmangudai at gmail.com  Mon Aug 26 05:56:09 2013
From: gmangudai at gmail.com (Vincent Xia)
Date: Mon, 26 Aug 2013 09:56:09 +0800
Subject: [Freeswitch-users] is it possible to have FS forcilby unregiter
 a registered UA
In-Reply-To: <CAFiqYukMVqxHhs6rwL7QKZBn0Y-bZ+s3bWqAfUHOdR=33iWqUQ@mail.gmail.com>
References: <CAKQo6tnkgwmneo3QiG0cnBZF5LsApapsYZ4o778fT9SH+5SUGg@mail.gmail.com>
	<CAFiqYukMVqxHhs6rwL7QKZBn0Y-bZ+s3bWqAfUHOdR=33iWqUQ@mail.gmail.com>
Message-ID: <CAKQo6t=BQHZPigtGCDg=O_eMW=F2GFDxsaPZOem=Uf6sZ98RqQ@mail.gmail.com>

yes from the same user, but i have 2 profiles, 1 for LAN and 1 for
INTERNET, and i don't want 1 user to register both in LAN and INTERNET, so
im considering to kill the previous registration once i got register
request from the same user but a different profile.


2013/8/24 Steven Ayre <steveayre at gmail.com>

> to go one step further, how can i do the unregister upon receiving another
>> register request?
>
>
> From the same user? By default the user can only be registered at one
> location.
>
> Bear in mind registrations are timeout based, and refreshed by sending
> further REGISTER packets periodically.
>
> That means if you have 2 phones using the same user details the 2nd will
> register and replace the 1st, then the 1st reregister and replace the 2nd,
> then the 2nd reregister and replace the 1st...
>
> You can't stop them reregistering. Probably wouldn't want to either, or
> you'll stop blocking legitimate reregistrations, such as when a phone moves
> location.
>
> Not something you can do much about. You shouldn't use the same account in
> multiple places. Or allow them to register from multiple locations at once (
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#multiple-registrations)
> but bear in mind that'll affect anything that expects them in only one
> place.
>
> -Steve
>
>
>
> On 23 August 2013 11:52, Vincent Xia <gmangudai at gmail.com> wrote:
>
>> is it possible to have FS forcilby unregiter a registered UA, for
>> example, user/1001?
>>
>> to go one step further, how can i do the unregister upon receiving
>> another register request?
>> i checked event list of the event socket but cannot find any register
>> event, can anyone shed any light on this?
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nandy1925 at gmail.com  Mon Aug 26 07:41:22 2013
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Mon, 26 Aug 2013 11:41:22 +0800
Subject: [Freeswitch-users] Call drops when attended transfer is being
	performed
In-Reply-To: <BLU174-W48D3A69832BBCDD51976C19D480@phx.gbl>
References: <BLU174-W48D3A69832BBCDD51976C19D480@phx.gbl>
Message-ID: <CAHKqMb6b0nYHd7e4ZkUd026gedSubZPcHc0oFRnRBJUKFhf57g@mail.gmail.com>

I think you have to wait for Party #3 to pickup the the call in step 5. You
will hear a busy tone. Just hangup the phone - no need to press Transfer
again.

/Nandy



On Mon, Aug 26, 2013 at 2:02 AM, John Platts <john_platts at hotmail.com>wrote:

> Whenever I do an attended transfer a call, the caller gets dropped instead
> of being transferred to the next party. Here are the steps to reproducing
> the problem:
>
>    1. Make a call to a phone that is registered with the FreeSWITCH
>    server.
>    2. Answer the call.
>    3. Hit the transfer button on the phone that the call rings on to
>    initiate an attended transfer.
>    4. Initiate the call to party #3, which is the party that the call
>    needs to be transferred to.
>    5. Once the call to party #3 is made, hit the transfer button again to
>    complete the transfer.
>    6. The caller should now be transferred to party #3, but the call
>    drops instead.
>
>
> I am running FreeSWITCH 1.2.12 on the switch where I am experiencing the
> attended transfer problem. How do I get the call to get transferred from
> the caller to party #3, and how do I prevent the caller from getting
> disconnected when a call is transferred?
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nandy1925 at gmail.com  Mon Aug 26 07:45:17 2013
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Mon, 26 Aug 2013 11:45:17 +0800
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
	<CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
	<CALXCt0rb7A8wXHFwNmMOLEYiEwer234NNsTaNGfjw5hneQUdFQ@mail.gmail.com>
	<CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>
Message-ID: <CAHKqMb7p3m_cS0AeA2z_EL_o20eA5b3HrugPM=-KNN7FGeVibw@mail.gmail.com>

It should because the switchmode switches the dongle from appearing as a
CD-ROM drive to a modem.


On Sun, Aug 25, 2013 at 6:40 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Thank you so much..after this will my dongle be easily detected by FS..???
> On Aug 25, 2013 4:09 PM, "Giovanni Maruzzelli" <gmaruzz at celliax.org>
> wrote:
>
>> apt-get install usb-modeswitch-data usb-modeswitch
>>
>>
>>
>> On Sun, Aug 25, 2013 at 12:26 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> Thank you,
>>> Which prerequisites are you talking about...??? The prerequisites for
>>> mod_gsmopen? I have followed the wiki page that entails the prerequisites
>>> and perhaps to some extent i feel that i have installed all the
>>> prerequisites. But if you guys i will give it another shot.
>>>
>>> Regards
>>> Ashish Mishra
>>> On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> you have probably not installed all the prerequisites as per the wiki
>>>> page.
>>>> Your linux does not see the dongle.
>>>> Please follow the wiki page step by step
>>>>
>>>>
>>>>
>>>> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>
>>>>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>>>>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> Hi jayanth,
>>>>>> This is the new thread that i am starting. I am using the dongle
>>>>>> E1732 huawei which works well as far as receiving and dialing the calls is
>>>>>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 GB at cm.nl  Mon Aug 26 10:26:21 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Mon, 26 Aug 2013 08:26:21 +0200
Subject: [Freeswitch-users] Tuning DTMF
In-Reply-To: <CAHgk4BWrESZ=cnZonBHTPQr0r3aUzOYZHJpPGOmzsA3xMoc-Rg@mail.gmail.com>
References: <FB7D97A214987F458242ACBDF87614073B1BD19CD6@clubvirtual40.ClubMessage.local>
	<CAHgk4BWrESZ=cnZonBHTPQr0r3aUzOYZHJpPGOmzsA3xMoc-Rg@mail.gmail.com>
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD19E26@clubvirtual40.ClubMessage.local>

Yes, that works fine. The cracking DTMF tones only happen when the remote party is sending DTMF tones. So if I call my cellphone with X-Lite through FS and my cellphones starts sending DTMF tones, that's when the tones come in with some sort of cracking sound.

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster
Sent: Saturday, August 24, 2013 3:13 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Tuning DTMF


Have you tried using Xlite on an IVR or something? Call your bank or your cable company and see if you can get through the menus fine. I'd bet that you can. Post back and let us know. They sound short but that should be ok.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com<mailto:bdfoster at davri.com>
Indianapolis, Indiana

Sent from a mobile device.
On Aug 23, 2013 3:21 AM, "Grant Bagdasarian" <GB at cm.nl<mailto:GB at cm.nl>> wrote:
Hello,

Setup is as following:

X-Lite (Test phone) ------>  SIP Proxy ------> FS (as SBC) ------> Carrier ------> End-user phone

Whenever I place a call using X-Lite to the End-user phone, in this case my own mobile phone or office phone, and start sending DTMF tones from the remote telephone, the DMTF tones arrive with a crackling sound at the end of the tone.

Console debug:
2013-08-23 09:00:13.813207 [DEBUG] switch_rtp.c:3829 Send start packet for [2] ts=3778476449 dur=160/160/1280 seq=5098 lw=-516490847
2013-08-23 09:00:13.833208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=320/320/1280<tel:320%2F320%2F1280> seq=5099 lw=161
2013-08-23 09:00:13.853208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=480/480/1280<tel:480%2F480%2F1280> seq=5100 lw=161
2013-08-23 09:00:13.873208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=640/640/1280 seq=5101 lw=161
2013-08-23 09:00:13.893208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=800/800/1280<tel:800%2F800%2F1280> seq=5102 lw=161
2013-08-23 09:00:13.913208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=960/960/1280 seq=5103 lw=161
2013-08-23 09:00:13.933208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=1120/1120/1280 seq=5104 lw=161
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5105 lw=1
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5106 lw=1
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5107 lw=1
2013-08-23 09:00:14.813206 [DEBUG] switch_rtp.c:5117 RTP RECV DTMF 1:1280
2013-08-23 09:00:14.813206 [DEBUG] switch_channel.c:471 RECV DTMF 1:1280

Both internal and external sip profiles have the following configuration for DTMF:
<param name="liberal-dtmf" value="true"/>
<param name="rfc2833-pt" value="101"/>
<param name="dtmf-duration" value="2000"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="inbound-late-negotiation" value="true"/>

Internal.xml (Dialplan)
    <extension name="internal_to_external">
      <condition field="${sip_req_host}" expression="^10\.0\.0\.1$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="sofia/external/${sip_req_uri}"/>
      </condition>
    </extension>

Is there a way to tune the DTMF tones? If so, which parameters do I need to add or edit?

Thanks!

Grant


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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
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From GB at cm.nl  Mon Aug 26 10:35:24 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Mon, 26 Aug 2013 08:35:24 +0200
Subject: [Freeswitch-users] Remove custom SIP headers
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD19E29@clubvirtual40.ClubMessage.local>

Hello,

So I found this page which shows how to remove custom SIP headers by using the unset application and passing in the custom header name prefixed by sip_h_[HEADER].
Is there a configuration setting which can be set to disallow custom headers being copied when bridging calls?

The scenario is as following;

1)      Each call is going through a Kamailio SIP Proxy first, which adds some custom headers.

2)      Kamailio routes the INVITE to one of our FS SBC's.

3)      FS bridges the call to the carrier.

When FS creates a new call leg towards the carrier, I would expect it to NOT copy the custom headers added by Kamailio, but it does.

This is my dialplan:

    <extension name="internal_to_external">
      <condition field="${sip_req_host}" expression="^10\.0\.0\.1$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="privacy" data="full"/>
        <action application="set" data="sip_h_Privacy=id"/>
        <action application="set" data="privacy=yes"/>
        <action application="bridge" data="sofia/external/${sip_req_uri}"/>
      </condition>
    </extension>

Did I configure something wrong, or is this just normal behavior?
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From gmangudai at gmail.com  Mon Aug 26 11:42:26 2013
From: gmangudai at gmail.com (Vincent Xia)
Date: Mon, 26 Aug 2013 15:42:26 +0800
Subject: [Freeswitch-users] display the expected caller id and name
Message-ID: <CAKQo6tk7pGSqQ+QPbuuwf09OZHY_NFB5KME+fWPnQVX6kbkC4A@mail.gmail.com>

while making call from FS console to user 1005 and 1006 with

originate sofia/internal/1005%192.168.1.35 &bridge(user/1006) XML default
101 102

the displayed caller id and name at 1005 is 101 and 102, at 1006 it's
Outbound Call and 1005%192.168.1.35(but %19 here is recognaized as an ASCII
SYMBOL and the display looks like 1005?2.168.1.35 and it's not
user-friendly), so im trying to maniuplate the caller id name and number, as

originate {origination_caller_id_name=123}sofia/internal/1005%192.168.1.35
&bridge(user/1006) XML default 101 102

but this only affects the display at 1005, how can i have both 1005 and
1006 to display the expected caller id and name.
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From khorsmann at gmail.com  Mon Aug 26 11:54:06 2013
From: khorsmann at gmail.com (Karsten Horsmann)
Date: Mon, 26 Aug 2013 09:54:06 +0200
Subject: [Freeswitch-users] Remove custom SIP headers
In-Reply-To: <FB7D97A214987F458242ACBDF87614073B1BD19E29@clubvirtual40.ClubMessage.local>
References: <FB7D97A214987F458242ACBDF87614073B1BD19E29@clubvirtual40.ClubMessage.local>
Message-ID: <CAFArqsbAMfavS+sAw+qqsR_PoPxc5hfG7JnyxgQBDMOp9X8vyA@mail.gmail.com>

Hi Grant,


IMHO there is no way dont copy the sip-x header to the new call-leg.

The simple way to resolve this, is to remove your kamailio sip-x headers
after you read it and before you create the b-leg.

http://wiki.freeswitch.org/wiki/Strip_SIP_Headers



2013/8/26 Grant Bagdasarian <GB at cm.nl>

> Hello,****
>
> ** **
>
> So I found this page which shows how to remove custom SIP headers by using
> the unset application and passing in the custom header name prefixed by
> sip_h_[HEADER].****
>
> Is there a configuration setting which can be set to disallow custom
> headers being copied when bridging calls? ****
>
> ** **
>
> The scenario is as following; ****
>
> **1)      **Each call is going through a Kamailio SIP Proxy first, which
> adds some custom headers.****
>
> **2)      **Kamailio routes the INVITE to one of our FS SBC?s.****
>
> **3)      **FS bridges the call to the carrier.****
>
> ** **
>
> When FS creates a new call leg towards the carrier, I would expect it to
> NOT copy the custom headers added by Kamailio, but it does. ****
>
> ** **
>
> This is my dialplan:****
>
> ** **
>
>     <extension name="internal_to_external">****
>
>       <condition field="${sip_req_host}" expression="^10\.0\.0\.1$">****
>
>         <action application="set" data="hangup_after_bridge=false"/>****
>
>         <action application="set" data="continue_on_fail=true"/>****
>
>         <action application="privacy" data="full"/>****
>
>         <action application="set" data="sip_h_Privacy=id"/>****
>
>         <action application="set" data="privacy=yes"/>****
>
>         <action application="bridge" data="sofia/external/${sip_req_uri}"/>
> ****
>
>       </condition>****
>
>     </extension>****
>
> ** **
>
> Did I configure something wrong, or is this just normal behavior?****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Mit freundlichen Gr??en
*Karsten Horsmann*
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From GB at cm.nl  Mon Aug 26 12:33:47 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Mon, 26 Aug 2013 10:33:47 +0200
Subject: [Freeswitch-users] Remove custom SIP headers
In-Reply-To: <CAFArqsbAMfavS+sAw+qqsR_PoPxc5hfG7JnyxgQBDMOp9X8vyA@mail.gmail.com>
References: <FB7D97A214987F458242ACBDF87614073B1BD19E29@clubvirtual40.ClubMessage.local>
	<CAFArqsbAMfavS+sAw+qqsR_PoPxc5hfG7JnyxgQBDMOp9X8vyA@mail.gmail.com>
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD19E98@clubvirtual40.ClubMessage.local>

Hello,

Yeah, that's the page I was talking about. I also found this: http://wiki.freeswitch.org/wiki/Variable_sip_copy_custom_headers

This action works. It strips all the custom headers. It would have been nice if I could configure this in a sip_profile.
<action application="set" data="sip_copy_custom_headers=false" />

Is the sip_copy_custom_headers variable a channel variable or can it also be set in a sip_profile?


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Karsten Horsmann
Sent: Monday, August 26, 2013 9:54 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Remove custom SIP headers

Hi Grant,

IMHO there is no way dont copy the sip-x header to the new call-leg.
The simple way to resolve this, is to remove your kamailio sip-x headers after you read it and before you create the b-leg.

http://wiki.freeswitch.org/wiki/Strip_SIP_Headers

2013/8/26 Grant Bagdasarian <GB at cm.nl<mailto:GB at cm.nl>>
Hello,

So I found this page which shows how to remove custom SIP headers by using the unset application and passing in the custom header name prefixed by sip_h_[HEADER].
Is there a configuration setting which can be set to disallow custom headers being copied when bridging calls?

The scenario is as following;

1)      Each call is going through a Kamailio SIP Proxy first, which adds some custom headers.

2)      Kamailio routes the INVITE to one of our FS SBC's.

3)      FS bridges the call to the carrier.

When FS creates a new call leg towards the carrier, I would expect it to NOT copy the custom headers added by Kamailio, but it does.

This is my dialplan:

    <extension name="internal_to_external">
      <condition field="${sip_req_host}" expression="^10\.0\.0\.1$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="privacy" data="full"/>
        <action application="set" data="sip_h_Privacy=id"/>
        <action application="set" data="privacy=yes"/>
        <action application="bridge" data="sofia/external/${sip_req_uri}"/>
      </condition>
    </extension>

Did I configure something wrong, or is this just normal behavior?

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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



--
Mit freundlichen Gr??en
*Karsten Horsmann*
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From GB at cm.nl  Mon Aug 26 13:01:28 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Mon, 26 Aug 2013 11:01:28 +0200
Subject: [Freeswitch-users] FS Caller id behavior
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD19EAB@clubvirtual40.ClubMessage.local>

Hello,

Consider the following scenario:

SIP Server --------> Freeswitch SBC --------> Carrier

The SIP Server originates a call through FS SBC to the Carrier. FS bridges A and B Legs.
The caller id is set by the SIP Server:

1)      First attempt with caller id visible

2)      Second attempt with caller id blocked (anonymous)

I tested this, but in both attempts the caller id was shown on the called party's phone.

The Remote-Party-ID of the SIP INVITE (B-Leg: FS to Carrier) in the second attempt was like this, because the A leg INVITE didn't contain a Remote-Party-ID header:
Remote-Party-ID: "anonymous" <sip:31123456789 at 10.0.0.1>;party=calling;screen=yes;privacy=off.

When I configure FS to always dial out anonymously by setting the below variables, the call is bridged anonymously.
        <action application="privacy" data="full"/>
        <action application="set" data="sip_h_Privacy=id"/>
        <action application="set" data="privacy=yes"/>

In the case of dialing anonymously the Remote-Party-ID looks like this:
Remote-Party-ID: "31123456789" <sip:31123456789 at 10.0.0.1>;party=calling;screen=yes;privacy=full.

So I'm assuming the problem lies with the SIP Server which initially initiates the call. It doesn't set the Remote-Party-ID header (correctly).

If it did set it correctly, and I didn't configure the  variables for anonymous dialing in FS, would FS still dial out anonymously? Or will the Remote-Party-ID be reset by FS?
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From steveayre at gmail.com  Mon Aug 26 14:16:59 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Mon, 26 Aug 2013 11:16:59 +0100
Subject: [Freeswitch-users] display the expected caller id and name
In-Reply-To: <CAKQo6tk7pGSqQ+QPbuuwf09OZHY_NFB5KME+fWPnQVX6kbkC4A@mail.gmail.com>
References: <CAKQo6tk7pGSqQ+QPbuuwf09OZHY_NFB5KME+fWPnQVX6kbkC4A@mail.gmail.com>
Message-ID: <CAFiqYukxLrVd-NuYU=BsOXm5E_AGBd=eP+Q_CUUZF-eXfFPfsA@mail.gmail.com>

Try also setting effective_callerid_name to set the CallerID on the B-leg,
and perhaps the _number variants too.


On 26 August 2013 08:42, Vincent Xia <gmangudai at gmail.com> wrote:

> while making call from FS console to user 1005 and 1006 with
>
> originate sofia/internal/1005%192.168.1.35 &bridge(user/1006) XML default
> 101 102
>
> the displayed caller id and name at 1005 is 101 and 102, at 1006 it's
> Outbound Call and 1005%192.168.1.35(but %19 here is recognaized as an ASCII
> SYMBOL and the display looks like 1005?2.168.1.35 and it's not
> user-friendly), so im trying to maniuplate the caller id name and number, as
>
> originate {origination_caller_id_name=123}sofia/internal/1005%192.168.1.35
> &bridge(user/1006) XML default 101 102
>
> but this only affects the display at 1005, how can i have both 1005 and
> 1006 to display the expected caller id and name.
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 manuel at aguete.org  Mon Aug 26 14:42:55 2013
From: manuel at aguete.org (=?UTF-8?Q?Manuel_Dur=C3=A1n_Aguete?=)
Date: Mon, 26 Aug 2013 12:42:55 +0200
Subject: [Freeswitch-users] Capture DTMF and play and detect speech in Lua
Message-ID: <CADfSCeTpDVsFRSELq=fQhTnDCrRfHrC1zv95i4EjT3P8toxRpQ@mail.gmail.com>

Hello,

I'm trying to capture DTMF when doing play_and_detect_speech:

function capture_dtmf(session,input_type,data,arg)
        if input_type == "dtmf" then
                freeswitch.consoleLog("CRIT","Key pressed: " ..
data["digit"] .. "\n")
        end
end

session:answer()
session:sleep(1000)
session:setInputCallback("capture_dtmf","true")

session:execute("play_and_detect_speech",
"phrase:wellcome detect:unimrcp
{start-input-timers=false,no-input-timeout=5000,recognition-timeout=5000,
speech-language=es-ES}builtin:grammar/digits")


xml = session:getVariable('detect_speech_result')
if xml ~= nil then
        freeswitch.consoleLog("CRIT", xml .."\n")
else
        freeswitch.consoleLog("CRIT", "No result!\n")
end

The asr seems to work, but when no dtmf is captured in capture_dtmf. Is
DTMF capture with input callback supported by play_and_detect_speech ? Need
to use detect_speech ?

Thank you. Regards.
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From gmangudai at gmail.com  Mon Aug 26 15:33:34 2013
From: gmangudai at gmail.com (Vincent Xia)
Date: Mon, 26 Aug 2013 19:33:34 +0800
Subject: [Freeswitch-users] is it possible to have FS forcilby unregiter
 a registered UA
In-Reply-To: <CAKQo6t=BQHZPigtGCDg=O_eMW=F2GFDxsaPZOem=Uf6sZ98RqQ@mail.gmail.com>
References: <CAKQo6tnkgwmneo3QiG0cnBZF5LsApapsYZ4o778fT9SH+5SUGg@mail.gmail.com>
	<CAFiqYukMVqxHhs6rwL7QKZBn0Y-bZ+s3bWqAfUHOdR=33iWqUQ@mail.gmail.com>
	<CAKQo6t=BQHZPigtGCDg=O_eMW=F2GFDxsaPZOem=Uf6sZ98RqQ@mail.gmail.com>
Message-ID: <CAKQo6t=N+GG+faiNLQM2SofGnuboOnSDx=4=x0B02E5TnO4aaQ@mail.gmail.com>

this is what i see from
http://wiki.freeswitch.org/wiki/Sofia-SIP#Flushing_and_rebooting_registered_endpoints

sofia profile <profile_name> flush_inbound_reg [<call_id>|<user at host>] [reboot]

but, what indeed is call_id??? to unregister the default user 1006, i tried
sofia profile internal flush_inbound_reg 1006
sofia profile internal flush_inbound_reg 1006 at xxx.xxx.xxx.xxx
sofia profile internal flush_inbound_reg user/1006

but seems nothing is working, the reg is still there.
sofia profile internal flush_inbound_reg will do but all endpoints get
unreged.



2013/8/26 Vincent Xia <gmangudai at gmail.com>

> yes from the same user, but i have 2 profiles, 1 for LAN and 1 for
> INTERNET, and i don't want 1 user to register both in LAN and INTERNET, so
> im considering to kill the previous registration once i got register
> request from the same user but a different profile.
>
>
> 2013/8/24 Steven Ayre <steveayre at gmail.com>
>
>> to go one step further, how can i do the unregister upon receiving
>>> another register request?
>>
>>
>> From the same user? By default the user can only be registered at one
>> location.
>>
>> Bear in mind registrations are timeout based, and refreshed by sending
>> further REGISTER packets periodically.
>>
>> That means if you have 2 phones using the same user details the 2nd will
>> register and replace the 1st, then the 1st reregister and replace the 2nd,
>> then the 2nd reregister and replace the 1st...
>>
>> You can't stop them reregistering. Probably wouldn't want to either, or
>> you'll stop blocking legitimate reregistrations, such as when a phone moves
>> location.
>>
>> Not something you can do much about. You shouldn't use the same account
>> in multiple places. Or allow them to register from multiple locations at
>> once (
>> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#multiple-registrations)
>> but bear in mind that'll affect anything that expects them in only one
>> place.
>>
>> -Steve
>>
>>
>>
>> On 23 August 2013 11:52, Vincent Xia <gmangudai at gmail.com> wrote:
>>
>>> is it possible to have FS forcilby unregiter a registered UA, for
>>> example, user/1001?
>>>
>>> to go one step further, how can i do the unregister upon receiving
>>> another register request?
>>> i checked event list of the event socket but cannot find any register
>>> event, can anyone shed any light on this?
>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 fs.user at fordior.net  Mon Aug 26 15:48:47 2013
From: fs.user at fordior.net (EL)
Date: Mon, 26 Aug 2013 13:48:47 +0200
Subject: [Freeswitch-users] is it possible to have FS forcilby unregiter
 a registered UA
In-Reply-To: <CAKQo6t=N+GG+faiNLQM2SofGnuboOnSDx=4=x0B02E5TnO4aaQ@mail.gmail.com>
References: <CAKQo6tnkgwmneo3QiG0cnBZF5LsApapsYZ4o778fT9SH+5SUGg@mail.gmail.com>
	<CAFiqYukMVqxHhs6rwL7QKZBn0Y-bZ+s3bWqAfUHOdR=33iWqUQ@mail.gmail.com>
	<CAKQo6t=BQHZPigtGCDg=O_eMW=F2GFDxsaPZOem=Uf6sZ98RqQ@mail.gmail.com>
	<CAKQo6t=N+GG+faiNLQM2SofGnuboOnSDx=4=x0B02E5TnO4aaQ@mail.gmail.com>
Message-ID: <20130826114847.GB21825@0rdior.com>

Like I said:

sofia profile default flush_inbound_reg 1002 at domain

> but, what indeed is call_id??? to unregister the default user 1006....

See:
sofia status profile internal reg

and you will see registrations and the relevant fields.


-- 
EL


From vipkilla at gmail.com  Mon Aug 26 16:12:50 2013
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 26 Aug 2013 08:12:50 -0400
Subject: [Freeswitch-users] Pacemaker Resource Agent
In-Reply-To: <CAMbW9oLarp-soSycAtDXpo_eJuV0PMdRhgZ6JjSNcrDzCEtUDA@mail.gmail.com>
References: <CAMbW9oJ3F_K3nMz0qKj9KFfYq_ZLB-buLkYAOu9s6xNooq6f1Q@mail.gmail.com>
	<00000140b257ea79-cfb47be2-f8c6-4466-86a0-c6bde0cc4ec9-000000@email.amazonses.com>
	<CAMbW9o+hQYHDNVCtTbB-3RSDBSDU8FA-9+SAf3Vs6qbvYA-EMw@mail.gmail.com>
	<CAMbW9oL7HYmRgxNBJz-DkDn5xF5X1cejz9LD5pYmusMYrrQKMA@mail.gmail.com>
	<CAMbW9oLarp-soSycAtDXpo_eJuV0PMdRhgZ6JjSNcrDzCEtUDA@mail.gmail.com>
Message-ID: <CAC-LwPN9EyFH6MLWqgH+W+kUwvYE1WquFmEur69R-7c5L8NXTw@mail.gmail.com>

I wrote both those wiki pages:
http://wiki.freeswitch.org/wiki/Enterprise_deployment_IP_Failover
http://wiki.freeswitch.org/wiki/Enterprise_deployment_with_Corosync

I only tested them on Debian Squeeze -- they both worked on Squeeze.
Please feel free to update the wiki for Debian Wheezy or any other distro.

<snip>
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From itsme.kunnu at gmail.com  Mon Aug 26 17:17:51 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Mon, 26 Aug 2013 18:47:51 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAHKqMb7p3m_cS0AeA2z_EL_o20eA5b3HrugPM=-KNN7FGeVibw@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
	<CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
	<CALXCt0rb7A8wXHFwNmMOLEYiEwer234NNsTaNGfjw5hneQUdFQ@mail.gmail.com>
	<CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>
	<CAHKqMb7p3m_cS0AeA2z_EL_o20eA5b3HrugPM=-KNN7FGeVibw@mail.gmail.com>
Message-ID: <CAMvhjJtYBsgCQzmWW5qXj80jWNd5Vu4p3EOi4FUO0u_3d6xBcA@mail.gmail.com>

Hi
I reinstalled all the prerequisites for gsmopen but still the same problem
again my dongle with sim is not recognised by ubuntu nor by freeswitch.
Also when i am typing wvdialconf the terminal shows that no modem detected.
On Aug 26, 2013 9:19 AM, "Nandy Dagondon" <nandy1925 at gmail.com> wrote:

> It should because the switchmode switches the dongle from appearing as a
> CD-ROM drive to a modem.
>
>
> On Sun, Aug 25, 2013 at 6:40 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Thank you so much..after this will my dongle be easily detected by FS..???
>> On Aug 25, 2013 4:09 PM, "Giovanni Maruzzelli" <gmaruzz at celliax.org>
>> wrote:
>>
>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>
>>>
>>>
>>> On Sun, Aug 25, 2013 at 12:26 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> Thank you,
>>>> Which prerequisites are you talking about...??? The prerequisites for
>>>> mod_gsmopen? I have followed the wiki page that entails the prerequisites
>>>> and perhaps to some extent i feel that i have installed all the
>>>> prerequisites. But if you guys i will give it another shot.
>>>>
>>>> Regards
>>>> Ashish Mishra
>>>> On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
>>>> wrote:
>>>>
>>>>> you have probably not installed all the prerequisites as per the wiki
>>>>> page.
>>>>> Your linux does not see the dongle.
>>>>> Please follow the wiki page step by step
>>>>>
>>>>>
>>>>>
>>>>> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <itsme.kunnu at gmail.com
>>>>> > wrote:
>>>>>
>>>>>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>>>>>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> Hi jayanth,
>>>>>>> This is the new thread that i am starting. I am using the dongle
>>>>>>> E1732 huawei which works well as far as receiving and dialing the calls is
>>>>>>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> Cell : +39-347-2665618
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 valter at fastway.com.br  Mon Aug 26 17:56:25 2013
From: valter at fastway.com.br (Valter Nogueira)
Date: Mon, 26 Aug 2013 10:56:25 -0300
Subject: [Freeswitch-users] Porting C AGI from Asterisk to Freeswitch
Message-ID: <CAGu7vdamGGFNjDCLnr-OLfzi9yeKv4DkkmHhJG+tuC6ST_ZGXw@mail.gmail.com>

I have several agi's written in C which I would like to port to Freeswitch.

What is the best approach? Buld it as a freeswitch module?

Thanks,

Valter
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From vipkilla at gmail.com  Mon Aug 26 18:10:11 2013
From: vipkilla at gmail.com (Vik Killa)
Date: Mon, 26 Aug 2013 10:10:11 -0400
Subject: [Freeswitch-users] Porting C AGI from Asterisk to Freeswitch
In-Reply-To: <CAGu7vdamGGFNjDCLnr-OLfzi9yeKv4DkkmHhJG+tuC6ST_ZGXw@mail.gmail.com>
References: <CAGu7vdamGGFNjDCLnr-OLfzi9yeKv4DkkmHhJG+tuC6ST_ZGXw@mail.gmail.com>
Message-ID: <CAC-LwPPPuEAO5YdLBVJf-CLkabt8QD2VDpPgx2BcO64GRrGAug@mail.gmail.com>

Yes, building a module is the best way if you want to keep your code in C
<snip>
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From fs.user at fordior.net  Mon Aug 26 18:37:34 2013
From: fs.user at fordior.net (EL)
Date: Mon, 26 Aug 2013 16:37:34 +0200
Subject: [Freeswitch-users] is it possible to have FS forcilby unregiter
 a registered UA
In-Reply-To: <CAKQo6t=BQHZPigtGCDg=O_eMW=F2GFDxsaPZOem=Uf6sZ98RqQ@mail.gmail.com>
References: <CAKQo6tnkgwmneo3QiG0cnBZF5LsApapsYZ4o778fT9SH+5SUGg@mail.gmail.com>
	<CAFiqYukMVqxHhs6rwL7QKZBn0Y-bZ+s3bWqAfUHOdR=33iWqUQ@mail.gmail.com>
	<CAKQo6t=BQHZPigtGCDg=O_eMW=F2GFDxsaPZOem=Uf6sZ98RqQ@mail.gmail.com>
Message-ID: <20130826143734.GB22574@0rdior.com>

> yes from the same user, but i have 2 profiles, 1 for LAN and 1 for
> INTERNET, and i don't want 1 user to register both in LAN and INTERNET, so
> im considering to kill the previous registration once i got register
> request from the same user but a different profile.

I guess the 'internet' side would be for your mobile phone (while both
are the same user)? I think it's better to implement a vpn. Or do you
really want to allow the internetside (public profile) to make calls
through your box? 

Anyway, there is a security warning for this; see section 'External' here:
https://wiki.freeswitch.org/wiki/Getting_Started_Guide#What_SIP_Profiles_Do 

-- 
EL


From soeren.sprenger at aerea.de  Mon Aug 26 18:50:27 2013
From: soeren.sprenger at aerea.de (=?ISO-8859-1?Q?S=F6ren_Sprenger?=)
Date: Mon, 26 Aug 2013 16:50:27 +0200
Subject: [Freeswitch-users] Porting C AGI from Asterisk to Freeswitch
In-Reply-To: <CAGu7vdamGGFNjDCLnr-OLfzi9yeKv4DkkmHhJG+tuC6ST_ZGXw@mail.gmail.com>
References: <CAGu7vdamGGFNjDCLnr-OLfzi9yeKv4DkkmHhJG+tuC6ST_ZGXw@mail.gmail.com>
Message-ID: <521B6B33.1040900@aerea.de>

Hi,

you could write an application for the outbound event socket mode and
call that application from the dialplan using "socket".
Take a look here:
http://wiki.freeswitch.org/wiki/Mod_event_socket
and here:
http://wiki.freeswitch.org/wiki/Event_socket_outbound

That's the way i've done similar stuff. This makes small amount of
overhead and is more like the asterisk's FastAGI approach instead of the
normal AGI but this was for me the better solution + debugging is very easy.

  Soeren

On 26.08.2013 15:56, Valter Nogueira wrote:
> I have several agi's written in C which I would like to port to
> Freeswitch.
>
> What is the best approach? Buld it as a freeswitch module?


-- 
AereA NetworX UG (haftungsbeschr?nkt)
Im Camisch 8
07768 Kahla
Gesch?ftsf?hrerin: Franziska Sprenger
Handelsregister: Jena B504724
E-Mail: info at aerea.de
Web: http://www.aerea.de
Fon: +49 (0) 36424 760823
Fax: +49 (0) 36651 1390009

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From jayachar88 at gmail.com  Mon Aug 26 18:53:32 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Mon, 26 Aug 2013 20:23:32 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAMvhjJtYBsgCQzmWW5qXj80jWNd5Vu4p3EOi4FUO0u_3d6xBcA@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
	<CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
	<CALXCt0rb7A8wXHFwNmMOLEYiEwer234NNsTaNGfjw5hneQUdFQ@mail.gmail.com>
	<CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>
	<CAHKqMb7p3m_cS0AeA2z_EL_o20eA5b3HrugPM=-KNN7FGeVibw@mail.gmail.com>
	<CAMvhjJtYBsgCQzmWW5qXj80jWNd5Vu4p3EOi4FUO0u_3d6xBcA@mail.gmail.com>
Message-ID: <CAHDL-sznAVmo49jyJx3J1NMcR8w=Ca5vfBTBeD4U3g--e75UOA@mail.gmail.com>

Including this --

apt-get install usb-modeswitch-data usb-modeswitch


 ??
Then, you have restarted your PC ?


On Mon, Aug 26, 2013 at 6:47 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Hi
> I reinstalled all the prerequisites for gsmopen but still the same problem
> again my dongle with sim is not recognised by ubuntu nor by freeswitch.
> Also when i am typing wvdialconf the terminal shows that no modem detected.
> On Aug 26, 2013 9:19 AM, "Nandy Dagondon" <nandy1925 at gmail.com> wrote:
>
>> It should because the switchmode switches the dongle from appearing as a
>> CD-ROM drive to a modem.
>>
>>
>> On Sun, Aug 25, 2013 at 6:40 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> Thank you so much..after this will my dongle be easily detected by
>>> FS..???
>>> On Aug 25, 2013 4:09 PM, "Giovanni Maruzzelli" <gmaruzz at celliax.org>
>>> wrote:
>>>
>>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>>
>>>>
>>>>
>>>> On Sun, Aug 25, 2013 at 12:26 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>
>>>>> Thank you,
>>>>> Which prerequisites are you talking about...??? The prerequisites for
>>>>> mod_gsmopen? I have followed the wiki page that entails the prerequisites
>>>>> and perhaps to some extent i feel that i have installed all the
>>>>> prerequisites. But if you guys i will give it another shot.
>>>>>
>>>>> Regards
>>>>> Ashish Mishra
>>>>> On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
>>>>> wrote:
>>>>>
>>>>>> you have probably not installed all the prerequisites as per the wiki
>>>>>> page.
>>>>>> Your linux does not see the dongle.
>>>>>> Please follow the wiki page step by step
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <
>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>
>>>>>>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>>>>>>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> Hi jayanth,
>>>>>>>> This is the new thread that i am starting. I am using the dongle
>>>>>>>> E1732 huawei which works well as far as receiving and dialing the calls is
>>>>>>>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Sincerely,
>>>>>>
>>>>>> Giovanni Maruzzelli
>>>>>> Cell : +39-347-2665618
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Mon Aug 26 20:20:18 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Mon, 26 Aug 2013 21:50:18 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAHDL-sznAVmo49jyJx3J1NMcR8w=Ca5vfBTBeD4U3g--e75UOA@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
	<CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
	<CALXCt0rb7A8wXHFwNmMOLEYiEwer234NNsTaNGfjw5hneQUdFQ@mail.gmail.com>
	<CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>
	<CAHKqMb7p3m_cS0AeA2z_EL_o20eA5b3HrugPM=-KNN7FGeVibw@mail.gmail.com>
	<CAMvhjJtYBsgCQzmWW5qXj80jWNd5Vu4p3EOi4FUO0u_3d6xBcA@mail.gmail.com>
	<CAHDL-sznAVmo49jyJx3J1NMcR8w=Ca5vfBTBeD4U3g--e75UOA@mail.gmail.com>
Message-ID: <CAMvhjJu10VtF=X9285ST4SicA=-=iXGMvGxFcUOsaqgzQr6vaQ@mail.gmail.com>

yes i did restart my pc but still my HUAWEI E1732 dongle not getting
recognised by freeSwitch.. :(


On Mon, Aug 26, 2013 at 8:23 PM, Jayanth Acharya <jayachar88 at gmail.com>wrote:

> Including this --
>
> apt-get install usb-modeswitch-data usb-modeswitch
>
>
>  ??
> Then, you have restarted your PC ?
>
>
> On Mon, Aug 26, 2013 at 6:47 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Hi
>> I reinstalled all the prerequisites for gsmopen but still the same
>> problem again my dongle with sim is not recognised by ubuntu nor by
>> freeswitch.
>> Also when i am typing wvdialconf the terminal shows that no modem
>> detected.
>> On Aug 26, 2013 9:19 AM, "Nandy Dagondon" <nandy1925 at gmail.com> wrote:
>>
>>> It should because the switchmode switches the dongle from appearing as a
>>> CD-ROM drive to a modem.
>>>
>>>
>>> On Sun, Aug 25, 2013 at 6:40 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> Thank you so much..after this will my dongle be easily detected by
>>>> FS..???
>>>> On Aug 25, 2013 4:09 PM, "Giovanni Maruzzelli" <gmaruzz at celliax.org>
>>>> wrote:
>>>>
>>>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>>>
>>>>>
>>>>>
>>>>> On Sun, Aug 25, 2013 at 12:26 PM, Ashish Mishra <itsme.kunnu at gmail.com
>>>>> > wrote:
>>>>>
>>>>>> Thank you,
>>>>>> Which prerequisites are you talking about...??? The prerequisites for
>>>>>> mod_gsmopen? I have followed the wiki page that entails the prerequisites
>>>>>> and perhaps to some extent i feel that i have installed all the
>>>>>> prerequisites. But if you guys i will give it another shot.
>>>>>>
>>>>>> Regards
>>>>>> Ashish Mishra
>>>>>> On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> you have probably not installed all the prerequisites as per the
>>>>>>> wiki page.
>>>>>>> Your linux does not see the dongle.
>>>>>>> Please follow the wiki page step by step
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <
>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>
>>>>>>>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>>>>>>>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> Hi jayanth,
>>>>>>>>> This is the new thread that i am starting. I am using the dongle
>>>>>>>>> E1732 huawei which works well as far as receiving and dialing the calls is
>>>>>>>>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Sincerely,
>>>>>>>
>>>>>>> Giovanni Maruzzelli
>>>>>>> Cell : +39-347-2665618
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> Cell : +39-347-2665618
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 gmail.com  Mon Aug 26 20:48:36 2013
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 26 Aug 2013 18:48:36 +0200
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAMvhjJu10VtF=X9285ST4SicA=-=iXGMvGxFcUOsaqgzQr6vaQ@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
	<CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
	<CALXCt0rb7A8wXHFwNmMOLEYiEwer234NNsTaNGfjw5hneQUdFQ@mail.gmail.com>
	<CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>
	<CAHKqMb7p3m_cS0AeA2z_EL_o20eA5b3HrugPM=-KNN7FGeVibw@mail.gmail.com>
	<CAMvhjJtYBsgCQzmWW5qXj80jWNd5Vu4p3EOi4FUO0u_3d6xBcA@mail.gmail.com>
	<CAHDL-sznAVmo49jyJx3J1NMcR8w=Ca5vfBTBeD4U3g--e75UOA@mail.gmail.com>
	<CAMvhjJu10VtF=X9285ST4SicA=-=iXGMvGxFcUOsaqgzQr6vaQ@mail.gmail.com>
Message-ID: <CALXCt0pO8HCZE0QyhdP_CmQ4qkg2d4ziuvB+5n_mzNyk2ydxcw@mail.gmail.com>

you probably have some hardware problem, or some Linux install problem.

Try reinstalling Linux on the machine (eg: reformat the hard disk,
reinstall Ubuntu server  64 bit from cdrom), then reinstall freeswitch and
gsmopen.
If this does not work, abandon.

-giovanni



On Mon, Aug 26, 2013 at 6:20 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> yes i did restart my pc but still my HUAWEI E1732 dongle not getting
> recognised by freeSwitch.. :(
>
>
> On Mon, Aug 26, 2013 at 8:23 PM, Jayanth Acharya <jayachar88 at gmail.com>wrote:
>
>> Including this --
>>
>>
>> apt-get install usb-modeswitch-data usb-modeswitch
>>
>>
>>  ??
>> Then, you have restarted your PC ?
>>
>>
>> On Mon, Aug 26, 2013 at 6:47 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> Hi
>>> I reinstalled all the prerequisites for gsmopen but still the same
>>> problem again my dongle with sim is not recognised by ubuntu nor by
>>> freeswitch.
>>> Also when i am typing wvdialconf the terminal shows that no modem
>>> detected.
>>> On Aug 26, 2013 9:19 AM, "Nandy Dagondon" <nandy1925 at gmail.com> wrote:
>>>
>>>> It should because the switchmode switches the dongle from appearing as
>>>> a CD-ROM drive to a modem.
>>>>
>>>>
>>>> On Sun, Aug 25, 2013 at 6:40 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>
>>>>> Thank you so much..after this will my dongle be easily detected by
>>>>> FS..???
>>>>> On Aug 25, 2013 4:09 PM, "Giovanni Maruzzelli" <gmaruzz at celliax.org>
>>>>> wrote:
>>>>>
>>>>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Sun, Aug 25, 2013 at 12:26 PM, Ashish Mishra <
>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>
>>>>>>> Thank you,
>>>>>>> Which prerequisites are you talking about...??? The prerequisites
>>>>>>> for mod_gsmopen? I have followed the wiki page that entails the
>>>>>>> prerequisites and perhaps to some extent i feel that i have installed all
>>>>>>> the prerequisites. But if you guys i will give it another shot.
>>>>>>>
>>>>>>> Regards
>>>>>>> Ashish Mishra
>>>>>>> On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> you have probably not installed all the prerequisites as per the
>>>>>>>> wiki page.
>>>>>>>> Your linux does not see the dongle.
>>>>>>>> Please follow the wiki page step by step
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <
>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>>>>>>>>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com>
>>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>>> Hi jayanth,
>>>>>>>>>> This is the new thread that i am starting. I am using the dongle
>>>>>>>>>> E1732 huawei which works well as far as receiving and dialing the calls is
>>>>>>>>>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Sincerely,
>>>>>>>>
>>>>>>>> Giovanni Maruzzelli
>>>>>>>> Cell : +39-347-2665618
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Sincerely,
>>>>>>
>>>>>> Giovanni Maruzzelli
>>>>>> Cell : +39-347-2665618
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From krice at freeswitch.org  Mon Aug 26 21:14:42 2013
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 26 Aug 2013 12:14:42 -0500
Subject: [Freeswitch-users] Looking for a Digital Artist to help out with a
 small FreeSWITCH video project
Message-ID: <CE40F732.79455%krice@freeswitch.org>

Hey Guys,

I?m looking for someone with Digital Art and Video skills to help out with a
little project to create a FreeSWITCH Video 15 to 20 second Title/lead in
video... Anyone that can help on this or knows someone that might volunteer
a little time to help with this please contact me off list

Thanks!
K

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From mario_fs at mgtech.com  Mon Aug 26 21:01:08 2013
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 26 Aug 2013 10:01:08 -0700
Subject: [Freeswitch-users] Build error Mac OS X Mountain Lion
In-Reply-To: <459CF61A-F314-4318-A9B9-489A5979F03E@gmail.com>
References: <459CF61A-F314-4318-A9B9-489A5979F03E@gmail.com>
Message-ID: <105E3E60-92FF-4BB4-ADBA-ABC0D853BFD0@mgtech.com>

This is the same as Jira FS-4439 which was fixed July 2012. Hope it's not back again. I plan to do a full retest when 10.8.5 is released with the latest Xcode. Did you get past this? If not open a Jira.
Mario G

On Aug 21, 2013, at 2:53 PM, Chris Cachor <ccachor at gmail.com> wrote:

> Hello,
> 
> I'm trying to build Freeswitch on my machine and when I run the make command, I get an error towards the end:
> 
> /var/folders/zr/r7177m510r71w3z81959j9sc0000gn/T//ccUHViiz.s:58:suffix or operands invalid for `lea'
> /var/folders/zr/r7177m510r71w3z81959j9sc0000gn/T//ccUHViiz.s:63:suffix or operands invalid for `movq'
> make[7]: *** [gsm0610_rpe.lo] Error 1
> make[6]: *** [all] Error 2
> make[5]: *** [all-recursive] Error 1
> make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la] Error 2
> make[3]: *** [mod_spandsp-all] Error 1
> make[2]: *** [all-recursive] Error 1
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
> 
> I've built Freeswitch on this same machine about 2 months ago, so not sure why I'm running into this issue. Has anyone had luck with recent builds on OS X Mountain Lion?
> 
> - Chris
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Mon Aug 26 21:24:42 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Mon, 26 Aug 2013 22:54:42 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAMvhjJu10VtF=X9285ST4SicA=-=iXGMvGxFcUOsaqgzQr6vaQ@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
	<CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
	<CALXCt0rb7A8wXHFwNmMOLEYiEwer234NNsTaNGfjw5hneQUdFQ@mail.gmail.com>
	<CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>
	<CAHKqMb7p3m_cS0AeA2z_EL_o20eA5b3HrugPM=-KNN7FGeVibw@mail.gmail.com>
	<CAMvhjJtYBsgCQzmWW5qXj80jWNd5Vu4p3EOi4FUO0u_3d6xBcA@mail.gmail.com>
	<CAHDL-sznAVmo49jyJx3J1NMcR8w=Ca5vfBTBeD4U3g--e75UOA@mail.gmail.com>
	<CAMvhjJu10VtF=X9285ST4SicA=-=iXGMvGxFcUOsaqgzQr6vaQ@mail.gmail.com>
Message-ID: <CAHDL-sw83E0Zr+4j6SrTBdwavzgyxpG3fXSn9NsRcJ9WM1tMwg@mail.gmail.com>

Well in that case... pretty much the end of the line from my end.
I can only make few radical suggestions, like -
1. Try a different Linux distro -- I did my install on Debian 'Wheezy' 7.1
32-bit and have had some success (there's still a lot to do).
2. Try a different PC
3. Try using a powered USB hub... a good quality one, like the ones from
Belkin or D-Link. However, don't even think of the el-cheapo 7-port kinds.
Those are usually horrible... USB1.1 (over-spec'd to say USB2.0, while most
aren't) and with terrible power supplies (400mA stated as 2000mA).
4. If none of those work, try a different dongle.

Like I said... radical! Others might have better ideas. However, not
getting the 3 /dev/ttyUSB interfaces, in spite of usb-modeswitch, beats
me... totally. I have tried half a dozen dongles on my Linux PC's (GSM,
CDMA), and never did I once fail to get them working, once usb-modeswitch
kicked in. Of course, I was using those for pure data-connections.


On Mon, Aug 26, 2013 at 9:50 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> yes i did restart my pc but still my HUAWEI E1732 dongle not getting
> recognised by freeSwitch.. :(
>
>
> On Mon, Aug 26, 2013 at 8:23 PM, Jayanth Acharya <jayachar88 at gmail.com>wrote:
>
>> Including this --
>>
>> apt-get install usb-modeswitch-data usb-modeswitch
>>
>>
>>  ??
>> Then, you have restarted your PC ?
>>
>>
>> On Mon, Aug 26, 2013 at 6:47 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> Hi
>>> I reinstalled all the prerequisites for gsmopen but still the same
>>> problem again my dongle with sim is not recognised by ubuntu nor by
>>> freeswitch.
>>> Also when i am typing wvdialconf the terminal shows that no modem
>>> detected.
>>> On Aug 26, 2013 9:19 AM, "Nandy Dagondon" <nandy1925 at gmail.com> wrote:
>>>
>>>> It should because the switchmode switches the dongle from appearing as
>>>> a CD-ROM drive to a modem.
>>>>
>>>>
>>>> On Sun, Aug 25, 2013 at 6:40 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>
>>>>> Thank you so much..after this will my dongle be easily detected by
>>>>> FS..???
>>>>> On Aug 25, 2013 4:09 PM, "Giovanni Maruzzelli" <gmaruzz at celliax.org>
>>>>> wrote:
>>>>>
>>>>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Sun, Aug 25, 2013 at 12:26 PM, Ashish Mishra <
>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>
>>>>>>> Thank you,
>>>>>>> Which prerequisites are you talking about...??? The prerequisites
>>>>>>> for mod_gsmopen? I have followed the wiki page that entails the
>>>>>>> prerequisites and perhaps to some extent i feel that i have installed all
>>>>>>> the prerequisites. But if you guys i will give it another shot.
>>>>>>>
>>>>>>> Regards
>>>>>>> Ashish Mishra
>>>>>>> On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> you have probably not installed all the prerequisites as per the
>>>>>>>> wiki page.
>>>>>>>> Your linux does not see the dongle.
>>>>>>>> Please follow the wiki page step by step
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <
>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>>>>>>>>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com>
>>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>>> Hi jayanth,
>>>>>>>>>> This is the new thread that i am starting. I am using the dongle
>>>>>>>>>> E1732 huawei which works well as far as receiving and dialing the calls is
>>>>>>>>>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Sincerely,
>>>>>>>>
>>>>>>>> Giovanni Maruzzelli
>>>>>>>> Cell : +39-347-2665618
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Sincerely,
>>>>>>
>>>>>> Giovanni Maruzzelli
>>>>>> Cell : +39-347-2665618
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Mon Aug 26 21:29:08 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Mon, 26 Aug 2013 22:59:08 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAHDL-sw83E0Zr+4j6SrTBdwavzgyxpG3fXSn9NsRcJ9WM1tMwg@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
	<CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
	<CALXCt0rb7A8wXHFwNmMOLEYiEwer234NNsTaNGfjw5hneQUdFQ@mail.gmail.com>
	<CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>
	<CAHKqMb7p3m_cS0AeA2z_EL_o20eA5b3HrugPM=-KNN7FGeVibw@mail.gmail.com>
	<CAMvhjJtYBsgCQzmWW5qXj80jWNd5Vu4p3EOi4FUO0u_3d6xBcA@mail.gmail.com>
	<CAHDL-sznAVmo49jyJx3J1NMcR8w=Ca5vfBTBeD4U3g--e75UOA@mail.gmail.com>
	<CAMvhjJu10VtF=X9285ST4SicA=-=iXGMvGxFcUOsaqgzQr6vaQ@mail.gmail.com>
	<CAHDL-sw83E0Zr+4j6SrTBdwavzgyxpG3fXSn9NsRcJ9WM1tMwg@mail.gmail.com>
Message-ID: <CAHDL-szCQDw36QrC8kNgHEUm4Fdu25E1OckQKcT1_KbduPvwWQ@mail.gmail.com>

BTW, didn't you mention that you tried the dongle on Windows as well ? Is
that a different Windows machine, or the same one ?
What you could do, is to install Virtualbox with the PUEL extensions for
USB 2.0 pass-thru. Then install Linux in Virtualbox, and try the whole
process. The USB2.0 pass-thru thing requires some very simple configuration
on host side... and almost nothing on guest OS side. In fact, I have such a
similar setup under installation right now on my PC, since I was finding it
bit hard to work with multiple machines, while on the move.


On Mon, Aug 26, 2013 at 10:54 PM, Jayanth Acharya <jayachar88 at gmail.com>wrote:

> Well in that case... pretty much the end of the line from my end.
> I can only make few radical suggestions, like -
> 1. Try a different Linux distro -- I did my install on Debian 'Wheezy' 7.1
> 32-bit and have had some success (there's still a lot to do).
> 2. Try a different PC
> 3. Try using a powered USB hub... a good quality one, like the ones from
> Belkin or D-Link. However, don't even think of the el-cheapo 7-port kinds.
> Those are usually horrible... USB1.1 (over-spec'd to say USB2.0, while most
> aren't) and with terrible power supplies (400mA stated as 2000mA).
> 4. If none of those work, try a different dongle.
>
> Like I said... radical! Others might have better ideas. However, not
> getting the 3 /dev/ttyUSB interfaces, in spite of usb-modeswitch, beats
> me... totally. I have tried half a dozen dongles on my Linux PC's (GSM,
> CDMA), and never did I once fail to get them working, once usb-modeswitch
> kicked in. Of course, I was using those for pure data-connections.
>
>
> On Mon, Aug 26, 2013 at 9:50 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> yes i did restart my pc but still my HUAWEI E1732 dongle not getting
>> recognised by freeSwitch.. :(
>>
>>
>> On Mon, Aug 26, 2013 at 8:23 PM, Jayanth Acharya <jayachar88 at gmail.com>wrote:
>>
>>> Including this --
>>>
>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>
>>>
>>>  ??
>>> Then, you have restarted your PC ?
>>>
>>>
>>> On Mon, Aug 26, 2013 at 6:47 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> Hi
>>>> I reinstalled all the prerequisites for gsmopen but still the same
>>>> problem again my dongle with sim is not recognised by ubuntu nor by
>>>> freeswitch.
>>>> Also when i am typing wvdialconf the terminal shows that no modem
>>>> detected.
>>>> On Aug 26, 2013 9:19 AM, "Nandy Dagondon" <nandy1925 at gmail.com> wrote:
>>>>
>>>>> It should because the switchmode switches the dongle from appearing as
>>>>> a CD-ROM drive to a modem.
>>>>>
>>>>>
>>>>> On Sun, Aug 25, 2013 at 6:40 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>>
>>>>>> Thank you so much..after this will my dongle be easily detected by
>>>>>> FS..???
>>>>>> On Aug 25, 2013 4:09 PM, "Giovanni Maruzzelli" <gmaruzz at celliax.org>
>>>>>> wrote:
>>>>>>
>>>>>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Sun, Aug 25, 2013 at 12:26 PM, Ashish Mishra <
>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>
>>>>>>>> Thank you,
>>>>>>>> Which prerequisites are you talking about...??? The prerequisites
>>>>>>>> for mod_gsmopen? I have followed the wiki page that entails the
>>>>>>>> prerequisites and perhaps to some extent i feel that i have installed all
>>>>>>>> the prerequisites. But if you guys i will give it another shot.
>>>>>>>>
>>>>>>>> Regards
>>>>>>>> Ashish Mishra
>>>>>>>> On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> you have probably not installed all the prerequisites as per the
>>>>>>>>> wiki page.
>>>>>>>>> Your linux does not see the dongle.
>>>>>>>>> Please follow the wiki page step by step
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <
>>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>>>>>>>>>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>>> Hi jayanth,
>>>>>>>>>>> This is the new thread that i am starting. I am using the dongle
>>>>>>>>>>> E1732 huawei which works well as far as receiving and dialing the calls is
>>>>>>>>>>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> Sincerely,
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Sincerely,
>>>>>>>
>>>>>>> Giovanni Maruzzelli
>>>>>>> Cell : +39-347-2665618
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 itsme.kunnu at gmail.com  Mon Aug 26 21:54:43 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Mon, 26 Aug 2013 23:24:43 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CALXCt0pO8HCZE0QyhdP_CmQ4qkg2d4ziuvB+5n_mzNyk2ydxcw@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
	<CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
	<CALXCt0rb7A8wXHFwNmMOLEYiEwer234NNsTaNGfjw5hneQUdFQ@mail.gmail.com>
	<CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>
	<CAHKqMb7p3m_cS0AeA2z_EL_o20eA5b3HrugPM=-KNN7FGeVibw@mail.gmail.com>
	<CAMvhjJtYBsgCQzmWW5qXj80jWNd5Vu4p3EOi4FUO0u_3d6xBcA@mail.gmail.com>
	<CAHDL-sznAVmo49jyJx3J1NMcR8w=Ca5vfBTBeD4U3g--e75UOA@mail.gmail.com>
	<CAMvhjJu10VtF=X9285ST4SicA=-=iXGMvGxFcUOsaqgzQr6vaQ@mail.gmail.com>
	<CALXCt0pO8HCZE0QyhdP_CmQ4qkg2d4ziuvB+5n_mzNyk2ydxcw@mail.gmail.com>
Message-ID: <CAMvhjJu9VXb5A_j2OR_nds9OHwRj_GJTzt669txc6FhLVD34Gg@mail.gmail.com>

Okay i will run a parallel thread on another linux machine...but for the
time being can you tell me if my modem gets recognised...how will i start
making and receiving calls thru that dongle...???
On Aug 26, 2013 10:24 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:

> you probably have some hardware problem, or some Linux install problem.
>
> Try reinstalling Linux on the machine (eg: reformat the hard disk,
> reinstall Ubuntu server  64 bit from cdrom), then reinstall freeswitch and
> gsmopen.
> If this does not work, abandon.
>
> -giovanni
>
>
>
> On Mon, Aug 26, 2013 at 6:20 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> yes i did restart my pc but still my HUAWEI E1732 dongle not getting
>> recognised by freeSwitch.. :(
>>
>>
>> On Mon, Aug 26, 2013 at 8:23 PM, Jayanth Acharya <jayachar88 at gmail.com>wrote:
>>
>>> Including this --
>>>
>>>
>>>
>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>
>>>
>>>  ??
>>> Then, you have restarted your PC ?
>>>
>>>
>>> On Mon, Aug 26, 2013 at 6:47 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> Hi
>>>> I reinstalled all the prerequisites for gsmopen but still the same
>>>> problem again my dongle with sim is not recognised by ubuntu nor by
>>>> freeswitch.
>>>> Also when i am typing wvdialconf the terminal shows that no modem
>>>> detected.
>>>> On Aug 26, 2013 9:19 AM, "Nandy Dagondon" <nandy1925 at gmail.com> wrote:
>>>>
>>>>> It should because the switchmode switches the dongle from appearing as
>>>>> a CD-ROM drive to a modem.
>>>>>
>>>>>
>>>>> On Sun, Aug 25, 2013 at 6:40 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>>
>>>>>> Thank you so much..after this will my dongle be easily detected by
>>>>>> FS..???
>>>>>> On Aug 25, 2013 4:09 PM, "Giovanni Maruzzelli" <gmaruzz at celliax.org>
>>>>>> wrote:
>>>>>>
>>>>>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> On Sun, Aug 25, 2013 at 12:26 PM, Ashish Mishra <
>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>
>>>>>>>> Thank you,
>>>>>>>> Which prerequisites are you talking about...??? The prerequisites
>>>>>>>> for mod_gsmopen? I have followed the wiki page that entails the
>>>>>>>> prerequisites and perhaps to some extent i feel that i have installed all
>>>>>>>> the prerequisites. But if you guys i will give it another shot.
>>>>>>>>
>>>>>>>> Regards
>>>>>>>> Ashish Mishra
>>>>>>>> On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> you have probably not installed all the prerequisites as per the
>>>>>>>>> wiki page.
>>>>>>>>> Your linux does not see the dongle.
>>>>>>>>> Please follow the wiki page step by step
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <
>>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>>>>>>>>>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>>> Hi jayanth,
>>>>>>>>>>> This is the new thread that i am starting. I am using the dongle
>>>>>>>>>>> E1732 huawei which works well as far as receiving and dialing the calls is
>>>>>>>>>>> concerned on windows. I am currently using ubuntu 12.04 LTS for freeswitch.
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> Sincerely,
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> --
>>>>>>> Sincerely,
>>>>>>>
>>>>>>> Giovanni Maruzzelli
>>>>>>> Cell : +39-347-2665618
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mario_fs at mgtech.com  Mon Aug 26 22:39:04 2013
From: mario_fs at mgtech.com (Mario G)
Date: Mon, 26 Aug 2013 11:39:04 -0700
Subject: [Freeswitch-users] FreeSwitch and Yealink
In-Reply-To: <20130812212027.B410A86E9C2@mail.mydcs.ca>
References: <20130807044439.9A5F8F4002@mail.mydcs.ca>
	<CAAQCs31MW_pFNiD=p=udTp4YmDueu=9rY8uDhozrYTU5JGy-uw@mail.gmail.com>
	<20130810010116.3D80657E003@mail.mydcs.ca>
	<9A5E874A-D248-4FCF-84DE-C3AE553F416A@jerris.com>
	<20130812212027.B410A86E9C2@mail.mydcs.ca>
Message-ID: <CC2AB373-DAF3-41B1-A582-9A32FE54DE66@mgtech.com>

I have had zero FreeSwitch problems with the T32 (18 months now) and new T46 but I don't use OpenVPN. Try Yealink tech support, they have been great and fast at fixing issues (rings, dimmer, etc.) for me, usually less than a week. BTW, the phone have great logging and pcap built in, I would test and send that to Yealink.
Mario G

On Aug 12, 2013, at 2:20 PM, Paul <pasha at prosperity4ever.com> wrote:

> This is happening on both T38Gs and T32Gs, I have physical access to the T32, firmware 32.70.0.132
> 
> Paul
> 
> On Mon, 12 Aug, 2013 at 8:21 AM, Michael Jerris <mike at jerris.com> wrote:
>> What phone model and yealink firmware are you using?
>> 
>> On Aug 9, 2013, at 9:01 PM, Paul <pasha at prosperity4ever.com> wrote:
>> 
>>> I'm gonna try all of your guys' suggestions. 
>>> 
>>> Ivan, I do not think its codec related, I made /log 11 and siptrace on all profiles (internal/external) as well as pcap capture of the phone itself. For some strange reason the moment the PBX send a packe tot ye Yealink phone that the remote party picked up and the call should be bridges, the Yealink phone replies with a BYE packet, I'm thinking it might be a bug in the Yealink firmware, so needless to say I have gotten a hold of their support and opened a ticket, see if something comes of that, but meanwhile going to try some of the suggestions in this thread and see if I can get any close to work around.
>>> 
>>> Paul
>>> 
>>> On Fri, 9 Aug, 2013 at 10:59 AM, Nikolay Rogoshchenkov <nickolayr at gmail.com> wrote:
>>>> Check RTP Packet Size too.
>>>> 
>>>> --
>>>> Rogoshchenkov Nikolay
>>>> 
>>>> 
>>>> On Wed, Aug 7, 2013 at 12:44 AM, Paul <pasha at prosperity4ever.com> wrote:
>>>> Hi guys,
>>>> 
>>>> Has anyone had any issues using FreeSwitch with Yealink phones? My phones connect to FS via openvpn tunnel. All incoming calls work no problem, call comes through phones ring everyone can hear each other no issues, having a very strange issue though on the outgoing calls. As soon as the destination party picks up (this external calls) the call hangs up.
>>>> 
>>>> short FS LOG:
>>>> 
>>>> switch_ivr_bridge.c:475 Channel [sofia/internal/105 at 10.0.0.34] has been answered
>>>> sofia.c:6528 Hangup sofia/internal/105 at 10.0.0.34 [CS_EXECUTE] [ORIGINATOR_CANCEL]
>>>> switch_ivr_bridge.c:721 Hangup sofia/external/2503004900 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 
>>>> 
>>>> So FS thinks the phone sent a BYE packet (which I can see with siptrace) but the phone's timer keeps going as if it thinks the call is supposed to keep going.
>>>> 
>>>> Internal extension to extension works fine (even if the extensions are at a different physical location and subnet).
>>>> 
>>>> I setup a second account to one of my asterisk servers and outgoing/incoming work just fine, so it seems this strange combination of FS and Yealink ... does it on 2 models T32G and T38G (only phones I have).
>>>> 
>>>> I have updated firwmare to their latest version (which in the comments say freeswitch ready)
>>>> 
>>>> Wondering if anyone else had any experience with these, or has some thoughts?
>>>> 
>>>> Thanks
>>>> 
>>>> Paul
>>>> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Mon Aug 26 22:39:06 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Tue, 27 Aug 2013 00:09:06 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAMvhjJu9VXb5A_j2OR_nds9OHwRj_GJTzt669txc6FhLVD34Gg@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
	<CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
	<CALXCt0rb7A8wXHFwNmMOLEYiEwer234NNsTaNGfjw5hneQUdFQ@mail.gmail.com>
	<CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>
	<CAHKqMb7p3m_cS0AeA2z_EL_o20eA5b3HrugPM=-KNN7FGeVibw@mail.gmail.com>
	<CAMvhjJtYBsgCQzmWW5qXj80jWNd5Vu4p3EOi4FUO0u_3d6xBcA@mail.gmail.com>
	<CAHDL-sznAVmo49jyJx3J1NMcR8w=Ca5vfBTBeD4U3g--e75UOA@mail.gmail.com>
	<CAMvhjJu10VtF=X9285ST4SicA=-=iXGMvGxFcUOsaqgzQr6vaQ@mail.gmail.com>
	<CALXCt0pO8HCZE0QyhdP_CmQ4qkg2d4ziuvB+5n_mzNyk2ydxcw@mail.gmail.com>
	<CAMvhjJu9VXb5A_j2OR_nds9OHwRj_GJTzt669txc6FhLVD34Gg@mail.gmail.com>
Message-ID: <CAMvhjJuy=kkCRUn8Qu3VWdMaLqxn8aXy9kDmJ8aEHdo2FQbqJw@mail.gmail.com>

Thanks giovanni, jaynath, nandy for your co-operation...i finally got my
modem working actually the problem was in the sim card. Also now when i run
wvdialconf it shows found a modem on /dev/ttyUSB2
After this i ran /dev/ttyUSB* and this showed
Bash: /dev/ttyUSB0: permission denied
Kindly help me in interpretting these two outputs

Regards
Ashish Mishra
On Aug 26, 2013 11:24 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:

> Okay i will run a parallel thread on another linux machine...but for the
> time being can you tell me if my modem gets recognised...how will i start
> making and receiving calls thru that dongle...???
> On Aug 26, 2013 10:24 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:
>
>> you probably have some hardware problem, or some Linux install problem.
>>
>> Try reinstalling Linux on the machine (eg: reformat the hard disk,
>> reinstall Ubuntu server  64 bit from cdrom), then reinstall freeswitch and
>> gsmopen.
>> If this does not work, abandon.
>>
>> -giovanni
>>
>>
>>
>> On Mon, Aug 26, 2013 at 6:20 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>
>>> yes i did restart my pc but still my HUAWEI E1732 dongle not getting
>>> recognised by freeSwitch.. :(
>>>
>>>
>>> On Mon, Aug 26, 2013 at 8:23 PM, Jayanth Acharya <jayachar88 at gmail.com>wrote:
>>>
>>>> Including this --
>>>>
>>>>
>>>>
>>>>
>>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>>
>>>>
>>>>  ??
>>>> Then, you have restarted your PC ?
>>>>
>>>>
>>>> On Mon, Aug 26, 2013 at 6:47 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>
>>>>> Hi
>>>>> I reinstalled all the prerequisites for gsmopen but still the same
>>>>> problem again my dongle with sim is not recognised by ubuntu nor by
>>>>> freeswitch.
>>>>> Also when i am typing wvdialconf the terminal shows that no modem
>>>>> detected.
>>>>> On Aug 26, 2013 9:19 AM, "Nandy Dagondon" <nandy1925 at gmail.com> wrote:
>>>>>
>>>>>> It should because the switchmode switches the dongle from appearing
>>>>>> as a CD-ROM drive to a modem.
>>>>>>
>>>>>>
>>>>>> On Sun, Aug 25, 2013 at 6:40 PM, Ashish Mishra <itsme.kunnu at gmail.com
>>>>>> > wrote:
>>>>>>
>>>>>>> Thank you so much..after this will my dongle be easily detected by
>>>>>>> FS..???
>>>>>>> On Aug 25, 2013 4:09 PM, "Giovanni Maruzzelli" <gmaruzz at celliax.org>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> On Sun, Aug 25, 2013 at 12:26 PM, Ashish Mishra <
>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Thank you,
>>>>>>>>> Which prerequisites are you talking about...??? The prerequisites
>>>>>>>>> for mod_gsmopen? I have followed the wiki page that entails the
>>>>>>>>> prerequisites and perhaps to some extent i feel that i have installed all
>>>>>>>>> the prerequisites. But if you guys i will give it another shot.
>>>>>>>>>
>>>>>>>>> Regards
>>>>>>>>> Ashish Mishra
>>>>>>>>> On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
>>>>>>>>> wrote:
>>>>>>>>>
>>>>>>>>>> you have probably not installed all the prerequisites as per the
>>>>>>>>>> wiki page.
>>>>>>>>>> Your linux does not see the dongle.
>>>>>>>>>> Please follow the wiki page step by step
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <
>>>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>>>>>>>>>>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com>
>>>>>>>>>>> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Hi jayanth,
>>>>>>>>>>>> This is the new thread that i am starting. I am using the
>>>>>>>>>>>> dongle E1732 huawei which works well as far as receiving and dialing the
>>>>>>>>>>>> calls is concerned on windows. I am currently using ubuntu 12.04 LTS for
>>>>>>>>>>>> freeswitch.
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>>
>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> Sincerely,
>>>>>>>>>>
>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> --
>>>>>>>> Sincerely,
>>>>>>>>
>>>>>>>> Giovanni Maruzzelli
>>>>>>>> Cell : +39-347-2665618
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 gmail.com  Mon Aug 26 23:13:51 2013
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Mon, 26 Aug 2013 21:13:51 +0200
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAMvhjJuy=kkCRUn8Qu3VWdMaLqxn8aXy9kDmJ8aEHdo2FQbqJw@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
	<CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
	<CALXCt0rb7A8wXHFwNmMOLEYiEwer234NNsTaNGfjw5hneQUdFQ@mail.gmail.com>
	<CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>
	<CAHKqMb7p3m_cS0AeA2z_EL_o20eA5b3HrugPM=-KNN7FGeVibw@mail.gmail.com>
	<CAMvhjJtYBsgCQzmWW5qXj80jWNd5Vu4p3EOi4FUO0u_3d6xBcA@mail.gmail.com>
	<CAHDL-sznAVmo49jyJx3J1NMcR8w=Ca5vfBTBeD4U3g--e75UOA@mail.gmail.com>
	<CAMvhjJu10VtF=X9285ST4SicA=-=iXGMvGxFcUOsaqgzQr6vaQ@mail.gmail.com>
	<CALXCt0pO8HCZE0QyhdP_CmQ4qkg2d4ziuvB+5n_mzNyk2ydxcw@mail.gmail.com>
	<CAMvhjJu9VXb5A_j2OR_nds9OHwRj_GJTzt669txc6FhLVD34Gg@mail.gmail.com>
	<CAMvhjJuy=kkCRUn8Qu3VWdMaLqxn8aXy9kDmJ8aEHdo2FQbqJw@mail.gmail.com>
Message-ID: <CALXCt0pvZG-8h0ZgwWnbWrufX9AnkeXmtJa44QuC-rL8mQLBOQ@mail.gmail.com>

It's better if you find a friend with a good Linux knowledge, or you'll
waste a lot of time, and end up with much frustration.


On Mon, Aug 26, 2013 at 8:39 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:

> Thanks giovanni, jaynath, nandy for your co-operation...i finally got my
> modem working actually the problem was in the sim card. Also now when i run
> wvdialconf it shows found a modem on /dev/ttyUSB2
> After this i ran /dev/ttyUSB* and this showed
> Bash: /dev/ttyUSB0: permission denied
> Kindly help me in interpretting these two outputs
>
> Regards
> Ashish Mishra
> On Aug 26, 2013 11:24 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>
>> Okay i will run a parallel thread on another linux machine...but for the
>> time being can you tell me if my modem gets recognised...how will i start
>> making and receiving calls thru that dongle...???
>> On Aug 26, 2013 10:24 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
>> wrote:
>>
>>> you probably have some hardware problem, or some Linux install problem.
>>>
>>> Try reinstalling Linux on the machine (eg: reformat the hard disk,
>>> reinstall Ubuntu server  64 bit from cdrom), then reinstall freeswitch and
>>> gsmopen.
>>> If this does not work, abandon.
>>>
>>> -giovanni
>>>
>>>
>>>
>>> On Mon, Aug 26, 2013 at 6:20 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>
>>>> yes i did restart my pc but still my HUAWEI E1732 dongle not getting
>>>> recognised by freeSwitch.. :(
>>>>
>>>>
>>>> On Mon, Aug 26, 2013 at 8:23 PM, Jayanth Acharya <jayachar88 at gmail.com>wrote:
>>>>
>>>>> Including this --
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>>>
>>>>>
>>>>>  ??
>>>>> Then, you have restarted your PC ?
>>>>>
>>>>>
>>>>> On Mon, Aug 26, 2013 at 6:47 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>>
>>>>>> Hi
>>>>>> I reinstalled all the prerequisites for gsmopen but still the same
>>>>>> problem again my dongle with sim is not recognised by ubuntu nor by
>>>>>> freeswitch.
>>>>>> Also when i am typing wvdialconf the terminal shows that no modem
>>>>>> detected.
>>>>>> On Aug 26, 2013 9:19 AM, "Nandy Dagondon" <nandy1925 at gmail.com>
>>>>>> wrote:
>>>>>>
>>>>>>> It should because the switchmode switches the dongle from appearing
>>>>>>> as a CD-ROM drive to a modem.
>>>>>>>
>>>>>>>
>>>>>>> On Sun, Aug 25, 2013 at 6:40 PM, Ashish Mishra <
>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>
>>>>>>>> Thank you so much..after this will my dongle be easily detected by
>>>>>>>> FS..???
>>>>>>>> On Aug 25, 2013 4:09 PM, "Giovanni Maruzzelli" <gmaruzz at celliax.org>
>>>>>>>> wrote:
>>>>>>>>
>>>>>>>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> On Sun, Aug 25, 2013 at 12:26 PM, Ashish Mishra <
>>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>>
>>>>>>>>>> Thank you,
>>>>>>>>>> Which prerequisites are you talking about...??? The prerequisites
>>>>>>>>>> for mod_gsmopen? I have followed the wiki page that entails the
>>>>>>>>>> prerequisites and perhaps to some extent i feel that i have installed all
>>>>>>>>>> the prerequisites. But if you guys i will give it another shot.
>>>>>>>>>>
>>>>>>>>>> Regards
>>>>>>>>>> Ashish Mishra
>>>>>>>>>> On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
>>>>>>>>>> wrote:
>>>>>>>>>>
>>>>>>>>>>> you have probably not installed all the prerequisites as per the
>>>>>>>>>>> wiki page.
>>>>>>>>>>> Your linux does not see the dongle.
>>>>>>>>>>> Please follow the wiki page step by step
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <
>>>>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>>>>>>>>>>>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <itsme.kunnu at gmail.com>
>>>>>>>>>>>> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Hi jayanth,
>>>>>>>>>>>>> This is the new thread that i am starting. I am using the
>>>>>>>>>>>>> dongle E1732 huawei which works well as far as receiving and dialing the
>>>>>>>>>>>>> calls is concerned on windows. I am currently using ubuntu 12.04 LTS for
>>>>>>>>>>>>> freeswitch.
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>>
>>>>>>>>>>>> 
>>>>>>>>>>>> 
>>>>>>>>>>>>
>>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> --
>>>>>>>>>>> Sincerely,
>>>>>>>>>>>
>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>>
>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> --
>>>>>>>>> Sincerely,
>>>>>>>>>
>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>>
>>> --
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> Cell : +39-347-2665618
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From itsme.kunnu at gmail.com  Mon Aug 26 23:24:39 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Tue, 27 Aug 2013 00:54:39 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CALXCt0pvZG-8h0ZgwWnbWrufX9AnkeXmtJa44QuC-rL8mQLBOQ@mail.gmail.com>
References: <CAMvhjJu+GcCSj7026vXVJYLMi5suDHKBiEBFgGM+8252VK_HGw@mail.gmail.com>
	<CAMvhjJvWomUtnT2QooUH0nfo8ou9o72cBWVnbFi5Udk02y96eQ@mail.gmail.com>
	<CAMvhjJvYaxDRsOLcGktUa7SRmS5uqdfEb+S_OXpy6Xn4Qgyw0w@mail.gmail.com>
	<CALXCt0oqS+RvFpqEANUSv5fZOcwAPFb3Uyz1d+fM_RMteHf-Pw@mail.gmail.com>
	<CAMvhjJt4AZBv0Gu61iMVqvqVV_83xSX4d+dBENcCQARezrOVFA@mail.gmail.com>
	<CALXCt0rb7A8wXHFwNmMOLEYiEwer234NNsTaNGfjw5hneQUdFQ@mail.gmail.com>
	<CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>
	<CAHKqMb7p3m_cS0AeA2z_EL_o20eA5b3HrugPM=-KNN7FGeVibw@mail.gmail.com>
	<CAMvhjJtYBsgCQzmWW5qXj80jWNd5Vu4p3EOi4FUO0u_3d6xBcA@mail.gmail.com>
	<CAHDL-sznAVmo49jyJx3J1NMcR8w=Ca5vfBTBeD4U3g--e75UOA@mail.gmail.com>
	<CAMvhjJu10VtF=X9285ST4SicA=-=iXGMvGxFcUOsaqgzQr6vaQ@mail.gmail.com>
	<CALXCt0pO8HCZE0QyhdP_CmQ4qkg2d4ziuvB+5n_mzNyk2ydxcw@mail.gmail.com>
	<CAMvhjJu9VXb5A_j2OR_nds9OHwRj_GJTzt669txc6FhLVD34Gg@mail.gmail.com>
	<CAMvhjJuy=kkCRUn8Qu3VWdMaLqxn8aXy9kDmJ8aEHdo2FQbqJw@mail.gmail.com>
	<CALXCt0pvZG-8h0ZgwWnbWrufX9AnkeXmtJa44QuC-rL8mQLBOQ@mail.gmail.com>
Message-ID: <CAMvhjJtsUGKPPi1D7VyB9qy5506MxP1V16vJMFF=q+hcK-8kNw@mail.gmail.com>

Thanks for your advice...but i would rather request you to pls answer
actual question also.

Regards
Ashish Mishra
On Aug 27, 2013 12:49 AM, "Giovanni Maruzzelli" <gmaruzz at gmail.com> wrote:

> It's better if you find a friend with a good Linux knowledge, or you'll
> waste a lot of time, and end up with much frustration.
>
>
> On Mon, Aug 26, 2013 at 8:39 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>
>> Thanks giovanni, jaynath, nandy for your co-operation...i finally got my
>> modem working actually the problem was in the sim card. Also now when i run
>> wvdialconf it shows found a modem on /dev/ttyUSB2
>> After this i ran /dev/ttyUSB* and this showed
>> Bash: /dev/ttyUSB0: permission denied
>> Kindly help me in interpretting these two outputs
>>
>> Regards
>> Ashish Mishra
>> On Aug 26, 2013 11:24 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
>>
>>> Okay i will run a parallel thread on another linux machine...but for the
>>> time being can you tell me if my modem gets recognised...how will i start
>>> making and receiving calls thru that dongle...???
>>> On Aug 26, 2013 10:24 PM, "Giovanni Maruzzelli" <gmaruzz at gmail.com>
>>> wrote:
>>>
>>>> you probably have some hardware problem, or some Linux install problem.
>>>>
>>>> Try reinstalling Linux on the machine (eg: reformat the hard disk,
>>>> reinstall Ubuntu server  64 bit from cdrom), then reinstall freeswitch and
>>>> gsmopen.
>>>> If this does not work, abandon.
>>>>
>>>> -giovanni
>>>>
>>>>
>>>>
>>>> On Mon, Aug 26, 2013 at 6:20 PM, Ashish Mishra <itsme.kunnu at gmail.com>wrote:
>>>>
>>>>> yes i did restart my pc but still my HUAWEI E1732 dongle not getting
>>>>> recognised by freeSwitch.. :(
>>>>>
>>>>>
>>>>> On Mon, Aug 26, 2013 at 8:23 PM, Jayanth Acharya <jayachar88 at gmail.com
>>>>> > wrote:
>>>>>
>>>>>> Including this --
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>>>>
>>>>>>
>>>>>>  ??
>>>>>> Then, you have restarted your PC ?
>>>>>>
>>>>>>
>>>>>> On Mon, Aug 26, 2013 at 6:47 PM, Ashish Mishra <itsme.kunnu at gmail.com
>>>>>> > wrote:
>>>>>>
>>>>>>> Hi
>>>>>>> I reinstalled all the prerequisites for gsmopen but still the same
>>>>>>> problem again my dongle with sim is not recognised by ubuntu nor by
>>>>>>> freeswitch.
>>>>>>> Also when i am typing wvdialconf the terminal shows that no modem
>>>>>>> detected.
>>>>>>> On Aug 26, 2013 9:19 AM, "Nandy Dagondon" <nandy1925 at gmail.com>
>>>>>>> wrote:
>>>>>>>
>>>>>>>> It should because the switchmode switches the dongle from appearing
>>>>>>>> as a CD-ROM drive to a modem.
>>>>>>>>
>>>>>>>>
>>>>>>>> On Sun, Aug 25, 2013 at 6:40 PM, Ashish Mishra <
>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>
>>>>>>>>> Thank you so much..after this will my dongle be easily detected by
>>>>>>>>> FS..???
>>>>>>>>> On Aug 25, 2013 4:09 PM, "Giovanni Maruzzelli" <
>>>>>>>>> gmaruzz at celliax.org> wrote:
>>>>>>>>>
>>>>>>>>>> apt-get install usb-modeswitch-data usb-modeswitch
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> On Sun, Aug 25, 2013 at 12:26 PM, Ashish Mishra <
>>>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>>>
>>>>>>>>>>> Thank you,
>>>>>>>>>>> Which prerequisites are you talking about...??? The
>>>>>>>>>>> prerequisites for mod_gsmopen? I have followed the wiki page that entails
>>>>>>>>>>> the prerequisites and perhaps to some extent i feel that i have installed
>>>>>>>>>>> all the prerequisites. But if you guys i will give it another shot.
>>>>>>>>>>>
>>>>>>>>>>> Regards
>>>>>>>>>>> Ashish Mishra
>>>>>>>>>>> On Aug 25, 2013 2:30 PM, "Giovanni Maruzzelli" <
>>>>>>>>>>> gmaruzz at gmail.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>> you have probably not installed all the prerequisites as per
>>>>>>>>>>>> the wiki page.
>>>>>>>>>>>> Your linux does not see the dongle.
>>>>>>>>>>>> Please follow the wiki page step by step
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> On Sun, Aug 25, 2013 at 10:09 AM, Ashish Mishra <
>>>>>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>> Also ls -l /dev/ttyUSB* gives me no such file or directory.
>>>>>>>>>>>>> On Aug 25, 2013 1:30 PM, "Ashish Mishra" <
>>>>>>>>>>>>> itsme.kunnu at gmail.com> wrote:
>>>>>>>>>>>>>
>>>>>>>>>>>>>> Hi jayanth,
>>>>>>>>>>>>>> This is the new thread that i am starting. I am using the
>>>>>>>>>>>>>> dongle E1732 huawei which works well as far as receiving and dialing the
>>>>>>>>>>>>>> calls is concerned on windows. I am currently using ubuntu 12.04 LTS for
>>>>>>>>>>>>>> freeswitch.
>>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>>>
>>>>>>>>>>>>> 
>>>>>>>>>>>>> 
>>>>>>>>>>>>>
>>>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>>>
>>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> --
>>>>>>>>>>>> Sincerely,
>>>>>>>>>>>>
>>>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>>
>>>>>>>>>>>> 
>>>>>>>>>>>> 
>>>>>>>>>>>>
>>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>>
>>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>>
>>>>>>>>>>> 
>>>>>>>>>>> 
>>>>>>>>>>>
>>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>>
>>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> --
>>>>>>>>>> Sincerely,
>>>>>>>>>>
>>>>>>>>>> Giovanni Maruzzelli
>>>>>>>>>> Cell : +39-347-2665618
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>> _________________________________________________________________________
>>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>>> consulting at freeswitch.org
>>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>>
>>>>>>>>>> 
>>>>>>>>>> 
>>>>>>>>>>
>>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>>> http://www.freeswitch.org
>>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>>> http://www.cluecon.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
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>> _________________________________________________________________________
>>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>>> consulting at freeswitch.org
>>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>>
>>>>>>>>> 
>>>>>>>>> 
>>>>>>>>>
>>>>>>>>> Official FreeSWITCH Sites
>>>>>>>>> http://www.freeswitch.org
>>>>>>>>> http://wiki.freeswitch.org
>>>>>>>>> http://www.cluecon.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
>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>> _________________________________________________________________________
>>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>>> consulting at freeswitch.org
>>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>>
>>>>>>>> 
>>>>>>>> 
>>>>>>>>
>>>>>>>> Official FreeSWITCH Sites
>>>>>>>> http://www.freeswitch.org
>>>>>>>> http://wiki.freeswitch.org
>>>>>>>> http://www.cluecon.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
>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>> _________________________________________________________________________
>>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>>> consulting at freeswitch.org
>>>>>>> http://www.freeswitchsolutions.com
>>>>>>>
>>>>>>> 
>>>>>>> 
>>>>>>>
>>>>>>> Official FreeSWITCH Sites
>>>>>>> http://www.freeswitch.org
>>>>>>> http://wiki.freeswitch.org
>>>>>>> http://www.cluecon.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
>>>>>>>
>>>>>>>
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> Cell : +39-347-2665618
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 red.rain.seven at gmail.com  Tue Aug 27 01:33:06 2013
From: red.rain.seven at gmail.com (Henry Huang)
Date: Mon, 26 Aug 2013 14:33:06 -0700
Subject: [Freeswitch-users] Pacemaker Resource Agent
In-Reply-To: <CAC-LwPN9EyFH6MLWqgH+W+kUwvYE1WquFmEur69R-7c5L8NXTw@mail.gmail.com>
References: <CAMbW9oJ3F_K3nMz0qKj9KFfYq_ZLB-buLkYAOu9s6xNooq6f1Q@mail.gmail.com>
	<00000140b257ea79-cfb47be2-f8c6-4466-86a0-c6bde0cc4ec9-000000@email.amazonses.com>
	<CAMbW9o+hQYHDNVCtTbB-3RSDBSDU8FA-9+SAf3Vs6qbvYA-EMw@mail.gmail.com>
	<CAMbW9oL7HYmRgxNBJz-DkDn5xF5X1cejz9LD5pYmusMYrrQKMA@mail.gmail.com>
	<CAMbW9oLarp-soSycAtDXpo_eJuV0PMdRhgZ6JjSNcrDzCEtUDA@mail.gmail.com>
	<CAC-LwPN9EyFH6MLWqgH+W+kUwvYE1WquFmEur69R-7c5L8NXTw@mail.gmail.com>
Message-ID: <CAP_G395zBo9tYJY+jVpv-vXP+gd_zi8hVt1z2oUyVMfChYpr+A@mail.gmail.com>

thank you for sharing


On Mon, Aug 26, 2013 at 5:12 AM, Vik Killa <vipkilla at gmail.com> wrote:

> I wrote both those wiki pages:
> http://wiki.freeswitch.org/wiki/Enterprise_deployment_IP_Failover
> http://wiki.freeswitch.org/wiki/Enterprise_deployment_with_Corosync
>
> I only tested them on Debian Squeeze -- they both worked on Squeeze.
> Please feel free to update the wiki for Debian Wheezy or any other distro.
>
> <snip>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at davri.com  Tue Aug 27 01:58:47 2013
From: bdfoster at davri.com (Brian Foster)
Date: Mon, 26 Aug 2013 17:58:47 -0400
Subject: [Freeswitch-users] Tuning DTMF
In-Reply-To: <FB7D97A214987F458242ACBDF87614073B1BD19E26@clubvirtual40.ClubMessage.local>
References: <FB7D97A214987F458242ACBDF87614073B1BD19CD6@clubvirtual40.ClubMessage.local>
	<CAHgk4BWrESZ=cnZonBHTPQr0r3aUzOYZHJpPGOmzsA3xMoc-Rg@mail.gmail.com>
	<FB7D97A214987F458242ACBDF87614073B1BD19E26@clubvirtual40.ClubMessage.local>
Message-ID: <CAHgk4BWt6coPoRLSPQVQ9q6YM48hGa_gNCkxv7L5mPJbugQAyQ@mail.gmail.com>

Yep that's perfectly normal. As long as FS is interpreting the DTMF
correctly you are good to go. As a quick test if you are still unsure call
into FS and transfer to the demo ivr and see if it does what you expect.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 26, 2013 2:32 AM, "Grant Bagdasarian" <GB at cm.nl> wrote:

> Yes, that works fine. The cracking DTMF tones only happen when the remote
> party is sending DTMF tones. So if I call my cellphone with X-Lite through
> FS and my cellphones starts sending DTMF tones, that?s when the tones come
> in with some sort of cracking sound.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian Foster
> *Sent:* Saturday, August 24, 2013 3:13 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Tuning DTMF****
>
> ** **
>
> Have you tried using Xlite on an IVR or something? Call your bank or your
> cable company and see if you can get through the menus fine. I'd bet that
> you can. Post back and let us know. They sound short but that should be ok.
> ****
>
> Thank you,****
>
> Brian Foster
> Project Manager/Owner's Rep.
> Davri Investments, Inc.
> O: 317-787-2686 x2102
> M: 317-600-9753
> E: bdfoster at davri.com
> Indianapolis, Indiana****
>
> Sent from a mobile device.****
>
> On Aug 23, 2013 3:21 AM, "Grant Bagdasarian" <GB at cm.nl> wrote:****
>
> Hello,****
>
>  ****
>
> Setup is as following:****
>
>  ****
>
> X-Lite (Test phone) ------>  SIP Proxy ------> FS (as SBC) ------> Carrier
> ------> End-user phone****
>
>  ****
>
> Whenever I place a call using X-Lite to the End-user phone, in this case
> my own mobile phone or office phone, and start sending DTMF tones from the
> remote telephone, the DMTF tones arrive with a crackling sound at the end
> of the tone.****
>
>  ****
>
> Console debug:****
>
> 2013-08-23 09:00:13.813207 [DEBUG] switch_rtp.c:3829 Send start packet for
> [2] ts=3778476449 dur=160/160/1280 seq=5098 lw=-516490847****
>
> 2013-08-23 09:00:13.833208 [DEBUG] switch_rtp.c:3729 Send middle packet
> for [2] ts=3778476449 dur=320/320/1280 seq=5099 lw=161****
>
> 2013-08-23 09:00:13.853208 [DEBUG] switch_rtp.c:3729 Send middle packet
> for [2] ts=3778476449 dur=480/480/1280 seq=5100 lw=161****
>
> 2013-08-23 09:00:13.873208 [DEBUG] switch_rtp.c:3729 Send middle packet
> for [2] ts=3778476449 dur=640/640/1280 seq=5101 lw=161****
>
> 2013-08-23 09:00:13.893208 [DEBUG] switch_rtp.c:3729 Send middle packet
> for [2] ts=3778476449 dur=800/800/1280 seq=5102 lw=161****
>
> 2013-08-23 09:00:13.913208 [DEBUG] switch_rtp.c:3729 Send middle packet
> for [2] ts=3778476449 dur=960/960/1280 seq=5103 lw=161****
>
> 2013-08-23 09:00:13.933208 [DEBUG] switch_rtp.c:3729 Send middle packet
> for [2] ts=3778476449 dur=1120/1120/1280 seq=5104 lw=161****
>
> 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for
> [2] ts=3778476449 dur=1280/1280/1280 seq=5105 lw=1****
>
> 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for
> [2] ts=3778476449 dur=1280/1280/1280 seq=5106 lw=1****
>
> 2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for
> [2] ts=3778476449 dur=1280/1280/1280 seq=5107 lw=1****
>
> 2013-08-23 09:00:14.813206 [DEBUG] switch_rtp.c:5117 RTP RECV DTMF 1:1280*
> ***
>
> 2013-08-23 09:00:14.813206 [DEBUG] switch_channel.c:471 RECV DTMF 1:1280**
> **
>
>  ****
>
> Both internal and external sip profiles have the following configuration
> for DTMF:****
>
> <param name="liberal-dtmf" value="true"/>****
>
> <param name="rfc2833-pt" value="101"/>****
>
> <param name="dtmf-duration" value="2000"/>****
>
> <param name="inbound-codec-negotiation" value="generous"/>****
>
> <param name="inbound-late-negotiation" value="true"/>****
>
>  ****
>
> Internal.xml (Dialplan)****
>
>     <extension name="internal_to_external">****
>
>       <condition field="${sip_req_host}" expression="^10\.0\.0\.1$">****
>
>         <action application="set" data="hangup_after_bridge=false"/>****
>
>         <action application="set" data="continue_on_fail=true"/>****
>
>         <action application="bridge" data="sofia/external/${sip_req_uri}"/>
> ****
>
>       </condition>****
>
>     </extension>****
>
>  ****
>
> Is there a way to tune the DTMF tones? If so, which parameters do I need
> to add or edit?****
>
>  ****
>
> Thanks!****
>
>  ****
>
> Grant****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Tue Aug 27 03:47:38 2013
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 26 Aug 2013 18:47:38 -0500
Subject: [Freeswitch-users] Thanks to all those that make ClueCon Possible!!
Message-ID: <CE41534A.79AD9%krice@freeswitch.org>

ClueCon 2013 was a smashing success and you had everything to do with it! We
were very glad to have you with us this year and we hope you enjoyed it
every bit as much as we did.

The ClueCon team is currently working on getting all of the videos and
presentation slides put up on the ClueCon website. More information will be
sent out soon. In the meantime be sure to join us each Wednesday at 1PM
Eastern time for the ClueCon Weekly conference call.

What about ClueCon 2014? It's already in the works! 2014 marks the 10th
anniversary of ClueCon - a milestone you'll not want to miss. We will be
putting out a call for speakers as well as getting our sponsors arranged in
the next few months.? Stay tuned for more information.

If you missed Michael Giagnocavo's ClueCon 2013 presentation, "Easy
Real-Time CDR Reporting with VoltDB,"?check it out here
<http://goo.gl/EywbDm> . And then?grab a free trial of VoltDB
<http://go.voltdb.com/voltdb-enterprise-software-download> ?? the database
that helps Michael manage high-volume CDRs.

The ClueCon team would like to thank the following sponsors:

Nenad, Moises, and all the good people at Sangoma <http://www.sangoma.com> .
Sangoma has been with us since the very first ClueCon.

Sean, Jeff, and the folks at Flowroute <http://www.flowroute.com> .

Arnie and the gang at NACT <http://www.nact.com> .

Kashif and the team over at Vestec <http://www.vestec.com> .

Norm and the VoiceNetwork <http://www.voicenetwork.ca>  team.

Venky and the good folks at Plivo <http://www.plivo.com> .

The whole OrecX <http://www.orecx.com>  team.

John, Nicole and the rest of the gang at OnSIP <http://www.onsip.com> .

Darren and all the techies at 2600hz <http://www.2600hz.com> .

John, Dave, and all the rest of the Truphone <http://www.truphone.com>
guys.

Kevin and the team at Ooma <http://www.ooma.com> .

Tommy and the folks at Snom <http://www.snom.com> .

The tokbox <http://www.tokbox.com>  team.

The NG Communications <http://www.ngcommunications.com>  team.

We hope to see you all again in 2014!


-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From fs.user at fordior.net  Tue Aug 27 04:13:21 2013
From: fs.user at fordior.net (EL)
Date: Tue, 27 Aug 2013 02:13:21 +0200
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <CAMvhjJtsUGKPPi1D7VyB9qy5506MxP1V16vJMFF=q+hcK-8kNw@mail.gmail.com>
References: <CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>
	<CAHKqMb7p3m_cS0AeA2z_EL_o20eA5b3HrugPM=-KNN7FGeVibw@mail.gmail.com>
	<CAMvhjJtYBsgCQzmWW5qXj80jWNd5Vu4p3EOi4FUO0u_3d6xBcA@mail.gmail.com>
	<CAHDL-sznAVmo49jyJx3J1NMcR8w=Ca5vfBTBeD4U3g--e75UOA@mail.gmail.com>
	<CAMvhjJu10VtF=X9285ST4SicA=-=iXGMvGxFcUOsaqgzQr6vaQ@mail.gmail.com>
	<CALXCt0pO8HCZE0QyhdP_CmQ4qkg2d4ziuvB+5n_mzNyk2ydxcw@mail.gmail.com>
	<CAMvhjJu9VXb5A_j2OR_nds9OHwRj_GJTzt669txc6FhLVD34Gg@mail.gmail.com>
	<CAMvhjJuy=kkCRUn8Qu3VWdMaLqxn8aXy9kDmJ8aEHdo2FQbqJw@mail.gmail.com>
	<CALXCt0pvZG-8h0ZgwWnbWrufX9AnkeXmtJa44QuC-rL8mQLBOQ@mail.gmail.com>
	<CAMvhjJtsUGKPPi1D7VyB9qy5506MxP1V16vJMFF=q+hcK-8kNw@mail.gmail.com>
Message-ID: <20130827001321.GB25035@0rdior.com>

Dear Ashish,

> >> Bash: /dev/ttyUSB0: permission denied

What have you already tried to solve this issue? 
--> http://bit.ly/15reJqV 

Although for several reasons I prefer to use ddg.gg myself.

A quick search for: 
"Bash: /dev/ttyUSB0: permission denied" +solved
gave lots of potential answers. 

Since a few weeks, I've seen several emails from you passing by. I would
like to provide you some usefull URLS (which are worth reading for everyone
who's working in IT, although the second URL is more focusing on
developers).

I plan te reread the following information at least once a year to 'check
and balance' my own approach: 8+)

[1] http://goo.gl/zi5V
[2] http://goo.gl/wzsul

Apply it, and become a master!

-- 
EL


From gmangudai at gmail.com  Tue Aug 27 04:55:30 2013
From: gmangudai at gmail.com (Vincent Xia)
Date: Tue, 27 Aug 2013 08:55:30 +0800
Subject: [Freeswitch-users] is it possible to have FS forcilby unregiter
 a registered UA
In-Reply-To: <20130826143734.GB22574@0rdior.com>
References: <CAKQo6tnkgwmneo3QiG0cnBZF5LsApapsYZ4o778fT9SH+5SUGg@mail.gmail.com>
	<CAFiqYukMVqxHhs6rwL7QKZBn0Y-bZ+s3bWqAfUHOdR=33iWqUQ@mail.gmail.com>
	<CAKQo6t=BQHZPigtGCDg=O_eMW=F2GFDxsaPZOem=Uf6sZ98RqQ@mail.gmail.com>
	<20130826143734.GB22574@0rdior.com>
Message-ID: <CAKQo6tk1divhwMb7tN7u0gOnaO=jGY22BXwh0gyuFtZ8ci4xMA@mail.gmail.com>

it now works 1006 at xxx.xxx.xxx.xxx, but i remember i've tried this option
before, maybe i had typo out there,
anyway many thanks!


2013/8/26 EL <fs.user at fordior.net>

> > yes from the same user, but i have 2 profiles, 1 for LAN and 1 for
> > INTERNET, and i don't want 1 user to register both in LAN and INTERNET,
> so
> > im considering to kill the previous registration once i got register
> > request from the same user but a different profile.
>
> I guess the 'internet' side would be for your mobile phone (while both
> are the same user)? I think it's better to implement a vpn. Or do you
> really want to allow the internetside (public profile) to make calls
> through your box?
>
> Anyway, there is a security warning for this; see section 'External' here:
> https://wiki.freeswitch.org/wiki/Getting_Started_Guide#What_SIP_Profiles_Do
>
> --
> EL
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 wesleyakio at tuntscorp.com  Tue Aug 27 05:13:15 2013
From: wesleyakio at tuntscorp.com (Wesley Akio)
Date: Mon, 26 Aug 2013 22:13:15 -0300
Subject: [Freeswitch-users] fifo recording, loopback and other evil stuff
Message-ID: <CAGTsDSseQTRBe2=_ObQUr8JaEdoQC+ccCMFdQceQYhTeY2gNxA@mail.gmail.com>

Hi all,

As far as I can tell there is no way to use fifo_record_template with
variables for on hook agents. I has been asked and asked again, with no
easy answer.

Keeping that in mind I went down a not so pleasant path:

<member timeout="10" simo="1"
lag="5">{fifo_member_wait=nowait,ignore_early_media=true,loopback_bowout=false,loopback/a1000/default</member>

As you can see there is a loopback right there. The recording is done in
the dialplan and things work as they should, to a point.

Since the softphone of choice has more than one line and nobody seems to be
able to disable it we use the limit app to avoid call waiting, hence the
loopback_bowout=false, otherwise the limit is freed on bowout.

Point being: Does someone know a way to record the outgoing call to a fifo
on hook agent in a cleaner way?

Best,

Wesley Akio
TuntsCorp.com
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From nneul at mst.edu  Tue Aug 27 05:34:44 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Mon, 26 Aug 2013 20:34:44 -0500
Subject: [Freeswitch-users] Configuration of freeswitch with Acme Packet
 Session Directors and BroadWorks
Message-ID: <521C0234.80709@mst.edu>

Running into an issue with getting a configuration up and running with a provider.

Setup is a broadworks backend, fronted by Acme packet directors at the provider. The trunk is set up with a pilot 
userid, and configured with the various TNs on the trunk.

Inbound calls to each of the DIDs are working without issue.

Outbound calls will only work at this point if they configure IP authentication as a workaround.

Have tried with and without the extension-in-contact/extension parameters, which had no effect on the outbound calls, 
but broke inbound due to it mapping everything to the single extension.



Net symptom is that on an outbound, it does an invite, gets a 100 trying, and then a 403 forbidden. It never gets a 407 
proxy auth required in response to the invite. The register request/interaction works fine though.


Does anyone have this type of setup deployed that can suggest any config options/etc? I don't have any details of the 
configuration on the provider side, but if there is something specific, I can ask.


-- Nathan

------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From byron at theclarkfamily.name  Mon Aug 26 21:36:15 2013
From: byron at theclarkfamily.name (Byron Clark)
Date: Mon, 26 Aug 2013 11:36:15 -0600
Subject: [Freeswitch-users] Recording Duration on EC2
Message-ID: <CAC5vAOQFMj4Oykxr-Adymc8HrzDjxKtcEx5bvoYJV24CE4uzwQ@mail.gmail.com>

I'm using FreeSWITCH on EC2 to call into a conference bridge and record the
audio from the conference. It works well except for one thing: the
recording contains all the audio from the call, but the duration of the
file is shorter than the call. The duration is typically 3-5 seconds
shorter than the call for each 5 minutes of call. On a recent test, wall
clock showed a duration of 15:32.93 but the duration of the recording file
was 15:19.96. The ugly part is that this only happens on EC2 instances
where there is some CPU steal time (<5%) occurring.

Here's my setup:
Operating System: Ubuntu 12.04
FreeSWITCH: 1.2.12

Command I'm using to start the call:
originate {record_waste_resources=true}sofia/external/SIPADDR
&record(/tmp/record1.wav) default default

Any ideas on how to make the recording duration actually match up with the
call duration? Or even what's really going wrong so I can work on fixing
that?

--
Byron Clark
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From terry at digital-outpost.com  Tue Aug 27 08:33:29 2013
From: terry at digital-outpost.com (Terry Barnum)
Date: Mon, 26 Aug 2013 21:33:29 -0700
Subject: [Freeswitch-users] Google voice call fails
Message-ID: <9665D20D-7235-4058-8666-14A654692503@digital-outpost.com>

I've been reading the freeswitch book (great read!) and playing with my first freeswitch install on a Mac mini host with freeswitch installed via git into a Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones and a Yealink T32G, internally between extensions, using a free DID to test incoming calls and a free but limited iptel account to test outbound.

My next step was to try and play with real outbound calls so I followed these instructions for Google voice: <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just PCMU but no change. xmpp appears to be connecting and authenticating to google correctly.

The log of an attempted call (edited for privacy) is at <http://pastebin.freeswitch.org/21338>

Snippets of the log that looks suspicious to a freeswitch noob:

2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates.
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates.
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates
2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888
2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates
2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto
2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305
2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates
2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates
2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates
2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888
2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543 at voice.google.com

Where should I be looking to try and fix? Freeswitch is behind the firewall with a private IP but STUN says it's working (or at least it looks that way to me). 

Thanks,
-Terry




From jleung at v10networks.ca  Tue Aug 27 08:44:42 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Mon, 26 Aug 2013 21:44:42 -0700
Subject: [Freeswitch-users] Google voice call fails
In-Reply-To: <9665D20D-7235-4058-8666-14A654692503@digital-outpost.com>
References: <9665D20D-7235-4058-8666-14A654692503@digital-outpost.com>
Message-ID: <006501cea2e0$26b95fe0$742c1fa0$@v10networks.ca>

Check to see if your IP addressing settings are set correctly within the
dingaling profile.

Most of the times you'll need to have the profile to listen internally but
have dingaling to specify an external IP address when it talks to the
outside world.

> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] On Behalf Of Terry Barnum
> Sent: Monday, August 26, 2013 9:33 PM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Google voice call fails
> 
> I've been reading the freeswitch book (great read!) and playing with my
first
> freeswitch install on a Mac mini host with freeswitch installed via git
into a
> Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones
and
> a Yealink T32G, internally between extensions, using a free DID to test
> incoming calls and a free but limited iptel account to test outbound.
> 
> My next step was to try and play with real outbound calls so I followed
these
> instructions for Google voice:
> <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and
> are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just
> PCMU but no change. xmpp appears to be connecting and authenticating to
> google correctly.
> 
> The log of an attempted call (edited for privacy) is at
> <http://pastebin.freeswitch.org/21338>
> 
> Snippets of the log that looks suspicious to a freeswitch noob:
> 
> 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0
> rtp candidates.
> 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0
> rtcp candidates.
> 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0
> video_rtp candidates
> 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0
> video_rctp candidates
> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing
> session for 2155444888
> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates
> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate
> 74.125.141.127:19305 PASS ACL wan.auto
> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp
> Candidate 74.125.141.127:19305
> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp
> candidates
> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates
> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp
> candidates
> 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing
> session for 2155444888
> 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup
> dingaling/gtalk/+7609876543 at voice.google.com
> 
> Where should I be looking to try and fix? Freeswitch is behind the
firewall
> with a private IP but STUN says it's working (or at least it looks that
way to
> me).
> 
> Thanks,
> -Terry
> 
> 
> 
> __________________________________________________________
> _______________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 sertys at gmail.com  Tue Aug 27 10:05:52 2013
From: sertys at gmail.com (Daniel Ivanov)
Date: Tue, 27 Aug 2013 08:05:52 +0200
Subject: [Freeswitch-users] Thanks to all those that make ClueCon
	Possible!!
In-Reply-To: <CE41534A.79AD9%krice@freeswitch.org>
References: <CE41534A.79AD9%krice@freeswitch.org>
Message-ID: <CAMeZp60uG702f8NfR-+pv=Nu03Ua-K-4dFe8t7nS9mEXdzbHiQ@mail.gmail.com>

Im overly happy with the success of cluecon. Am always superglad to hear a
real tech event marking a smashing success. Amazing job from the organizers
and speakers. Can't wait to view the media from it.
On Aug 27, 2013 2:53 AM, "Ken Rice" <krice at freeswitch.org> wrote:

>  ClueCon 2013 was a smashing success and you had everything to do with
> it! We were very glad to have you with us this year and we hope you enjoyed
> it every bit as much as we did.
>
> The ClueCon team is currently working on getting all of the videos and
> presentation slides put up on the ClueCon website. More information will be
> sent out soon. In the meantime be sure to join us each Wednesday at 1PM
> Eastern time for the ClueCon Weekly conference call.
>
> What about ClueCon 2014? It's already in the works! 2014 marks the 10th
> anniversary of ClueCon - a milestone you'll not want to miss. We will be
> putting out a call for speakers as well as getting our sponsors arranged in
> the next few months.  Stay tuned for more information.
>
> If you missed Michael Giagnocavo's ClueCon 2013 presentation, "Easy
> Real-Time CDR Reporting with VoltDB," check it out here <*
> http://goo.gl/EywbDm*> . And then grab a free trial of VoltDB <*
> http://go.voltdb.com/voltdb-enterprise-software-download*>  ? the
> database that helps Michael manage high-volume CDRs.
>
> The ClueCon team would like to thank the following sponsors:
>
> Nenad, Moises, and all the good people at Sangoma <*http://www.sangoma.com
> *> . Sangoma has been with us since the very first ClueCon.
>
> Sean, Jeff, and the folks at Flowroute <*http://www.flowroute.com*> .
>
> Arnie and the gang at NACT <*http://www.nact.com*> .
>
> Kashif and the team over at Vestec <*http://www.vestec.com*> .
>
> Norm and the VoiceNetwork <*http://www.voicenetwork.ca*>  team.
>
> Venky and the good folks at Plivo <*http://www.plivo.com*> .
>
> The whole OrecX <*http://www.orecx.com*>  team.
>
> John, Nicole and the rest of the gang at OnSIP <*http://www.onsip.com*> .
>
> Darren and all the techies at 2600hz <*http://www.2600hz.com*> .
>
> John, Dave, and all the rest of the Truphone <*http://www.truphone.com*>
>  guys.
>
> Kevin and the team at Ooma <*http://www.ooma.com*> .
>
> Tommy and the folks at Snom <*http://www.snom.com*> .
>
> The tokbox <*http://www.tokbox.com*>  team.
>
> The NG Communications <*http://www.ngcommunications.com*>  team.
>
> We hope to see you all again in 2014!
>
>
> --
> Ken
> *http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
> *irc.freenode.net #freeswitch
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 at cassidywebservices.co.uk  Tue Aug 27 12:15:50 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Tue, 27 Aug 2013 09:15:50 +0100
Subject: [Freeswitch-users] Pacemaker Resource Agent
In-Reply-To: <CAP_G395zBo9tYJY+jVpv-vXP+gd_zi8hVt1z2oUyVMfChYpr+A@mail.gmail.com>
References: <CAMbW9oJ3F_K3nMz0qKj9KFfYq_ZLB-buLkYAOu9s6xNooq6f1Q@mail.gmail.com>
	<00000140b257ea79-cfb47be2-f8c6-4466-86a0-c6bde0cc4ec9-000000@email.amazonses.com>
	<CAMbW9o+hQYHDNVCtTbB-3RSDBSDU8FA-9+SAf3Vs6qbvYA-EMw@mail.gmail.com>
	<CAMbW9oL7HYmRgxNBJz-DkDn5xF5X1cejz9LD5pYmusMYrrQKMA@mail.gmail.com>
	<CAMbW9oLarp-soSycAtDXpo_eJuV0PMdRhgZ6JjSNcrDzCEtUDA@mail.gmail.com>
	<CAC-LwPN9EyFH6MLWqgH+W+kUwvYE1WquFmEur69R-7c5L8NXTw@mail.gmail.com>
	<CAP_G395zBo9tYJY+jVpv-vXP+gd_zi8hVt1z2oUyVMfChYpr+A@mail.gmail.com>
Message-ID: <CAMbW9o+ok7GLiwbhpERY5GBM6K_bJ+9g22D32p7-acpDAJp87g@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Enterprise_deployment_Debian_Wheezy_Corosync_Pacemaker

My (possibly terrible) wiki page with the configuration that I got working.


On 26 August 2013 22:33, Henry Huang <red.rain.seven at gmail.com> wrote:

> thank you for sharing
>
>
> On Mon, Aug 26, 2013 at 5:12 AM, Vik Killa <vipkilla at gmail.com> wrote:
>
>> I wrote both those wiki pages:
>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_IP_Failover
>> http://wiki.freeswitch.org/wiki/Enterprise_deployment_with_Corosync
>>
>> I only tested them on Debian Squeeze -- they both worked on Squeeze.
>> Please feel free to update the wiki for Debian Wheezy or any other distro.
>>
>> <snip>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From GB at cm.nl  Tue Aug 27 13:16:25 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Tue, 27 Aug 2013 11:16:25 +0200
Subject: [Freeswitch-users] Tuning DTMF
In-Reply-To: <CAHgk4BWt6coPoRLSPQVQ9q6YM48hGa_gNCkxv7L5mPJbugQAyQ@mail.gmail.com>
References: <FB7D97A214987F458242ACBDF87614073B1BD19CD6@clubvirtual40.ClubMessage.local>
	<CAHgk4BWrESZ=cnZonBHTPQr0r3aUzOYZHJpPGOmzsA3xMoc-Rg@mail.gmail.com>
	<FB7D97A214987F458242ACBDF87614073B1BD19E26@clubvirtual40.ClubMessage.local>
	<CAHgk4BWt6coPoRLSPQVQ9q6YM48hGa_gNCkxv7L5mPJbugQAyQ@mail.gmail.com>
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD1A023@clubvirtual40.ClubMessage.local>

Everything works perfectly!

Thanks

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian Foster
Sent: Monday, August 26, 2013 11:59 PM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Tuning DTMF


Yep that's perfectly normal. As long as FS is interpreting the DTMF correctly you are good to go. As a quick test if you are still unsure call into FS and transfer to the demo ivr and see if it does what you expect.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com<mailto:bdfoster at davri.com>
Indianapolis, Indiana

Sent from a mobile device.
On Aug 26, 2013 2:32 AM, "Grant Bagdasarian" <GB at cm.nl<mailto:GB at cm.nl>> wrote:
Yes, that works fine. The cracking DTMF tones only happen when the remote party is sending DTMF tones. So if I call my cellphone with X-Lite through FS and my cellphones starts sending DTMF tones, that's when the tones come in with some sort of cracking sound.

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 Brian Foster
Sent: Saturday, August 24, 2013 3:13 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Tuning DTMF


Have you tried using Xlite on an IVR or something? Call your bank or your cable company and see if you can get through the menus fine. I'd bet that you can. Post back and let us know. They sound short but that should be ok.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102<tel:317-787-2686%20x2102>
M: 317-600-9753<tel:317-600-9753>
E: bdfoster at davri.com<mailto:bdfoster at davri.com>
Indianapolis, Indiana

Sent from a mobile device.
On Aug 23, 2013 3:21 AM, "Grant Bagdasarian" <GB at cm.nl<mailto:GB at cm.nl>> wrote:
Hello,

Setup is as following:

X-Lite (Test phone) ------>  SIP Proxy ------> FS (as SBC) ------> Carrier ------> End-user phone

Whenever I place a call using X-Lite to the End-user phone, in this case my own mobile phone or office phone, and start sending DTMF tones from the remote telephone, the DMTF tones arrive with a crackling sound at the end of the tone.

Console debug:
2013-08-23 09:00:13.813207 [DEBUG] switch_rtp.c:3829 Send start packet for [2] ts=3778476449 dur=160/160/1280 seq=5098 lw=-516490847
2013-08-23 09:00:13.833208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=320/320/1280<tel:320%2F320%2F1280> seq=5099 lw=161
2013-08-23 09:00:13.853208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=480/480/1280<tel:480%2F480%2F1280> seq=5100 lw=161
2013-08-23 09:00:13.873208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=640/640/1280 seq=5101 lw=161
2013-08-23 09:00:13.893208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=800/800/1280<tel:800%2F800%2F1280> seq=5102 lw=161
2013-08-23 09:00:13.913208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=960/960/1280 seq=5103 lw=161
2013-08-23 09:00:13.933208 [DEBUG] switch_rtp.c:3729 Send middle packet for [2] ts=3778476449 dur=1120/1120/1280 seq=5104 lw=161
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5105 lw=1
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5106 lw=1
2013-08-23 09:00:13.953207 [DEBUG] switch_rtp.c:3729 Send end packet for [2] ts=3778476449 dur=1280/1280/1280 seq=5107 lw=1
2013-08-23 09:00:14.813206 [DEBUG] switch_rtp.c:5117 RTP RECV DTMF 1:1280
2013-08-23 09:00:14.813206 [DEBUG] switch_channel.c:471 RECV DTMF 1:1280

Both internal and external sip profiles have the following configuration for DTMF:
<param name="liberal-dtmf" value="true"/>
<param name="rfc2833-pt" value="101"/>
<param name="dtmf-duration" value="2000"/>
<param name="inbound-codec-negotiation" value="generous"/>
<param name="inbound-late-negotiation" value="true"/>

Internal.xml (Dialplan)
    <extension name="internal_to_external">
      <condition field="${sip_req_host}" expression="^10\.0\.0\.1$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="bridge" data="sofia/external/${sip_req_uri}"/>
      </condition>
    </extension>

Is there a way to tune the DTMF tones? If so, which parameters do I need to add or edit?

Thanks!

Grant


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org<mailto:consulting at freeswitch.org>
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

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_________________________________________________________________________
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http://www.freeswitchsolutions.com




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http://wiki.freeswitch.org
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From shahrzad.aziminia at gmail.com  Tue Aug 27 15:40:40 2013
From: shahrzad.aziminia at gmail.com (Shahrzad A.)
Date: Tue, 27 Aug 2013 13:40:40 +0200
Subject: [Freeswitch-users] Issue with JSSIP + Freeswitch
Message-ID: <CAA7Zpyp=j2dg4+6NhnTtEgRYUSsQy7qBMs1EhcZM4kkRXguKCQ@mail.gmail.com>

Hi Michael

Thanks for your reply, Would you please advice me how to change the ACL to
fix this issue.

Here is my log when I try to call from a JSSIP/Sipml5 to another
JSSIP/Sipml5 client:
2013-08-27 13:33:37.742425 [NOTICE] switch_channel.c:1030 New Channel
sofia/internal/1008 at 10.0.14.16 [771e9dfb-ef68-4adc-9e5a-380e172e1945]
2013-08-27 13:33:37.762422 [INFO] mod_dialplan_xml.c:558 Processing Lala
<1008>->1004 in context public
2013-08-27 13:33:37.762422 [NOTICE] switch_ivr.c:1831 Transfer
sofia/internal/1008 at 10.0.14.16 to XML[1004 at default]
2013-08-27 13:33:37.762422 [INFO] mod_dialplan_xml.c:558 Processing Lala
<1008>->1004 in context default
2013-08-27 13:33:37.762422 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *1
execute_extension::dx XML features
2013-08-27 13:33:37.762422 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *2
record_session::/home/saz/freeService/recordings/1008.2013-08-27-13-33-37.wav
2013-08-27 13:33:37.762422 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *3
execute_extension::cf XML features
2013-08-27 13:33:37.762422 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *4
execute_extension::att_xfer XML features
2013-08-27 13:33:37.762422 [NOTICE] switch_channel.c:1030 New Channel
sofia/internal/sip:1004 at df7jal23ls0d.invalid[876787a6-a8f5-4d1c-bd98-ab2246a9114f]
2013-08-27 13:33:37.802411 [NOTICE] sofia.c:5898 Ring-Ready
sofia/internal/sip:1004 at df7jal23ls0d.invalid!
2013-08-27 13:33:37.802411 [WARNING] switch_channel.c:3250 rtp_secure_media
invalid in this context.
2013-08-27 13:33:37.802411 [INFO] switch_ivr_originate.c:1190 Sending early
media
2013-08-27 13:33:37.802411 [WARNING] switch_core_media.c:2074 NO candidate
ACL defined, Defaulting to wan.auto
2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2107 Choose audio
Candidate cid: 1 proto: udp type: host addr: 10.0.14.254:33865
2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2107 Choose audio
Candidate cid: 2 proto: udp type: host addr: 10.0.14.254:33865
2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2112 Save audio
Candidate cid: 1 proto: udp type: host addr: 10.0.14.253:53815
2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2112 Save audio
Candidate cid: 2 proto: udp type: host addr: 10.0.14.253:53815
2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2221 setting remote
audio ice addr to 10.0.14.254:33865 based on candidate
2013-08-27 13:33:37.802411 [NOTICE] switch_core_media.c:2239 setting remote
rtcp audio addr to 10.0.14.254:33865 based on candidate
2013-08-27 13:33:37.822462 [INFO] switch_core_media.c:4227 Activating Audio
ICE
2013-08-27 13:33:37.822462 [NOTICE] switch_rtp.c:3323 Activating RTP audio
ICE: ruUP7F49vDADu+Gp:GMkGzis5aAzjoC0v 10.0.14.254:33865
2013-08-27 13:33:37.822462 [INFO] switch_core_media.c:4270 Activating RTCP
PORT 33865
2013-08-27 13:33:37.822462 [INFO] switch_core_media.c:4278 Skipping RTCP
ICE (Same as RTP)
2013-08-27 13:33:37.822462 [INFO] switch_rtp.c:2711 Activating Audio Secure
RTP SEND
2013-08-27 13:33:37.822462 [INFO] switch_rtp.c:2689 Activating Audio Secure
RTP RECV
2013-08-27 13:33:37.822462 [NOTICE] sofia_media.c:92 Pre-Answer
sofia/internal/1008 at 10.0.14.16!
2013-08-27 13:33:37.922423 [NOTICE] switch_rtp.c:1053 Auto Changing
stun/rtp/dtls port from 10.0.14.254:33865 to 10.0.14.253:53815
2013-08-27 13:33:42.342422 [WARNING] switch_core_media.c:2074 NO candidate
ACL defined, Defaulting to wan.auto
2013-08-27 13:33:42.342422 [NOTICE] switch_core_media.c:2107 Choose audio
Candidate cid: 1 proto: udp type: host addr: 10.0.14.254:45217
2013-08-27 13:33:42.342422 [NOTICE] switch_core_media.c:2112 Save audio
Candidate cid: 1 proto: udp type: host addr: 10.0.14.253:38004
2013-08-27 13:33:42.342422 [NOTICE] switch_core_media.c:2221 setting remote
audio ice addr to 10.0.14.254:45217 based on candidate
2013-08-27 13:33:42.342422 [INFO] switch_core_media.c:4227 Activating Audio
ICE
2013-08-27 13:33:42.342422 [NOTICE] switch_rtp.c:3323 Activating RTP audio
ICE: gd9dRK830iEDLw2W:vECHRV23TjEY0c0I 10.0.14.254:45217
2013-08-27 13:33:42.342422 [INFO] switch_core_media.c:4270 Activating RTCP
PORT 45217
2013-08-27 13:33:42.342422 [INFO] switch_rtp.c:2478 Activate RTP/RTCP audio
DTLS server
2013-08-27 13:33:42.342422 [NOTICE] sofia.c:6561 Channel
[sofia/internal/sip:1004 at df7jal23ls0d.invalid] has been answered
2013-08-27 13:33:42.362407 [NOTICE] switch_ivr_originate.c:3437 Channel
[sofia/internal/1008 at 10.0.14.16] has been answered


As you can see there is no DTLS or handshake! But the interesting part is,
when I try to call from SIPML5 client to a JSSIP client I have the
following log with handshake and the voice is perfect! would you please
tell me what i'm missing!

2013-08-27 13:37:05.242424 [NOTICE] switch_channel.c:1030 New Channel
sofia/internal/1008 at 10.0.14.16 [14f54a7b-6077-4bc3-b845-0b8bbfa3d59d]
2013-08-27 13:37:05.282411 [INFO] mod_dialplan_xml.c:558 Processing Lala
<1008>->1003 in context public
2013-08-27 13:37:05.282411 [NOTICE] switch_ivr.c:1831 Transfer
sofia/internal/1008 at 10.0.14.16 to XML[1003 at default]
2013-08-27 13:37:05.282411 [INFO] mod_dialplan_xml.c:558 Processing Lala
<1008>->1003 in context default
2013-08-27 13:37:05.282411 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *1
execute_extension::dx XML features
2013-08-27 13:37:05.282411 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *2
record_session::/home/saz/freeService/recordings/1008.2013-08-27-13-37-05.wav
2013-08-27 13:37:05.282411 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *3
execute_extension::cf XML features
2013-08-27 13:37:05.282411 [INFO] switch_ivr_async.c:3628 Bound B-Leg: *4
execute_extension::att_xfer XML features
2013-08-27 13:37:05.282411 [NOTICE] switch_channel.c:1030 New Channel
sofia/internal/sip:75icg9r5 at i3ifqjalvn7j.invalid[d1fb6893-6870-4b5d-8d2c-28b038df9965]
2013-08-27 13:37:05.322412 [NOTICE] sofia.c:5898 Ring-Ready
sofia/internal/sip:75icg9r5 at i3ifqjalvn7j.invalid!
2013-08-27 13:37:05.322412 [WARNING] switch_channel.c:3250 rtp_secure_media
invalid in this context.
2013-08-27 13:37:05.322412 [INFO] switch_ivr_originate.c:1190 Sending early
media
2013-08-27 13:37:05.322412 [WARNING] switch_core_media.c:2074 NO candidate
ACL defined, Defaulting to wan.auto
2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2107 Choose audio
Candidate cid: 1 proto: udp type: host addr: 10.0.14.254:49236
2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2107 Choose audio
Candidate cid: 2 proto: udp type: host addr: 10.0.14.254:49236
2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2112 Save audio
Candidate cid: 1 proto: udp type: host addr: 10.0.14.253:40898
2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2112 Save audio
Candidate cid: 2 proto: udp type: host addr: 10.0.14.253:40898
2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2221 setting remote
audio ice addr to 10.0.14.254:49236 based on candidate
2013-08-27 13:37:05.322412 [NOTICE] switch_core_media.c:2239 setting remote
rtcp audio addr to 10.0.14.254:49236 based on candidate
2013-08-27 13:37:05.322412 [INFO] switch_core_media.c:4227 Activating Audio
ICE
2013-08-27 13:37:05.322412 [NOTICE] switch_rtp.c:3323 Activating RTP audio
ICE: 9nidt62aGODady6Y:Hbj7Rd3CDzM2yOIn 10.0.14.254:49236
2013-08-27 13:37:05.322412 [INFO] switch_core_media.c:4270 Activating RTCP
PORT 49236
2013-08-27 13:37:05.322412 [INFO] switch_core_media.c:4278 Skipping RTCP
ICE (Same as RTP)
2013-08-27 13:37:05.322412 [INFO] switch_rtp.c:2711 Activating Audio Secure
RTP SEND
2013-08-27 13:37:05.322412 [INFO] switch_rtp.c:2689 Activating Audio Secure
RTP RECV
2013-08-27 13:37:05.322412 [NOTICE] sofia_media.c:92 Pre-Answer
sofia/internal/1008 at 10.0.14.16!
2013-08-27 13:37:17.302425 [WARNING] switch_core_media.c:2074 NO candidate
ACL defined, Defaulting to wan.auto
2013-08-27 13:37:17.302425 [NOTICE] switch_core_media.c:2107 Choose audio
Candidate cid: 1 proto: udp type: host addr: 10.0.14.254:48285
2013-08-27 13:37:17.302425 [NOTICE] switch_core_media.c:2112 Save audio
Candidate cid: 1 proto: udp type: host addr: 10.0.14.253:53583
2013-08-27 13:37:17.302425 [NOTICE] switch_core_media.c:2221 setting remote
audio ice addr to 10.0.14.254:48285 based on candidate
2013-08-27 13:37:17.302425 [INFO] switch_core_media.c:4227 Activating Audio
ICE
2013-08-27 13:37:17.302425 [NOTICE] switch_rtp.c:3323 Activating RTP audio
ICE: O8IjxMVvzmi10qL4:brG3ubnHrh04oRt9 10.0.14.254:48285
2013-08-27 13:37:17.302425 [INFO] switch_core_media.c:4270 Activating RTCP
PORT 48285
2013-08-27 13:37:17.302425 [INFO] switch_rtp.c:2478 Activate RTP/RTCP audio
DTLS server
2013-08-27 13:37:17.302425 [NOTICE] sofia.c:6561 Channel
[sofia/internal/sip:75icg9r5 at i3ifqjalvn7j.invalid] has been answered
2013-08-27 13:37:17.322410 [NOTICE] switch_ivr_originate.c:3437 Channel
[sofia/internal/1008 at 10.0.14.16] has been answered
2013-08-27 13:37:19.322423 [NOTICE] switch_rtp.c:1053 Auto Changing
stun/rtp/dtls port from 10.0.14.254:48285 to 10.0.14.253:53583
2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2370 Changing audio DTLS
state from HANDSHAKE to SETUP
2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2289 audio Fingerprint
Verified.
2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2711 Activating Audio Secure
RTP SEND
2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2689 Activating Audio Secure
RTP RECV
2013-08-27 13:37:21.362423 [INFO] switch_rtp.c:2329 Changing audio DTLS
state from SETUP to READY
2013-08-27 13:37:31.362412 [INFO] switch_rtp.c:2391 Changing audio DTLS
state from READY to HANDSHAKE
2013-08-27 13:37:31.362412 [INFO] switch_rtp.c:2370 Changing audio DTLS
state from HANDSHAKE to SETUP
2013-08-27 13:37:31.362412 [INFO] switch_rtp.c:2289 audio Fingerprint
Verified.
2013-08-27 13:37:31.362412 [INFO] switch_rtp.c:2329 Changing audio DTLS
state from SETUP to READY
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From itsme.kunnu at gmail.com  Tue Aug 27 16:42:57 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Tue, 27 Aug 2013 18:12:57 +0530
Subject: [Freeswitch-users] Mod_gsmopen help
In-Reply-To: <20130827001321.GB25035@0rdior.com>
References: <CAMvhjJtxQcJoeKyDcEUd-Tk9WKjcg42F5LCj+LsXqmfDcaUW9g@mail.gmail.com>
	<CAHKqMb7p3m_cS0AeA2z_EL_o20eA5b3HrugPM=-KNN7FGeVibw@mail.gmail.com>
	<CAMvhjJtYBsgCQzmWW5qXj80jWNd5Vu4p3EOi4FUO0u_3d6xBcA@mail.gmail.com>
	<CAHDL-sznAVmo49jyJx3J1NMcR8w=Ca5vfBTBeD4U3g--e75UOA@mail.gmail.com>
	<CAMvhjJu10VtF=X9285ST4SicA=-=iXGMvGxFcUOsaqgzQr6vaQ@mail.gmail.com>
	<CALXCt0pO8HCZE0QyhdP_CmQ4qkg2d4ziuvB+5n_mzNyk2ydxcw@mail.gmail.com>
	<CAMvhjJu9VXb5A_j2OR_nds9OHwRj_GJTzt669txc6FhLVD34Gg@mail.gmail.com>
	<CAMvhjJuy=kkCRUn8Qu3VWdMaLqxn8aXy9kDmJ8aEHdo2FQbqJw@mail.gmail.com>
	<CALXCt0pvZG-8h0ZgwWnbWrufX9AnkeXmtJa44QuC-rL8mQLBOQ@mail.gmail.com>
	<CAMvhjJtsUGKPPi1D7VyB9qy5506MxP1V16vJMFF=q+hcK-8kNw@mail.gmail.com>
	<20130827001321.GB25035@0rdior.com>
Message-ID: <CAMvhjJu+-yDiFQmOePNm0q4+5V47Ft=Cz7KhU2vLoSv2d=pefQ@mail.gmail.com>

Hii jaynath,
May i ask you how did you set the the usb control/audio ports correctly in
the gsmopen.conf.xml file. As already mentioned the wvdial command shows me
that modem connected to /dev/ttyUSB2. Did you reload the mod_gsmopen in FS
after making these changes?
Kindly help!!!

Regards
Ashish Mishra
On Aug 27, 2013 5:47 AM, "EL" <fs.user at fordior.net> wrote:

> Dear Ashish,
>
> > >> Bash: /dev/ttyUSB0: permission denied
>
> What have you already tried to solve this issue?
> --> http://bit.ly/15reJqV
>
> Although for several reasons I prefer to use ddg.gg myself.
>
> A quick search for:
> "Bash: /dev/ttyUSB0: permission denied" +solved
> gave lots of potential answers.
>
> Since a few weeks, I've seen several emails from you passing by. I would
> like to provide you some usefull URLS (which are worth reading for everyone
> who's working in IT, although the second URL is more focusing on
> developers).
>
> I plan te reread the following information at least once a year to 'check
> and balance' my own approach: 8+)
>
> [1] http://goo.gl/zi5V
> [2] http://goo.gl/wzsul
>
> Apply it, and become a master!
>
> --
> EL
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 lloydie.t at gmail.com  Tue Aug 27 17:10:48 2013
From: lloydie.t at gmail.com (lloyd thomas)
Date: Tue, 27 Aug 2013 14:10:48 +0100
Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
Message-ID: <CAE=3qOZJkN4_RwvSNVXYzsCTYrqr+jhM2Wz00Yo4SB+kLuA_hg@mail.gmail.com>

I just need a little advice setting up a freeswitch VM box shared with a
http VM on hyper-v host and was wondering whether there are any gotcha's
that I should be aware of.


I expect about 50 devices to be registered on it at any time and not
expecting a high volume of calls.
It will be a fairly low spec dual core machine (dell sc1425), with 8gb ram

Thanks in advance for any advice.
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From Ola.Backstrom at loxysoft.se  Tue Aug 27 13:36:49 2013
From: Ola.Backstrom at loxysoft.se (=?iso-8859-1?Q?Ola__B=E4ckstr=F6m?=)
Date: Tue, 27 Aug 2013 09:36:49 +0000
Subject: [Freeswitch-users] Controlling eavesdrop with event instead of DTMF,
 problem with event headers
Message-ID: <42b7ceb701cb49e19a1e354fb41697b1@exsrv01.ls.local>

I have successfully used eavesdrop to allow a coach to help an agent during his call to a customer.
At first coach hear the agent and the customer, and after pressing 2 on the coach phone can he give advice to the agent. Later he'd press 0 to just listen etc.

Now I'd like disable the dtmf control by setting a session variable
    eavesdrop_enable_dtmf=false
and use the eavesdrop-command event mechanism outlined in this mail thread: http://lists.freeswitch.org/pipermail/freeswitch-users/2009-October/048655.html.
I'm using the mod_erlang_event (could also use mod_event_socket) mechanism sendevent.

The problem is that I don't create proper events that reaches the correct session.
I guess I need to properly name the event and provide the correct headers. Anybody has any ideas?

Currently I'm calling, using Erlang
   freeswitch:sendevent(Node, 'COMMAND', [{"eavesdrop-command", "2"}, {"unique-id", CoachSessionUUID}])
(where CoachSessionUUID is the coach freeswitch session identifier).

Using mod_event_socket, this would correspond to
    sendevent COMMAND
    eavesdrop-command: 2
    unique-id: 533da8e2-0ef0-11e3-9642-d9e368e496c6
(where I've given a particular CoachSessionUUID)

I've also tried to use NOTIFY instead of COMMAND but with no success.

(using Freeswitch 1.2.10, but I don't think the particular version matters)

Regards
/Ola
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From veerabhadrarao.kankatala at panamaxil.com  Tue Aug 27 17:33:50 2013
From: veerabhadrarao.kankatala at panamaxil.com (Veerabhadra Rao)
Date: Tue, 27 Aug 2013 19:03:50 +0530
Subject: [Freeswitch-users] regarding block DTMF
Message-ID: <521CAABE.5040703@panamaxil.com>

hi,

I am working on freeswitch-1.0.6 version. In this i want to block DTMF  
dialled in specific leg after bridging is done.

When i worked on 1.2.4 version i found  
"switch_ivr_block_dtmf_session(session)" which will block DTMF but now i 
am working on 1.0.6 , the above function is not working in this version. 
So is there any replacement for blocking dtmf in 1.0.6 version.

please help me,

thank you


From grcamauer at gmail.com  Tue Aug 27 17:43:54 2013
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Tue, 27 Aug 2013 10:43:54 -0300
Subject: [Freeswitch-users] regarding block DTMF
In-Reply-To: <521CAABE.5040703@panamaxil.com>
References: <521CAABE.5040703@panamaxil.com>
Message-ID: <-3105900189534064427@unknownmsgid>

Do you REALLY need to run 1.06?  That is a very old version. I find
that 1.2.12 is quite stable.  What is keeping you from upgrading?

Guillermo

Sent from my iPhone

On 27/08/2013, at 10:38, Veerabhadra Rao
<veerabhadrarao.kankatala at panamaxil.com> wrote:

> hi,
>
> I am working on freeswitch-1.0.6 version. In this i want to block DTMF
> dialled in specific leg after bridging is done.
>
> When i worked on 1.2.4 version i found
> "switch_ivr_block_dtmf_session(session)" which will block DTMF but now i
> am working on 1.0.6 , the above function is not working in this version.
> So is there any replacement for blocking dtmf in 1.0.6 version.
>
> please help me,
>
> thank you
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 ssinyagin at yahoo.com  Tue Aug 27 19:16:07 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Tue, 27 Aug 2013 08:16:07 -0700 (PDT)
Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
In-Reply-To: <CAE=3qOZJkN4_RwvSNVXYzsCTYrqr+jhM2Wz00Yo4SB+kLuA_hg@mail.gmail.com>
References: <CAE=3qOZJkN4_RwvSNVXYzsCTYrqr+jhM2Wz00Yo4SB+kLuA_hg@mail.gmail.com>
Message-ID: <1377616567.24060.YahooMailNeo@web126202.mail.ne1.yahoo.com>

You will likely have audio distortions. OpenVZ or Xen virtualization should do a better job (myself, I run two production servers under XEN and never had problems with audio quality).

Search in Google for FreeSWITCH or Asterisk under Hyper-V, and you will see the whole spectrum of opinions, and no real detailed test results.






>________________________________
> From: lloyd thomas <lloydie.t at gmail.com>
>To: freeswitch-users <FreeSWITCH-users at lists.freeswitch.org> 
>Sent: Tuesday, August 27, 2013 3:10 PM
>Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
> 
>
>
>I just need a little advice setting up a freeswitch VM box shared with a http VM on hyper-v host and was wondering whether there are any gotcha's that I should be aware of.
>
>
>
>I expect about 50 devices to be registered on it at any time and not expecting a high volume of calls.
>
>It will be a fairly low spec dual core machine (dell sc1425), with 8gb ram
>
>
>
>Thanks in advance for any advice.
>
>
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From jleung at v10networks.ca  Tue Aug 27 19:31:00 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Tue, 27 Aug 2013 08:31:00 -0700
Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
In-Reply-To: <1377616567.24060.YahooMailNeo@web126202.mail.ne1.yahoo.com>
References: <CAE=3qOZJkN4_RwvSNVXYzsCTYrqr+jhM2Wz00Yo4SB+kLuA_hg@mail.gmail.com>
	<1377616567.24060.YahooMailNeo@web126202.mail.ne1.yahoo.com>
Message-ID: <A70AC0C1B7E7FD478A6BF19705461AA0C8D8@IMBER.smb.curriegrad2004.ca>

I would have to say that one can expect audio distortion issues is just
not true anymore.

 

With the new Hyper-V integration code present since the 3.5 and 3.8
kernels which Ubuntu 12.04 and 13.04 uses, the Linux kernel can use a
synthetic clock source that Hyper-V provides just like KVM does. Even
the CentOS 6.4 kernel uses a synthetic clock source too as the guys at
Red Hat back ported a lot of changes from the upstream Hyper-V code back
to their kernel.

 

If the synthetic clock source doesn't work out for you, there is always
the good old mod_timerfd to fix issues like this.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Stanislav Sinyagin
Sent: Tuesday, August 27, 2013 8:16 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v

 

You will likely have audio distortions. OpenVZ or Xen virtualization
should do a better job (myself, I run two production servers under XEN
and never had problems with audio quality).

 

Search in Google for FreeSWITCH or Asterisk under Hyper-V, and you will
see the whole spectrum of opinions, and no real detailed test results.

 

 

 

	 

	
________________________________


	From: lloyd thomas <lloydie.t at gmail.com>
	To: freeswitch-users <FreeSWITCH-users at lists.freeswitch.org> 
	Sent: Tuesday, August 27, 2013 3:10 PM
	Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v

	 

	I just need a little advice setting up a freeswitch VM box
shared with a http VM on hyper-v host and was wondering whether there
are any gotcha's that I should be aware of.
	
	

	I expect about 50 devices to be registered on it at any time and
not expecting a high volume of calls.

	It will be a fairly low spec dual core machine (dell sc1425),
with 8gb ram

	 

	Thanks in advance for any advice.

	 

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From davide at internetone.it  Tue Aug 27 20:50:47 2013
From: davide at internetone.it (Davide Rossi)
Date: Tue, 27 Aug 2013 16:50:47 +0000
Subject: [Freeswitch-users] xml_curl bindings=phrases issue
Message-ID: <9041cca8c2bf482bb0391fd6ffc3f356@mail.internetone.local>

Hi everybody,
I'm a newbie using freeswitch (1.2.7) and having an issue with xml_curl phrases: reading logs I don't see any request to fetch data and using "xml_curl debug_on" no file is generated.
Bindings to "configuration" and "dialplan" worked like a charm... Do I have to configure something special for "phrases" bindings?
I have static IVR menus calling a dynamic phrase macro in http://localhost/getphrases.php.
Is it a misconfiguration or maybe a bug like http://jira.freeswitch.org/browse/FS-5563?
Can't find help on wiki, any hint is really appreciated.

---xml_curl.conf.xml---
<configuration name="xml_curl.conf" description="cURL XML Gateway">
<bindings> 
<binding name="phrases">
	<param name="gateway-url" bindings="phrases" value="http://localhost/getphrases.php"/>
</binding>
</bindings>
</configuration>

---console log [7]---
[DEBUG] switch_ivr_menu.c:469 Executing IVR menu xxx
[DEBUG] switch_ivr_play_say.c:70 No language specified - Using [xx]
[ERR] switch_ivr_play_say.c:145 Can't find macro xxx_ivr_main_menu.
[WARNING] switch_ivr_play_say.c:348 Macro [xxx_ivr_main_menu]: '' did not match any patterns
[DEBUG] switch_ivr_menu.c:363 waiting for 4/4 digits t/o 2000
[DEBUG] switch_ivr_menu.c:410 digits ''



From mike at jerris.com  Tue Aug 27 21:10:21 2013
From: mike at jerris.com (Michael Jerris)
Date: Tue, 27 Aug 2013 13:10:21 -0400
Subject: [Freeswitch-users] xml_curl bindings=phrases issue
In-Reply-To: <9041cca8c2bf482bb0391fd6ffc3f356@mail.internetone.local>
References: <9041cca8c2bf482bb0391fd6ffc3f356@mail.internetone.local>
Message-ID: <4B84AA44-096E-4D3C-AEF6-33F0A140D283@jerris.com>

Phrases area not loaded on demand, they are all loaded in the initial config load.  It requires a reloadxml to re-load them.

On Aug 27, 2013, at 12:50 PM, Davide Rossi <davide at internetone.it> wrote:

> Hi everybody,
> I'm a newbie using freeswitch (1.2.7) and having an issue with xml_curl phrases: reading logs I don't see any request to fetch data and using "xml_curl debug_on" no file is generated.
> Bindings to "configuration" and "dialplan" worked like a charm... Do I have to configure something special for "phrases" bindings?
> I have static IVR menus calling a dynamic phrase macro in http://localhost/getphrases.php.
> Is it a misconfiguration or maybe a bug like http://jira.freeswitch.org/browse/FS-5563?
> Can't find help on wiki, any hint is really appreciated.
> 
> ---xml_curl.conf.xml---
> <configuration name="xml_curl.conf" description="cURL XML Gateway">
> <bindings> 
> <binding name="phrases">
> 	<param name="gateway-url" bindings="phrases" value="http://localhost/getphrases.php"/>
> </binding>
> </bindings>
> </configuration>
> 
> ---console log [7]---
> [DEBUG] switch_ivr_menu.c:469 Executing IVR menu xxx
> [DEBUG] switch_ivr_play_say.c:70 No language specified - Using [xx]
> [ERR] switch_ivr_play_say.c:145 Can't find macro xxx_ivr_main_menu.
> [WARNING] switch_ivr_play_say.c:348 Macro [xxx_ivr_main_menu]: '' did not match any patterns
> [DEBUG] switch_ivr_menu.c:363 waiting for 4/4 digits t/o 2000
> [DEBUG] switch_ivr_menu.c:410 digits ''



From cjbujold at accra.ca  Tue Aug 27 21:18:28 2013
From: cjbujold at accra.ca (Charles Bujold)
Date: Tue, 27 Aug 2013 14:18:28 -0300
Subject: [Freeswitch-users] Newbie question for remote office
Message-ID: <006001cea349$72b41e30$581c5a90$@accra.ca>

We have 2 offices.  Office A has the freeswitch server and some phones.
"Office B" is new and we have 2 phones that we want to connect to the
Freeswitch server located in office A so that calls can be made by the staff
using Freeswitch.  The question is how is this done?   Here is how I see it
but it does not seem to work.  Please help!

 

Office A  has the Freeswitch server which connects to an external VOIP
provider.   The Internal clients connect via port 5060 to the Freeswitch
internal profile. There phones work no problems.  The IP for the Freeswitch
server in  office A is 192.168.50.50.  The Public IP is 142.xx.xxx.xxx.

 

Office B  has 2 phones both are configured to connect to the public IP of
"office A"  on port 5060  which is routed to the freeswitch server.   The
Phones have a local IP of 192.168.23.6 and 9. 

 

The issue is that the phones are never able to register ( both side have
identical settings) or transmit a call from one office to the other.
Freeswitch does not seem to know how to route the calls to the other office.
Ultimately we would like the staff from "office B" to be able to receive and
make calls coming into the Freeswitch server.  

 

1)      Is this not possible over the Internet?  Do we need to set up a VPN
between both offices for this to work?  Is there a way to tell freeswitch
that ext 340 is located in office B? 

 

 

Thanks

 

cjb

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From davide at internetone.it  Tue Aug 27 21:31:55 2013
From: davide at internetone.it (Davide Rossi)
Date: Tue, 27 Aug 2013 17:31:55 +0000
Subject: [Freeswitch-users] R:  xml_curl bindings=phrases issue
In-Reply-To: <4B84AA44-096E-4D3C-AEF6-33F0A140D283@jerris.com>
References: <9041cca8c2bf482bb0391fd6ffc3f356@mail.internetone.local>
	<4B84AA44-096E-4D3C-AEF6-33F0A140D283@jerris.com>
Message-ID: <2df99456acb64a7e948d7321da44a526@mail.internetone.local>

Actually I already tried reloadxml with "xml_curl debug_on" and I don't see any file generated, as it was with other types of bindings.

> -----Messaggio originale-----
> Da: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] Per conto di Michael Jerris
> Inviato: marted? 27 agosto 2013 19:10
> A: FreeSWITCH Users Help
> Oggetto: Re: [Freeswitch-users] xml_curl bindings=phrases issue
> 
> Phrases area not loaded on demand, they are all loaded in the initial config load.  It
> requires a reloadxml to re-load them.
> 
> On Aug 27, 2013, at 12:50 PM, Davide Rossi <davide at internetone.it> wrote:
> 
> > Hi everybody,
> > I'm a newbie using freeswitch (1.2.7) and having an issue with xml_curl phrases:
> reading logs I don't see any request to fetch data and using "xml_curl debug_on" no
> file is generated.
> > Bindings to "configuration" and "dialplan" worked like a charm... Do I have to
> configure something special for "phrases" bindings?
> > I have static IVR menus calling a dynamic phrase macro in
> http://localhost/getphrases.php.
> > Is it a misconfiguration or maybe a bug like http://jira.freeswitch.org/browse/FS-
> 5563?
> > Can't find help on wiki, any hint is really appreciated.
> >
> > ---xml_curl.conf.xml---
> > <configuration name="xml_curl.conf" description="cURL XML Gateway">
> > <bindings> <binding name="phrases">
> > 	<param name="gateway-url" bindings="phrases"
> > value="http://localhost/getphrases.php"/>
> > </binding>
> > </bindings>
> > </configuration>
> >
> > ---console log [7]---
> > [DEBUG] switch_ivr_menu.c:469 Executing IVR menu xxx [DEBUG]
> > switch_ivr_play_say.c:70 No language specified - Using [xx] [ERR]
> > switch_ivr_play_say.c:145 Can't find macro xxx_ivr_main_menu.
> > [WARNING] switch_ivr_play_say.c:348 Macro [xxx_ivr_main_menu]: '' did
> > not match any patterns [DEBUG] switch_ivr_menu.c:363 waiting for 4/4
> > digits t/o 2000 [DEBUG] switch_ivr_menu.c:410 digits ''
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 bdfoster at davri.com  Tue Aug 27 21:54:38 2013
From: bdfoster at davri.com (Brian Foster)
Date: Tue, 27 Aug 2013 13:54:38 -0400
Subject: [Freeswitch-users] Newbie question for remote office
In-Reply-To: <006001cea349$72b41e30$581c5a90$@accra.ca>
References: <006001cea349$72b41e30$581c5a90$@accra.ca>
Message-ID: <CAHgk4BUdX5uNUXuTUbLRJ2pv2Ay9efOWiVG1NEX92bPudkd-pg@mail.gmail.com>

http://wiki.freeswitch.org/wiki/Example_Offsite_phones

In general, you'd need a total of three profiles, all with different
ip/port combos. The third is used for external phones such as at Office B
to authenticate and act just like those on the internal profile.

Start with the documentation above. Let us know if/when you get hung up.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 27, 2013 1:23 PM, "Charles Bujold" <cjbujold at accra.ca> wrote:

> We have 2 offices.  Office A has the freeswitch server and some phones.
> ?Office B? is new and we have 2 phones that we want to connect to the
> Freeswitch server located in office A so that calls can be made by the
> staff using Freeswitch.  The question is how is this done?   Here is how I
> see it but it does not seem to work.  Please help!****
>
> ** **
>
> Office A  has the Freeswitch server which connects to an external VOIP
> provider.   The Internal clients connect via port 5060 to the Freeswitch
> internal profile. There phones work no problems.  The IP for the Freeswitch
> server in  office A is 192.168.50.50.  The Public IP is 142.xx.xxx.xxx.***
> *
>
> ** **
>
> Office B  has 2 phones both are configured to connect to the public IP of
> ?office A?  on port 5060  which is routed to the freeswitch server.   The
> Phones have a local IP of 192.168.23.6 and 9. ****
>
> ** **
>
> The issue is that the phones are never able to register ( both side have
> identical settings) or transmit a call from one office to the other.
> Freeswitch does not seem to know how to route the calls to the other
> office.  Ultimately we would like the staff from ?office B? to be able to
> receive and make calls coming into the Freeswitch server.  ****
>
> ** **
>
> **1)      **Is this not possible over the Internet?  Do we need to set up
> a VPN between both offices for this to work?  Is there a way to tell
> freeswitch that ext 340 is located in office B? ****
>
> ** **
>
> ** **
>
> Thanks****
>
> ** **
>
> cjb****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mike at jerris.com  Tue Aug 27 22:03:37 2013
From: mike at jerris.com (Michael Jerris)
Date: Tue, 27 Aug 2013 14:03:37 -0400
Subject: [Freeswitch-users] xml_curl bindings=phrases issue
In-Reply-To: <2df99456acb64a7e948d7321da44a526@mail.internetone.local>
References: <9041cca8c2bf482bb0391fd6ffc3f356@mail.internetone.local>
	<4B84AA44-096E-4D3C-AEF6-33F0A140D283@jerris.com>
	<2df99456acb64a7e948d7321da44a526@mail.internetone.local>
Message-ID: <E1655D8A-6D42-48EB-A001-88DAEBB6C6AC@jerris.com>

Having taken a closer look here? there is an issue in the code.. its actually looking in "languages" which is correct, but we have the section listed as "phrases"  Can you please open up a jira on this issue.

Mike

On Aug 27, 2013, at 1:31 PM, Davide Rossi <davide at internetone.it> wrote:

> Actually I already tried reloadxml with "xml_curl debug_on" and I don't see any file generated, as it was with other types of bindings.
> 
>> -----Messaggio originale-----
>> Da: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-
>> bounces at lists.freeswitch.org] Per conto di Michael Jerris
>> Inviato: marted? 27 agosto 2013 19:10
>> A: FreeSWITCH Users Help
>> Oggetto: Re: [Freeswitch-users] xml_curl bindings=phrases issue
>> 
>> Phrases area not loaded on demand, they are all loaded in the initial config load.  It
>> requires a reloadxml to re-load them.
>> 
>> On Aug 27, 2013, at 12:50 PM, Davide Rossi <davide at internetone.it> wrote:
>> 
>>> Hi everybody,
>>> I'm a newbie using freeswitch (1.2.7) and having an issue with xml_curl phrases:
>> reading logs I don't see any request to fetch data and using "xml_curl debug_on" no
>> file is generated.
>>> Bindings to "configuration" and "dialplan" worked like a charm... Do I have to
>> configure something special for "phrases" bindings?
>>> I have static IVR menus calling a dynamic phrase macro in
>> http://localhost/getphrases.php.
>>> Is it a misconfiguration or maybe a bug like http://jira.freeswitch.org/browse/FS-
>> 5563?
>>> Can't find help on wiki, any hint is really appreciated.
>>> 
>>> ---xml_curl.conf.xml---
>>> <configuration name="xml_curl.conf" description="cURL XML Gateway">
>>> <bindings> <binding name="phrases">
>>> 	<param name="gateway-url" bindings="phrases"
>>> value="http://localhost/getphrases.php"/>
>>> </binding>
>>> </bindings>
>>> </configuration>
>>> 
>>> ---console log [7]---
>>> [DEBUG] switch_ivr_menu.c:469 Executing IVR menu xxx [DEBUG]
>>> switch_ivr_play_say.c:70 No language specified - Using [xx] [ERR]
>>> switch_ivr_play_say.c:145 Can't find macro xxx_ivr_main_menu.
>>> [WARNING] switch_ivr_play_say.c:348 Macro [xxx_ivr_main_menu]: '' did
>>> not match any patterns [DEBUG] switch_ivr_menu.c:363 waiting for 4/4
>>> digits t/o 2000 [DEBUG] switch_ivr_menu.c:410 digits ''
>> 
>> 
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 fs.user at fordior.net  Tue Aug 27 22:17:59 2013
From: fs.user at fordior.net (EL)
Date: Tue, 27 Aug 2013 20:17:59 +0200
Subject: [Freeswitch-users] Newbie question for remote office
In-Reply-To: <006001cea349$72b41e30$581c5a90$@accra.ca>
References: <006001cea349$72b41e30$581c5a90$@accra.ca>
Message-ID: <20130827181759.GC25035@0rdior.com>

Use a vpn to connect office B as internal (LAN) in office A. 

In /conf/sip_profiles/external.xml:

    <param name="rtp-ip" value="public ip"/>
    <param name="sip-ip" value="public ip"/>
    <param name="ext-rtp-ip" value="public ip"/>
    <param name="ext-sip-ip" value="public ip"/>

In /conf/sip_profiles/internal.xml give the above settings al your
vpn ip. In vars.xml, you might wanne set your domain to the vpn ip.

If you run into issues, use wireshark to analyse your traffic & check
your logs.

-- 
EL


From bdfoster at davri.com  Tue Aug 27 22:27:57 2013
From: bdfoster at davri.com (Brian Foster)
Date: Tue, 27 Aug 2013 14:27:57 -0400
Subject: [Freeswitch-users] Newbie question for remote office
In-Reply-To: <20130827181759.GC25035@0rdior.com>
References: <006001cea349$72b41e30$581c5a90$@accra.ca>
	<20130827181759.GC25035@0rdior.com>
Message-ID: <CAHgk4BWXyKz+4658efa4D2=oyHeHNZvNxgc-q2n_5ZgQ4-Axsg@mail.gmail.com>

If you want to use a VPN great it"s not required but it has nothing to do
with the external profile or the internal profile if the VPN is managed
locally on the machine that has a seperate IP for the VPN. You'd need
another profile exactly like the internal profile but using that extra IP.
Better yet make your router manage the VPN and leave FS alone.

Thank you,

Brian Foster
Project Manager/Owner's Rep.
Davri Investments, Inc.
O: 317-787-2686 x2102
M: 317-600-9753
E: bdfoster at davri.com
Indianapolis, Indiana

Sent from a mobile device.
On Aug 27, 2013 2:23 PM, "EL" <fs.user at fordior.net> wrote:

> Use a vpn to connect office B as internal (LAN) in office A.
>
> In /conf/sip_profiles/external.xml:
>
>     <param name="rtp-ip" value="public ip"/>
>     <param name="sip-ip" value="public ip"/>
>     <param name="ext-rtp-ip" value="public ip"/>
>     <param name="ext-sip-ip" value="public ip"/>
>
> In /conf/sip_profiles/internal.xml give the above settings al your
> vpn ip. In vars.xml, you might wanne set your domain to the vpn ip.
>
> If you run into issues, use wireshark to analyse your traffic & check
> your logs.
>
> --
> EL
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 drk at drkngs.net  Tue Aug 27 22:52:02 2013
From: drk at drkngs.net (Dave R. Kompel)
Date: Tue, 27 Aug 2013 11:52:02 -0700
Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
In-Reply-To: <CAE=3qOZJkN4_RwvSNVXYzsCTYrqr+jhM2Wz00Yo4SB+kLuA_hg@mail.gmail.com>
Message-ID: <20130827185202.583031b6@mail.tritonwest.net>

Hyper-V virtuals are very friendly to any OS running FS. You should have no problems, just make sure you are using the native network adapters, not the emulated ones.  
   
And if you think it kicks ass on your current system, just wait till 2012R2, this Oct 18. You should see how FS performs on a new Generation 2 VM!   
   
--Dave
      _____  

  From: lloyd thomas [mailto:lloydie.t at gmail.com]
To: freeswitch-users [mailto:FreeSWITCH-users at lists.freeswitch.org]
Sent: Tue, 27 Aug 2013 06:10:48 -0700
Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v

  
  
I just need a little advice setting up a freeswitch VM box shared with a http VM on hyper-v host and was wondering whether there are any gotcha's that I should be aware of.


  
I expect about 50 devices to be registered on it at any time and not expecting a high volume of calls.
  
It will be a fairly low spec dual core machine (dell sc1425), with 8gb ram
  

  
Thanks in advance for any advice.
      
   
 
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From drk at drkngs.net  Tue Aug 27 22:55:42 2013
From: drk at drkngs.net (Dave R. Kompel)
Date: Tue, 27 Aug 2013 11:55:42 -0700
Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
In-Reply-To: <1377616567.24060.YahooMailNeo@web126202.mail.ne1.yahoo.com>
Message-ID: <20130827185542.b1bd71f5@mail.tritonwest.net>

That is 100% not true. Have you actually tried it yourself? The only Hypervisor that I have never had any timing issues on guest OSs is Hyper-V.   
   
There were some problems in the HV-Versoin 1 days, but V2 and up, no problems any guest OS.  
   
If you are using linux on Hyper-V, you should not use kernels older then 3.5. This is whan Microsoft started commiting all the HV-* modules directly to the kernel repos.  
   
--Dave
      _____  

  From: Stanislav Sinyagin [mailto:ssinyagin at yahoo.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Tue, 27 Aug 2013 08:16:07 -0700
Subject: Re: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v

  
  
You will likely have audio distortions. OpenVZ or Xen virtualization should do a better job (myself, I run two production servers under XEN and never had problems with audio quality).  

  
Search in Google for FreeSWITCH or Asterisk under Hyper-V, and you will see the whole spectrum of opinions, and no real detailed test results.  

  

  

  

    
  
  
    _____  

  From: lloyd thomas <lloydie.t at gmail.com>
To: freeswitch-users <FreeSWITCH-users at lists.freeswitch.org> 
Sent: Tuesday, August 27, 2013 3:10 PM
Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
  

  
  
  
I just need a little advice setting up a freeswitch VM box shared with a http VM on hyper-v host and was wondering whether there are any gotcha's that I should be aware of.


  
I expect about 50 devices to be registered on it at any time and not expecting a high volume of calls.
  
It will be a fairly low spec dual core machine (dell sc1425), with 8gb ram
  

  
Thanks in advance for any advice.

      
   
 
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From steveayre at gmail.com  Tue Aug 27 23:46:21 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Tue, 27 Aug 2013 20:46:21 +0100
Subject: [Freeswitch-users] Newbie question for remote office
In-Reply-To: <006001cea349$72b41e30$581c5a90$@accra.ca>
References: <006001cea349$72b41e30$581c5a90$@accra.ca>
Message-ID: <CAFiqYumu8X0J9NMgeZ1AygACAbiY5S4w8gNkFhU7=_MiCOWW0A@mail.gmail.com>

Does the REGISTER succeed, but calls routed to the user fail?

NAT is probably your issue. Read
http://wiki.freeswitch.org/wiki/NAT_Traversal

In particular check the Contact header of the REGISTER arriving on the FS
host - it should contain the public IP not the phone's internal one.

Enabling STUN on your phones is probably your best bet providing the phones
support it.

If you can't fix the NAT traversal, then a site-to-site VPN could sidestep
the NAT issue entirely.

-Steve




On 27 August 2013 18:18, Charles Bujold <cjbujold at accra.ca> wrote:

> We have 2 offices.  Office A has the freeswitch server and some phones.
> ?Office B? is new and we have 2 phones that we want to connect to the
> Freeswitch server located in office A so that calls can be made by the
> staff using Freeswitch.  The question is how is this done?   Here is how I
> see it but it does not seem to work.  Please help!****
>
> ** **
>
> Office A  has the Freeswitch server which connects to an external VOIP
> provider.   The Internal clients connect via port 5060 to the Freeswitch
> internal profile. There phones work no problems.  The IP for the Freeswitch
> server in  office A is 192.168.50.50.  The Public IP is 142.xx.xxx.xxx.***
> *
>
> ** **
>
> Office B  has 2 phones both are configured to connect to the public IP of
> ?office A?  on port 5060  which is routed to the freeswitch server.   The
> Phones have a local IP of 192.168.23.6 and 9. ****
>
> ** **
>
> The issue is that the phones are never able to register ( both side have
> identical settings) or transmit a call from one office to the other.
> Freeswitch does not seem to know how to route the calls to the other
> office.  Ultimately we would like the staff from ?office B? to be able to
> receive and make calls coming into the Freeswitch server.  ****
>
> ** **
>
> **1)      **Is this not possible over the Internet?  Do we need to set up
> a VPN between both offices for this to work?  Is there a way to tell
> freeswitch that ext 340 is located in office B? ****
>
> ** **
>
> ** **
>
> Thanks****
>
> ** **
>
> cjb****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 covici at ccs.covici.com  Wed Aug 28 01:02:52 2013
From: covici at ccs.covici.com (covici at ccs.covici.com)
Date: Tue, 27 Aug 2013 17:02:52 -0400
Subject: [Freeswitch-users] Newbie question for remote office
In-Reply-To: <20130827181759.GC25035@0rdior.com>
References: <006001cea349$72b41e30$581c5a90$@accra.ca>
	<20130827181759.GC25035@0rdior.com>
Message-ID: <5667.1377637372@ccs.covici.com>

Much easier to  uncomment what's at the end of public.xml -- then any
phone can register just on port 5060 and he will get authorized and dial
through the default context.

EL <fs.user at fordior.net> wrote:

> Use a vpn to connect office B as internal (LAN) in office A. 
> 
> In /conf/sip_profiles/external.xml:
> 
>     <param name="rtp-ip" value="public ip"/>
>     <param name="sip-ip" value="public ip"/>
>     <param name="ext-rtp-ip" value="public ip"/>
>     <param name="ext-sip-ip" value="public ip"/>
> 
> In /conf/sip_profiles/internal.xml give the above settings al your
> vpn ip. In vars.xml, you might wanne set your domain to the vpn ip.
> 
> If you run into issues, use wireshark to analyse your traffic & check
> your logs.
> 
> -- 
> EL
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
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 msc at freeswitch.org  Wed Aug 28 01:04:47 2013
From: msc at freeswitch.org (Michael S Collins)
Date: Tue, 27 Aug 2013 14:04:47 -0700
Subject: [Freeswitch-users] Evidently I Got Hacked, Please Disregard
Message-ID: <01d801cea369$10c0d6d0$32428470$@freeswitch.org>

Gang,

 

Apologies but it seems someone or something sent out a naughty email. This
looks eerily similar to something I rec'd from TJ Vanderpoel a few days ago.
Please disregard that link while I go buy some Lysol to disinfect my
computer.

 

-MC

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From fs.user at fordior.net  Wed Aug 28 01:20:39 2013
From: fs.user at fordior.net (EL)
Date: Tue, 27 Aug 2013 23:20:39 +0200
Subject: [Freeswitch-users] Newbie question for remote office
In-Reply-To: <5667.1377637372@ccs.covici.com>
References: <006001cea349$72b41e30$581c5a90$@accra.ca>
	<20130827181759.GC25035@0rdior.com>
	<5667.1377637372@ccs.covici.com>
Message-ID: <20130827212039.GA28980@0rdior.com>


> Much easier to  uncomment what's at the end of public.xml -- then any
> phone can register just on port 5060 and he will get authorized and dial
> through the default context.

Als much more unsecure, if you ask me... 

Please note that the provided info was for FS + vpn on the same box
located WAN --> not located inside an office network.

-- 
EL


From ssinyagin at yahoo.com  Wed Aug 28 01:27:02 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Tue, 27 Aug 2013 14:27:02 -0700 (PDT)
Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
In-Reply-To: <A70AC0C1B7E7FD478A6BF19705461AA0C8D8@IMBER.smb.curriegrad2004.ca>
References: <CAE=3qOZJkN4_RwvSNVXYzsCTYrqr+jhM2Wz00Yo4SB+kLuA_hg@mail.gmail.com>	<1377616567.24060.YahooMailNeo@web126202.mail.ne1.yahoo.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C8D8@IMBER.smb.curriegrad2004.ca>
Message-ID: <1377638822.27407.YahooMailNeo@web126205.mail.ne1.yahoo.com>

thanks, good to know.
The only (silly, rhetorical) question is, who in his own mind would run a Windows server to host Linux VM's :-))






>________________________________
> From: Jeff Leung <jleung at v10networks.ca>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Tuesday, August 27, 2013 5:31 PM
>Subject: Re: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
> 
>
>
>I would have to say that one can expect audio distortion issues is just not true anymore.
>?
>With the new Hyper-V integration code present since the 3.5 and 3.8 kernels which Ubuntu 12.04 and 13.04 uses, the Linux kernel can use a synthetic clock source that Hyper-V provides just like KVM does. Even the CentOS 6.4 kernel uses a synthetic clock source too as the guys at Red Hat back ported a lot of changes from the upstream Hyper-V code back to their kernel.
>?
>If the synthetic clock source doesn?t work out for you, there is always the good old mod_timerfd to fix issues like this.
>?
>From:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin
>Sent: Tuesday, August 27, 2013 8:16 AM
>To: FreeSWITCH Users Help
>Subject: Re: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
>?
>You will likely have audio distortions. OpenVZ or Xen virtualization should do a better job (myself, I run two production servers under XEN and never had problems with audio quality).
>?
>Search in Google for FreeSWITCH or Asterisk under Hyper-V, and you will see the whole spectrum of opinions, and no real detailed test results.
>?
>?
>?
>?
>>
>>________________________________
>>
>>From:lloyd thomas <lloydie.t at gmail.com>
>>To: freeswitch-users <FreeSWITCH-users at lists.freeswitch.org> 
>>Sent: Tuesday, August 27, 2013 3:10 PM
>>Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
>>?
>>I just need a little advice setting up a freeswitch VM box shared with a http VM on hyper-v host and was wondering whether there are any gotcha's that I should be aware of.
>>
>>
>>I expect about 50 devices to be registered on it at any time and not expecting a high volume of calls.
>>It will be a fairly low spec dual core machine (dell sc1425), with 8gb ram
>>?
>>Thanks in advance for any advice.
>>?
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 regan at newzealand.co.nz  Wed Aug 28 02:27:35 2013
From: regan at newzealand.co.nz (Regan Yelcich)
Date: Wed, 28 Aug 2013 10:27:35 +1200
Subject: [Freeswitch-users] Roaming Extensions?
Message-ID: <934ED40B-C55D-48A2-8730-6F3B9FF1CFE3@newzealand.co.nz>

Can anyone tell me if there's a roaming extension type feature in FreeSwitch?

ie. the ability for a user to sit down at any phone, sign in, and that phone becomes theirs.

I read about this feature in the FreePBX docs where they describe it as a dynamic user.

Thanks.



From max at nysolutions.com  Wed Aug 28 02:45:14 2013
From: max at nysolutions.com (Moishe Grunstein)
Date: Tue, 27 Aug 2013 22:45:14 +0000
Subject: [Freeswitch-users] Roaming Extensions?
In-Reply-To: <934ED40B-C55D-48A2-8730-6F3B9FF1CFE3@newzealand.co.nz>
References: <934ED40B-C55D-48A2-8730-6F3B9FF1CFE3@newzealand.co.nz>
Message-ID: <BA6A05D48E282C4C9989496521266BA102987A2AEF@ex2.connex.local>

You are looking for hotdesking, see https://wiki.freeswitch.org/wiki/Quick_and_nasty_autoprovisioning_and_dynamic_directories_and_queues_for_Snom_and_Polycom

If you are using FusionPBX as a GUI the feature is there.


Thanks,

Moishe Grunstein
Tornado Computer Systems, Inc.
212.400.7650 888.IPPBX.US
Service Request Email: support at nysolutions.com 
Polycom Certified VAR
Microsoft Small Business Specialist, Cisco SMB Select Certified

Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Regan Yelcich
Sent: Tuesday, August 27, 2013 6:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Roaming Extensions?

Can anyone tell me if there's a roaming extension type feature in FreeSwitch?

ie. the ability for a user to sit down at any phone, sign in, and that phone becomes theirs.

I read about this feature in the FreePBX docs where they describe it as a dynamic user.

Thanks.


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com

 

Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 lloydie.t at gmail.com  Wed Aug 28 03:20:25 2013
From: lloydie.t at gmail.com (lloyd thomas)
Date: Wed, 28 Aug 2013 00:20:25 +0100
Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
In-Reply-To: <1377638822.27407.YahooMailNeo@web126205.mail.ne1.yahoo.com>
References: <CAE=3qOZJkN4_RwvSNVXYzsCTYrqr+jhM2Wz00Yo4SB+kLuA_hg@mail.gmail.com>
	<1377616567.24060.YahooMailNeo@web126202.mail.ne1.yahoo.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C8D8@IMBER.smb.curriegrad2004.ca>
	<1377638822.27407.YahooMailNeo@web126205.mail.ne1.yahoo.com>
Message-ID: <CAE=3qOYCWkHPde+4hU_urag3_MvTvFKMrwi8GWAq0QPN5F8OHA@mail.gmail.com>

Thanks for the advice, I have used hyper-v for http server and spam filter
and have not had a problem yet, so just nneded to make sure FS would not be
a problem.


On 27 August 2013 22:27, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:

> thanks, good to know.
> The only (silly, rhetorical) question is, who in his own mind would run a
> Windows server to host Linux VM's :-))
>
>
>
>   ------------------------------
>  *From:* Jeff Leung <jleung at v10networks.ca>
> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 27, 2013 5:31 PM
>
> *Subject:* Re: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
>
> I would have to say that one can expect audio distortion issues is just
> not true anymore.
>
> With the new Hyper-V integration code present since the 3.5 and 3.8
> kernels which Ubuntu 12.04 and 13.04 uses, the Linux kernel can use a
> synthetic clock source that Hyper-V provides just like KVM does. Even the
> CentOS 6.4 kernel uses a synthetic clock source too as the guys at Red Hat
> back ported a lot of changes from the upstream Hyper-V code back to their
> kernel.
>
> If the synthetic clock source doesn?t work out for you, there is always
> the good old mod_timerfd to fix issues like this.
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Stanislav
> Sinyagin
> *Sent:* Tuesday, August 27, 2013 8:16 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
>
> You will likely have audio distortions. OpenVZ or Xen virtualization
> should do a better job (myself, I run two production servers under XEN and
> never had problems with audio quality).
>
> Search in Google for FreeSWITCH or Asterisk under Hyper-V, and you will
> see the whole spectrum of opinions, and no real detailed test results.
>
>
>
>
>
> ------------------------------
> *From:* lloyd thomas <lloydie.t at gmail.com>
> *To:* freeswitch-users <FreeSWITCH-users at lists.freeswitch.org>
> *Sent:* Tuesday, August 27, 2013 3:10 PM
> *Subject:* [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
>
> I just need a little advice setting up a freeswitch VM box shared with a
> http VM on hyper-v host and was wondering whether there are any gotcha's
> that I should be aware of.
>
> I expect about 50 devices to be registered on it at any time and not
> expecting a high volume of calls.
> It will be a fairly low spec dual core machine (dell sc1425), with 8gb ram
>
> Thanks in advance for any advice.
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 fs.user at fordior.net  Wed Aug 28 03:31:48 2013
From: fs.user at fordior.net (EL)
Date: Wed, 28 Aug 2013 01:31:48 +0200
Subject: [Freeswitch-users] Roaming Extensions?
In-Reply-To: <BA6A05D48E282C4C9989496521266BA102987A2AEF@ex2.connex.local>
References: <934ED40B-C55D-48A2-8730-6F3B9FF1CFE3@newzealand.co.nz>
	<BA6A05D48E282C4C9989496521266BA102987A2AEF@ex2.connex.local>
Message-ID: <20130827233148.GB28980@0rdior.com>

> You are looking for hotdesking, see https://wiki.freeswitch.org/wiki/Quick_and_nasty_autoprovisioning_and_dynamic_directories_and_queues_for_Snom_and_Polycom
> 
> If you are using FusionPBX as a GUI the feature is there.

Is this the only known (working) implementation for hotdesking without a
GUI? Also, are other brands like Yealink not supported?

Thanks for your input.

-- 
EL


From drk at drkngs.net  Wed Aug 28 03:34:23 2013
From: drk at drkngs.net (Dave R. Kompel)
Date: Tue, 27 Aug 2013 16:34:23 -0700
Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
In-Reply-To: <CAE=3qOYCWkHPde+4hU_urag3_MvTvFKMrwi8GWAq0QPN5F8OHA@mail.gmail.com>
Message-ID: <20130827233423.0d7c91c9@mail.tritonwest.net>

Just so you know, I have real production (carrier class) workloads on FS running on HV at a number of client sites. Peak loads process more then 1500CPS at one of them.  
   
If you run into problems feel free to contact me off list, for some help.  
   
--Dave
      _____  

  From: lloyd thomas [mailto:lloydie.t at gmail.com]
To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
Sent: Tue, 27 Aug 2013 16:20:25 -0700
Subject: Re: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v

  
Thanks for the advice, I have used hyper-v for http server and spam filter and have not had a problem yet, so just nneded to make sure FS would not be a problem.
  


  
On 27 August 2013 22:27, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
    
  
thanks, good to know.
The only (silly, rhetorical) question is, who in his own mind would run a Windows server to host Linux VM's :-))

  

  

    
  
  
    _____  

  From: Jeff Leung <jleung at v10networks.ca>
To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
Sent: Tuesday, August 27, 2013 5:31 PM  

Subject: Re: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
  
  

  
  
  
  
I would have to say that one can expect audio distortion issues is just not true anymore.  
   
With the new Hyper-V integration code present since the 3.5 and 3.8 kernels which Ubuntu 12.04 and 13.04 uses, the Linux kernel can use a synthetic clock source that Hyper-V provides just like KVM does. Even the CentOS 6.4 kernel uses a synthetic clock source too as the guys at Red Hat back ported a lot of changes from the upstream Hyper-V code back to their kernel.  
   
If the synthetic clock source doesn?t work out for you, there is always the good old mod_timerfd to fix issues like this.  
   
  
  
  
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Stanislav Sinyagin
Sent: Tuesday, August 27, 2013 8:16 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v  
   
  
  
You will likely have audio distortions. OpenVZ or Xen virtualization should do a better job (myself, I run two production servers under XEN and never had problems with audio quality).  
  
   
  
Search in Google for FreeSWITCH or Asterisk under Hyper-V, and you will see the whole spectrum of opinions, and no real detailed test results.  
  
   
  
   
  
   
    
   
  
  
  
    _____  

    
From: lloyd thomas <lloydie.t at gmail.com>
To: freeswitch-users <FreeSWITCH-users at lists.freeswitch.org> 
Sent: Tuesday, August 27, 2013 3:10 PM
Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v  
  
   
  
  
  
I just need a little advice setting up a freeswitch VM box shared with a http VM on hyper-v host and was wondering whether there are any gotcha's that I should be aware of.

  
  
I expect about 50 devices to be registered on it at any time and not expecting a high volume of calls.  
  
It will be a fairly low spec dual core machine (dell sc1425), with 8gb ram  
  
   
  
Thanks in advance for any advice.  
 
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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



_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 jleung at v10networks.ca  Wed Aug 28 04:40:21 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Tue, 27 Aug 2013 17:40:21 -0700
Subject: [Freeswitch-users] Evidently I Got Hacked, Please Disregard
In-Reply-To: <01d801cea369$10c0d6d0$32428470$@freeswitch.org>
References: <01d801cea369$10c0d6d0$32428470$@freeswitch.org>
Message-ID: <001801cea387$2df05280$89d0f780$@v10networks.ca>

>From the looks of it you got phished or XSS'd from the wget output of that
link ;)

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
S Collins
Sent: Tuesday, August 27, 2013 2:05 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Evidently I Got Hacked, Please Disregard

 

Gang,

 

Apologies but it seems someone or something sent out a naughty email. This
looks eerily similar to something I rec'd from TJ Vanderpoel a few days ago.
Please disregard that link while I go buy some Lysol to disinfect my
computer.

 

-MC

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From soeren.sprenger at aerea.de  Wed Aug 28 10:38:07 2013
From: soeren.sprenger at aerea.de (=?ISO-8859-1?Q?S=F6ren_Sprenger?=)
Date: Wed, 28 Aug 2013 08:38:07 +0200
Subject: [Freeswitch-users] Roaming Extensions?
In-Reply-To: <20130827233148.GB28980@0rdior.com>
References: <934ED40B-C55D-48A2-8730-6F3B9FF1CFE3@newzealand.co.nz>
	<BA6A05D48E282C4C9989496521266BA102987A2AEF@ex2.connex.local>
	<20130827233148.GB28980@0rdior.com>
Message-ID: <521D9ACF.4010000@aerea.de>

Hi,

On 28.08.2013 01:31, EL wrote:
> Is this the only known (working) implementation for hotdesking without
> a GUI? Also, are other brands like Yealink not supported? 

no, you will need to implement a provisioning service by yourself. You
can deploy the configuration to the yealink phones using an http/https
server - may tftp will work on many kinds of phones, too.

  Soeren

-- 
AereA NetworX UG (haftungsbeschr?nkt)
Im Camisch 8
07768 Kahla
Gesch?ftsf?hrerin: Franziska Sprenger
Handelsregister: Jena B504724
E-Mail: info at aerea.de
Web: http://www.aerea.de
Fon: +49 (0) 36424 760823
Fax: +49 (0) 36651 1390009



From hunterj91 at hotmail.com  Wed Aug 28 11:13:02 2013
From: hunterj91 at hotmail.com (Jonathan Hunter)
Date: Wed, 28 Aug 2013 07:13:02 +0000
Subject: [Freeswitch-users] RFC4579- Adding a Participant by Focus- Invite
	with contact ; isfocus
In-Reply-To: <DUB121-W49C018631A3F6B4749DB92BD4C0@phx.gbl>
References: <DUB121-W49C018631A3F6B4749DB92BD4C0@phx.gbl>
Message-ID: <DUB121-W149E10FE52746B6E4CE687BD4B0@phx.gbl>

Hi Guys,
Any comments on this please out there? 
Do I need to create Jira for this?
Many thanks
Jon

From: hunterj91 at hotmail.com
To: freeswitch-users at lists.freeswitch.org
Subject: RFC4579- Adding a Participant by Focus- Invite with contact ;isfocus
Date: Wed, 21 Aug 2013 15:00:11 +0000




Hi Guys,
I previously posted questions about getting RFC4575/4579 working with Freeswitch, and in terms of SUBSCRIBE/NOTIFY this is working fine so thanks very much for the help.
 I just wondered if it had been tested, in particular around RFC4579, and section 5.2;
 5.2.  INVITE: Adding a Participant by the Focus - Dial-Out
   To directly add a participant to a conference, a focus SHOULD send an   INVITE to the participant containing a Contact header field with the   conference URI and the 'isfocus' feature parameter.

I have tested this, I cant see ;isfocus on the contact header field, when getting freeswitch to add a participant using originate, be it manually or via api.
Is there again any particular syntax required for this to work, or has it been tested, as I just see the standard contact header of <sip:mod_sofia at xx.xx.xx.xx:5080> and no ;isfocus.
Many thanks
Jon 		 	   		   		 	   		  
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From lloydie.t at gmail.com  Wed Aug 28 11:45:25 2013
From: lloydie.t at gmail.com (lloyd thomas)
Date: Wed, 28 Aug 2013 08:45:25 +0100
Subject: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
In-Reply-To: <20130827233423.0d7c91c9@mail.tritonwest.net>
References: <CAE=3qOYCWkHPde+4hU_urag3_MvTvFKMrwi8GWAq0QPN5F8OHA@mail.gmail.com>
	<20130827233423.0d7c91c9@mail.tritonwest.net>
Message-ID: <CAE=3qOadbwb6D5SzyJttVzpE7=YX0yUzaWarKL-te1=dDLuaQQ@mail.gmail.com>

Dave, nice to know that it is possible and thanks for your offer. I hope to
use it to provide small scale virtual pbx to a few clients. Will start
building the FS box in a couple of days.


On 28 August 2013 00:34, Dave R. Kompel <drk at drkngs.net> wrote:

> **
> Just so you know, I have real production (carrier class) workloads on FS
> running on HV at a number of client sites. Peak loads process more then
> 1500CPS at one of them.
>
> If you run into problems feel free to contact me off list, for some help.
>
> --Dave
>
>  ------------------------------
> *From:* lloyd thomas [mailto:lloydie.t at gmail.com]
> *To:* FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org]
> *Sent:* Tue, 27 Aug 2013 16:20:25 -0700
>
> *Subject:* Re: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
>
> Thanks for the advice, I have used hyper-v for http server and spam filter
> and have not had a problem yet, so just nneded to make sure FS would not be
> a problem.
>
>
> On 27 August 2013 22:27, Stanislav Sinyagin <ssinyagin at yahoo.com> wrote:
>
>>  thanks, good to know.
>> The only (silly, rhetorical) question is, who in his own mind would run a
>> Windows server to host Linux VM's :-))
>>
>>
>>
>>   ------------------------------
>> *From:* Jeff Leung <jleung at v10networks.ca>
>> *To:* FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org>
>> *Sent:* Tuesday, August 27, 2013 5:31 PM
>>
>> *Subject:* Re: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
>>
>>   I would have to say that one can expect audio distortion issues is
>> just not true anymore.
>>
>> With the new Hyper-V integration code present since the 3.5 and 3.8
>> kernels which Ubuntu 12.04 and 13.04 uses, the Linux kernel can use a
>> synthetic clock source that Hyper-V provides just like KVM does. Even the
>> CentOS 6.4 kernel uses a synthetic clock source too as the guys at Red Hat
>> back ported a lot of changes from the upstream Hyper-V code back to their
>> kernel.
>>
>> If the synthetic clock source doesn?t work out for you, there is always
>> the good old mod_timerfd to fix issues like this.
>>
>>   *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Stanislav
>> Sinyagin
>> *Sent:* Tuesday, August 27, 2013 8:16 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
>>
>>  You will likely have audio distortions. OpenVZ or Xen virtualization
>> should do a better job (myself, I run two production servers under XEN and
>> never had problems with audio quality).
>>
>>  Search in Google for FreeSWITCH or Asterisk under Hyper-V, and you will
>> see the whole spectrum of opinions, and no real detailed test results.
>>
>>
>>
>>
>>
>>   ------------------------------
>>  *From:* lloyd thomas <lloydie.t at gmail.com>
>> *To:* freeswitch-users <FreeSWITCH-users at lists.freeswitch.org>
>> *Sent:* Tuesday, August 27, 2013 3:10 PM
>> *Subject:* [Freeswitch-users] freeswitch on ubuntu VM on hyper-v
>>
>>   I just need a little advice setting up a freeswitch VM box shared with
>> a http VM on hyper-v host and was wondering whether there are any gotcha's
>> that I should be aware of.
>>
>>  I expect about 50 devices to be registered on it at any time and not
>> expecting a high volume of calls.
>>  It will be a fairly low spec dual core machine (dell sc1425), with 8gb
>> ram
>>
>>  Thanks in advance for any advice.
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 at cassidywebservices.co.uk  Wed Aug 28 12:23:23 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Wed, 28 Aug 2013 09:23:23 +0100
Subject: [Freeswitch-users] Roaming Extensions?
In-Reply-To: <521D9ACF.4010000@aerea.de>
References: <934ED40B-C55D-48A2-8730-6F3B9FF1CFE3@newzealand.co.nz>
	<BA6A05D48E282C4C9989496521266BA102987A2AEF@ex2.connex.local>
	<20130827233148.GB28980@0rdior.com> <521D9ACF.4010000@aerea.de>
Message-ID: <CAMbW9oKAECac-Mbs6w7NzU-+YT97V_wB99jo5hdw6rBXOSN8Nw@mail.gmail.com>

You could quite easily build something into freeswitch yourself. For
example have usernames rather than extensions for the phones and have users
dial a number and enter a pin to 'log in' to the phone, updating a database
to which phone they're logged in to, then modify your dialplans to look up
the extension in the database and call the correct phone.

There are many different ways of implmenting this, either directly in the
dialplan using the db application or external code through
mod_lua/mod_python etc.

This approach should in theory support all makes and models of phone.


On 28 August 2013 07:38, S?ren Sprenger <soeren.sprenger at aerea.de> wrote:

> Hi,
>
> On 28.08.2013 01:31, EL wrote:
> > Is this the only known (working) implementation for hotdesking without
> > a GUI? Also, are other brands like Yealink not supported?
>
> no, you will need to implement a provisioning service by yourself. You
> can deploy the configuration to the yealink phones using an http/https
> server - may tftp will work on many kinds of phones, too.
>
>   Soeren
>
> --
> AereA NetworX UG (haftungsbeschr?nkt)
> Im Camisch 8
> 07768 Kahla
> Gesch?ftsf?hrerin: Franziska Sprenger
> Handelsregister: Jena B504724
> E-Mail: info at aerea.de
> Web: http://www.aerea.de
> Fon: +49 (0) 36424 760823
> Fax: +49 (0) 36651 1390009
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>



-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From ssinyagin at yahoo.com  Wed Aug 28 12:37:16 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Wed, 28 Aug 2013 01:37:16 -0700 (PDT)
Subject: [Freeswitch-users] Roaming Extensions?
In-Reply-To: <CAMbW9oKAECac-Mbs6w7NzU-+YT97V_wB99jo5hdw6rBXOSN8Nw@mail.gmail.com>
References: <934ED40B-C55D-48A2-8730-6F3B9FF1CFE3@newzealand.co.nz>	<BA6A05D48E282C4C9989496521266BA102987A2AEF@ex2.connex.local>	<20130827233148.GB28980@0rdior.com>
	<521D9ACF.4010000@aerea.de>
	<CAMbW9oKAECac-Mbs6w7NzU-+YT97V_wB99jo5hdw6rBXOSN8Nw@mail.gmail.com>
Message-ID: <1377679036.25915.YahooMailNeo@web126204.mail.ne1.yahoo.com>

+1, just wanted to write the same.


A simple IVR where the user would identify themselves, then it updates some internal database, and the rest of the dialplan looks up the user in this database when the calls are routed to them. Also for outgoing calls, effective caller ID could be set according to the current user status. 


I guess some hardware phones are able to display messages on their LCD screens, and then you can also tell the user their name and login status.





>________________________________
> From: Andrew Cassidy <andrew at cassidywebservices.co.uk>
>To: FreeSWITCH Users Help <freeswitch-users at lists.freeswitch.org> 
>Sent: Wednesday, August 28, 2013 10:23 AM
>Subject: Re: [Freeswitch-users] Roaming Extensions?
> 
>
>
>You could quite easily build something into freeswitch yourself. For example have usernames rather than extensions for the phones and have users dial a number and enter a pin to 'log in' to the phone, updating a database to which phone they're logged in to, then modify your dialplans to look up the extension in the database and call the correct phone.
>
>
>There are many different ways of implmenting this, either directly in the dialplan using the db application or external code through mod_lua/mod_python etc.
>
>
>This approach should in theory support all makes and models of phone.
>
>
>
>On 28 August 2013 07:38, S?ren Sprenger <soeren.sprenger at aerea.de> wrote:
>
>Hi,
>>
>>
>>On 28.08.2013 01:31, EL wrote:
>>> Is this the only known (working) implementation for hotdesking without
>>> a GUI? Also, are other brands like Yealink not supported?
>>
>>no, you will need to implement a provisioning service by yourself. You
>>can deploy the configuration to the yealink phones using an http/https
>>server - may tftp will work on many kinds of phones, too.
>>
>>? Soeren
>>
>>--
>>AereA NetworX UG (haftungsbeschr?nkt)
>>Im Camisch 8
>>07768 Kahla
>>Gesch?ftsf?hrerin: Franziska Sprenger
>>Handelsregister: Jena B504724
>>E-Mail: info at aerea.de
>>Web: http://www.aerea.de
>>Fon: +49 (0) 36424 760823
>>Fax: +49 (0) 36651 1390009
>>
>>
>>
>>_________________________________________________________________________
>>Professional FreeSWITCH Consulting Services:
>>consulting at freeswitch.org
>>http://www.freeswitchsolutions.com
>>
>>
>>
>>
>>Official FreeSWITCH Sites
>>http://www.freeswitch.org
>>http://wiki.freeswitch.org
>>http://www.cluecon.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
>>
>
>
>
>-- 
>Andrew Cassidy BSc (Hons) MBCS SSCA
>Managing Director
>
>
>
>
>T?03300 100 960? F?03300 100 961
>E?andrew at cassidywebservices.co.uk
>W?www.cassidywebservices.co.uk
>_________________________________________________________________________
>Professional FreeSWITCH Consulting Services:
>consulting at freeswitch.org
>http://www.freeswitchsolutions.com
>
>
>
>
>Official FreeSWITCH Sites
>http://www.freeswitch.org
>http://wiki.freeswitch.org
>http://www.cluecon.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 georgi_mei at abv.bg  Wed Aug 28 14:46:05 2013
From: georgi_mei at abv.bg (Georgi Stefanov)
Date: Wed, 28 Aug 2013 13:46:05 +0300 (EEST)
Subject: [Freeswitch-users] Fw: SQL error on 'show calls count'
Message-ID: <1433076356.232024.1377686765651.JavaMail.apache@mail22.abv.bg>

 Hello all,

I have just found something strange

Please see the following

freeswitch at internal> status
UP 0 years, 0 days, 0 hours, 0 minutes, 20 seconds, 669 milliseconds, 247 microseconds
FreeSWITCH (Version 1.4.0 ) is ready
0 session(s) since startup
0 session(s) - 0 out of max 2200 per sec
3000 session(s) max
min idle cpu 0.00/94.00
Current Stack Size/Max 240K/8192K

freeswitch at internal> version
FreeSWITCH Version 1.4.0 ()

freeswitch at internal> show calls count

0 total.

2013-08-28 13:40:28.554849 [ERR] switch_core_sqldb.c:1126 SQL ERR: [select count(*) from basic_calls where hostname='my-domain-name-as-example'] database disk image is malformed
freeswitch at internal>


How this happened?
1.I have two freeswitches (same version 1.4.0 installed yesterday)
2.2000 calls from FS1-> FS2
3.FS3 just died
4.I have launched FS2 again (freeswitch -nc)
5.Open fs_cli
6.Execute the commands shown in listing above

Is this bug?
Am I missing something?

p.s. Forgive me if this is not the right list to post this message


From fs.user at fordior.net  Wed Aug 28 14:46:52 2013
From: fs.user at fordior.net (EL)
Date: Wed, 28 Aug 2013 12:46:52 +0200
Subject: [Freeswitch-users] Roaming Extensions?
In-Reply-To: <1377679036.25915.YahooMailNeo@web126204.mail.ne1.yahoo.com>
References: <934ED40B-C55D-48A2-8730-6F3B9FF1CFE3@newzealand.co.nz>
	<BA6A05D48E282C4C9989496521266BA102987A2AEF@ex2.connex.local>
	<20130827233148.GB28980@0rdior.com> <521D9ACF.4010000@aerea.de>
	<CAMbW9oKAECac-Mbs6w7NzU-+YT97V_wB99jo5hdw6rBXOSN8Nw@mail.gmail.com>
	<1377679036.25915.YahooMailNeo@web126204.mail.ne1.yahoo.com>
Message-ID: <20130828104652.GA31829@0rdior.com>

Thanks for the input Soeren, Andrew and Stanislav. 

I'm kind of wondering that nobody has implemented a clean method of
hotdesking yet and/or documented to relevant configuration. Something
about reinventing the wheel so to say... ;)

I'm currently not in an urgent need of hotdesking, but that might change
in the near future.
-- 
EL


From GB at cm.nl  Wed Aug 28 17:02:58 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Wed, 28 Aug 2013 15:02:58 +0200
Subject: [Freeswitch-users] Get B-Leg Call-ID from Dialplan
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD1A274@clubvirtual40.ClubMessage.local>

Hello,

I'm using the below extension to export the A-Leg Call-ID to the B-Leg. The A-Leg  Call-ID is visible in the B-Leg INVITE as the header X-OCallID.

    <extension name="external_to_internal">
      <condition field="destination_number" expression="(\d+)$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="export" data="sip_h_X-OCallID=${sip_call_id}" />
        <action application="bridge" data="sofia/internal/${destination_number}@$${sip_proxy_ip}:$${sip_proxy_port}"/>
      </condition>
    </extension>

Next to this, I also want to be able to append the Call-ID of the B-Leg to a custom header of the A-Leg.
The B-Leg Call-ID would be appended to a response message, like 100, 200 or 4xx, 5xx, 6xx.
Is this possible? If so, how? I took a look at using Lua hooks in lua.xml.conf, but I can't get the script to work.
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From mehroz.ashraf85 at gmail.com  Wed Aug 28 17:44:44 2013
From: mehroz.ashraf85 at gmail.com (mehroz)
Date: Wed, 28 Aug 2013 06:44:44 -0700 (PDT)
Subject: [Freeswitch-users] ZRTP : Identical SAS with default media mode?
Message-ID: <1377697484899-7594388.post@n2.nabble.com>

HI All, 

Is it possible to get same SAS with default media mode i,e when media is
processed ?

Assuming action application="set" data="zrtp_enrollment=true" , should set
the FS as trusted MiTM, and pass the same SAS to called party! 
But I don't see that behaviour  !



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ZRTP-Identical-SAS-with-default-media-mode-tp7594388.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From lloyd.aloysius at sunteltech.ca  Wed Aug 28 17:58:13 2013
From: lloyd.aloysius at sunteltech.ca (Lloyd Aloysius)
Date: Wed, 28 Aug 2013 09:58:13 -0400
Subject: [Freeswitch-users] Fw: SQL error on 'show calls count'
In-Reply-To: <1433076356.232024.1377686765651.JavaMail.apache@mail22.abv.bg>
References: <1433076356.232024.1377686765651.JavaMail.apache@mail22.abv.bg>
Message-ID: <CAGvuWYabOpdXOdK_wegwF3ni1zYr-ny5+ZbK_FzbcDVxX6OPHw@mail.gmail.com>

for 2000 calls, please change the default database to mysql or postgresql

Lloyd
*
*
*  <http://www.sunteltech.ca/blog/>*


On Wed, Aug 28, 2013 at 6:46 AM, Georgi Stefanov <georgi_mei at abv.bg> wrote:

>  Hello all,
>
> I have just found something strange
>
> Please see the following
>
> freeswitch at internal> status
> UP 0 years, 0 days, 0 hours, 0 minutes, 20 seconds, 669 milliseconds, 247
> microseconds
> FreeSWITCH (Version 1.4.0 ) is ready
> 0 session(s) since startup
> 0 session(s) - 0 out of max 2200 per sec
> 3000 session(s) max
> min idle cpu 0.00/94.00
> Current Stack Size/Max 240K/8192K
>
> freeswitch at internal> version
> FreeSWITCH Version 1.4.0 ()
>
> freeswitch at internal> show calls count
>
> 0 total.
>
> 2013-08-28 13:40:28.554849 [ERR] switch_core_sqldb.c:1126 SQL ERR: [select
> count(*) from basic_calls where hostname='my-domain-name-as-example']
> database disk image is malformed
> freeswitch at internal>
>
>
> How this happened?
> 1.I have two freeswitches (same version 1.4.0 installed yesterday)
> 2.2000 calls from FS1-> FS2
> 3.FS3 just died
> 4.I have launched FS2 again (freeswitch -nc)
> 5.Open fs_cli
> 6.Execute the commands shown in listing above
>
> Is this bug?
> Am I missing something?
>
> p.s. Forgive me if this is not the right list to post this message
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 davide at internetone.it  Wed Aug 28 12:56:15 2013
From: davide at internetone.it (Davide Rossi)
Date: Wed, 28 Aug 2013 08:56:15 +0000
Subject: [Freeswitch-users] R:  xml_curl bindings=phrases issue
In-Reply-To: <E1655D8A-6D42-48EB-A001-88DAEBB6C6AC@jerris.com>
References: <9041cca8c2bf482bb0391fd6ffc3f356@mail.internetone.local>
	<4B84AA44-096E-4D3C-AEF6-33F0A140D283@jerris.com>
	<2df99456acb64a7e948d7321da44a526@mail.internetone.local>
	<E1655D8A-6D42-48EB-A001-88DAEBB6C6AC@jerris.com>
Message-ID: <c8e30afee40f4c3caba5809c74e4cb2f@mail.internetone.local>

Thanks Mike for your suggestion.
I have just opened an issue on jira (http://jira.freeswitch.org/browse/FS-5734).
Hoping somebody will help with that soon.

Davide

> -----Messaggio originale-----
> Da: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-
> bounces at lists.freeswitch.org] Per conto di Michael Jerris
> Inviato: marted? 27 agosto 2013 20:04
> A: FreeSWITCH Users Help
> Oggetto: Re: [Freeswitch-users] xml_curl bindings=phrases issue
> 
> Having taken a closer look here. there is an issue in the code.. its actually looking in
> "languages" which is correct, but we have the section listed as "phrases"  Can you
> please open up a jira on this issue.
> 
> Mike
> 
> On Aug 27, 2013, at 1:31 PM, Davide Rossi <davide at internetone.it> wrote:
> 
> > Actually I already tried reloadxml with "xml_curl debug_on" and I don't see any file
> generated, as it was with other types of bindings.
> >
> >> -----Messaggio originale-----
> >> Da: freeswitch-users-bounces at lists.freeswitch.org
> >> [mailto:freeswitch-users- bounces at lists.freeswitch.org] Per conto di
> >> Michael Jerris
> >> Inviato: marted? 27 agosto 2013 19:10
> >> A: FreeSWITCH Users Help
> >> Oggetto: Re: [Freeswitch-users] xml_curl bindings=phrases issue
> >>
> >> Phrases area not loaded on demand, they are all loaded in the initial
> >> config load.  It requires a reloadxml to re-load them.
> >>
> >> On Aug 27, 2013, at 12:50 PM, Davide Rossi <davide at internetone.it> wrote:
> >>
> >>> Hi everybody,
> >>> I'm a newbie using freeswitch (1.2.7) and having an issue with xml_curl phrases:
> >> reading logs I don't see any request to fetch data and using
> >> "xml_curl debug_on" no file is generated.
> >>> Bindings to "configuration" and "dialplan" worked like a charm... Do
> >>> I have to
> >> configure something special for "phrases" bindings?
> >>> I have static IVR menus calling a dynamic phrase macro in
> >> http://localhost/getphrases.php.
> >>> Is it a misconfiguration or maybe a bug like
> >>> http://jira.freeswitch.org/browse/FS-
> >> 5563?
> >>> Can't find help on wiki, any hint is really appreciated.
> >>>
> >>> ---xml_curl.conf.xml---
> >>> <configuration name="xml_curl.conf" description="cURL XML Gateway">
> >>> <bindings> <binding name="phrases">
> >>> 	<param name="gateway-url" bindings="phrases"
> >>> value="http://localhost/getphrases.php"/>
> >>> </binding>
> >>> </bindings>
> >>> </configuration>
> >>>
> >>> ---console log [7]---
> >>> [DEBUG] switch_ivr_menu.c:469 Executing IVR menu xxx [DEBUG]
> >>> switch_ivr_play_say.c:70 No language specified - Using [xx] [ERR]
> >>> switch_ivr_play_say.c:145 Can't find macro xxx_ivr_main_menu.
> >>> [WARNING] switch_ivr_play_say.c:348 Macro [xxx_ivr_main_menu]: ''
> >>> did not match any patterns [DEBUG] switch_ivr_menu.c:363 waiting for
> >>> 4/4 digits t/o 2000 [DEBUG] switch_ivr_menu.c:410 digits ''
> >>
> >>
> >> _____________________________________________________________________
> >> ____ Professional FreeSWITCH Consulting Services:
> >> consulting at freeswitch.org
> >> http://www.freeswitchsolutions.com
> >>
> >> 
> >> 
> >>
> >> Official FreeSWITCH Sites
> >> http://www.freeswitch.org
> >> http://wiki.freeswitch.org
> >> http://www.cluecon.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-us
> >> ers
> >> http://www.freeswitch.org
> >
> > ______________________________________________________________________
> > ___ Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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-use
> > rs
> > http://www.freeswitch.org
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 krc at retrospekt.dk  Wed Aug 28 13:51:36 2013
From: krc at retrospekt.dk (Kim Rostgaard Christensen)
Date: Wed, 28 Aug 2013 11:51:36 +0200
Subject: [Freeswitch-users] Implications of
	[inbound|outbound]-use-callid-as-uuid in sofia.xml
Message-ID: <521DC828.5010609@retrospekt.dk>

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

Hi list

First post, so I would just like to give a big hat tip to the people
involved - extremely nice project. Well documented, nice community
We recently moved from Asterisk to FreeSWITCH, and have not looked
back since. Asterisk is a very nice project, but AMI is less suited
our needs than ESL.

Today, while mucking about with PJSUA, I discovered the call_id to be
(entirely) different from the uuid assigned to the channel in FS.
While this appears to remedied by the *-use-callid-as-uuid options in
sofia.xml I still wonder if there are any potential
complications/slowdown associated with using the option?

Best
- -- 
Kim Rostgaard Christensen | krc at retrospekt.dk | http://retrospekt.dk
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From soapee01.fs at stubbornroses.com  Wed Aug 28 18:53:13 2013
From: soapee01.fs at stubbornroses.com (soapee01.fs at stubbornroses.com)
Date: Wed, 28 Aug 2013 09:53:13 -0500
Subject: [Freeswitch-users] Roaming Extensions?
In-Reply-To: <20130828104652.GA31829@0rdior.com>
References: <934ED40B-C55D-48A2-8730-6F3B9FF1CFE3@newzealand.co.nz>
	<BA6A05D48E282C4C9989496521266BA102987A2AEF@ex2.connex.local>
	<20130827233148.GB28980@0rdior.com> <521D9ACF.4010000@aerea.de>
	<CAMbW9oKAECac-Mbs6w7NzU-+YT97V_wB99jo5hdw6rBXOSN8Nw@mail.gmail.com>
	<1377679036.25915.YahooMailNeo@web126204.mail.ne1.yahoo.com>
	<20130828104652.GA31829@0rdior.com>
Message-ID: <521E0ED9.1000807@stubbornroses.com>

On 8/28/2013 5:46 AM, EL wrote:
> Thanks for the input Soeren, Andrew and Stanislav.
>
> I'm kind of wondering that nobody has implemented a clean method of
> hotdesking yet and/or documented to relevant configuration. Something
> about reinventing the wheel so to say... ;)
>
> I'm currently not in an urgent need of hotdesking, but that might change
> in the near future.
Exactly, why implement the wheel?  It's all xml and lua in FusionPBX, 
and MPL.  The PHP just generates the xml.

Log In:
http://code.google.com/p/fusionpbx/source/browse/trunk/fusionpbx/app/hot_desking/resources/xml/dialplan/470_hot-desk-login.xml

Log Out:
http://code.google.com/p/fusionpbx/source/browse/trunk/fusionpbx/app/hot_desking/resources/xml/dialplan/475_hot-desk-logout.xml

Lua:
http://code.google.com/p/fusionpbx/source/browse/trunk/fusionpbx/includes/install/scripts/dial_string.lua

config.lua (inlcuded by the above lua script) is generated at install.  
IIRC on the latest versions it contains the db settings as well. My 
older system looks like:
admin_pin = "4123";
sounds_dir = "/usr/local/freeswitch/sounds";
recordings_dir = "/usr/local/freeswitch/recordings";
tmp_dir = "/tmp";

You could always just install fusion on a vm, enable hot desking and 
copy it off and tweak it.  Probably the easiest thing to do.  Mark's 
done some very cool things with lua.  Improvements are always welcome too.

Regards,

James aka soapee01


From anthony.minessale at gmail.com  Wed Aug 28 19:10:23 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Aug 2013 10:10:23 -0500
Subject: [Freeswitch-users] RFC4579- Adding a Participant by Focus-
 Invite with contact ; isfocus
In-Reply-To: <DUB121-W149E10FE52746B6E4CE687BD4B0@phx.gbl>
References: <DUB121-W49C018631A3F6B4749DB92BD4C0@phx.gbl>
	<DUB121-W149E10FE52746B6E4CE687BD4B0@phx.gbl>
Message-ID: <CAKbxfG_fyjO4OO40TtrJPvae=8sXw0hE2spo2sd7xyHQKh-78Q@mail.gmail.com>

I thought I answered this but I don't see it in my history now.

set sip_invite_contact_params to ~isfocus

If you update to latest master you will get that automatically on invites
generated after receiving an REFER.



On Wed, Aug 28, 2013 at 2:13 AM, Jonathan Hunter <hunterj91 at hotmail.com>wrote:

> Hi Guys,
>
> Any comments on this please out there?
>
> Do I need to create Jira for this?
>
> Many thanks
>
> Jon
>
> ------------------------------
> From: hunterj91 at hotmail.com
> To: freeswitch-users at lists.freeswitch.org
> Subject: RFC4579- Adding a Participant by Focus- Invite with contact
> ;isfocus
> Date: Wed, 21 Aug 2013 15:00:11 +0000
>
>
> Hi Guys,
>
> I previously posted questions about getting RFC4575/4579 working with
> Freeswitch, and in terms of SUBSCRIBE/NOTIFY this is working fine so thanks
> very much for the help.
>
>  I just wondered if it had been tested, in particular around RFC4579, and
> section 5.2;
>
>  5.2.  INVITE: Adding a Participant by the Focus - Dial-Out
>
>    To directly add a participant to a conference, a focus SHOULD send an
>    INVITE to the participant containing a Contact header field with the
>    conference URI and the 'isfocus' feature parameter.
>
>
> I have tested this, I cant see ;isfocus on the contact header field, when
> getting freeswitch to add a participant using originate, be it manually or
> via api.
>
> Is there again any particular syntax required for this to work, or has it
> been tested, as I just see the standard contact header of
> <sip:mod_sofia at xx.xx.xx.xx:5080> and no ;isfocus.
>
> Many thanks
>
> Jon
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From fs.user at fordior.net  Wed Aug 28 21:24:19 2013
From: fs.user at fordior.net (EL)
Date: Wed, 28 Aug 2013 19:24:19 +0200
Subject: [Freeswitch-users] Roaming Extensions?
In-Reply-To: <521E0ED9.1000807@stubbornroses.com>
References: <934ED40B-C55D-48A2-8730-6F3B9FF1CFE3@newzealand.co.nz>
	<BA6A05D48E282C4C9989496521266BA102987A2AEF@ex2.connex.local>
	<20130827233148.GB28980@0rdior.com> <521D9ACF.4010000@aerea.de>
	<CAMbW9oKAECac-Mbs6w7NzU-+YT97V_wB99jo5hdw6rBXOSN8Nw@mail.gmail.com>
	<1377679036.25915.YahooMailNeo@web126204.mail.ne1.yahoo.com>
	<20130828104652.GA31829@0rdior.com>
	<521E0ED9.1000807@stubbornroses.com>
Message-ID: <20130828172419.GC31829@0rdior.com>

Hi James,

Thanks a lot for informing us so well about the relevant existing code
for hotdesking. In the near future I will give this a shot.

-- 
EL


From ben at langfeld.co.uk  Wed Aug 28 22:40:34 2013
From: ben at langfeld.co.uk (Ben Langfeld)
Date: Wed, 28 Aug 2013 15:40:34 -0300
Subject: [Freeswitch-users] Implications of
 [inbound|outbound]-use-callid-as-uuid in sofia.xml
In-Reply-To: <521DC828.5010609@retrospekt.dk>
References: <521DC828.5010609@retrospekt.dk>
Message-ID: <CAAyX+KFMQg=b35ST+yx72Xuy_6e5F-HGqqDq+rxxoCLFZ5zG9g@mail.gmail.com>

I would imagine the issue is that an incoming INVITE CallID might not
strictly be unique. I'm not sure if "use-callid-as-uuid" deals with this at
all...


On 28 August 2013 06:51, Kim Rostgaard Christensen <krc at retrospekt.dk>wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi list
>
> First post, so I would just like to give a big hat tip to the people
> involved - extremely nice project. Well documented, nice community
> We recently moved from Asterisk to FreeSWITCH, and have not looked
> back since. Asterisk is a very nice project, but AMI is less suited
> our needs than ESL.
>
> Today, while mucking about with PJSUA, I discovered the call_id to be
> (entirely) different from the uuid assigned to the channel in FS.
> While this appears to remedied by the *-use-callid-as-uuid options in
> sofia.xml I still wonder if there are any potential
> complications/slowdown associated with using the option?
>
> Best
> - --
> Kim Rostgaard Christensen | krc at retrospekt.dk | http://retrospekt.dk
> -----BEGIN PGP SIGNATURE-----
> Version: GnuPG v1.4.11 (GNU/Linux)
> Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/
>
> iQEcBAEBAgAGBQJSHcgfAAoJEKRZG9Ehk6ybKZQH/3MvKC1QBmgX+6mH3fwnZWcI
> ezJ2XxL6EX+09uqBTbXnk2i8AJHMhZbMAokKJuMcyZHfJvOcQiBnj66fK7njPSBX
> A5/6ieXEzsxI+QHSu26t1pnhFLltgvkOGD2wVVf0e+bQhu027zDM6WW46IJtx3oa
> NBK3gC/oFOw/by0ejeEHsWOwpUvtimEdFRgSJFOYA03L+SEQWQn34uoiCC0fMHmS
> yKt1axCU04GDeN9nIEhrJTC+n/B5BiK4FcwbeMeUvj/VrK/+i57XCF+Wnr2qWJjy
> kBo4Rg1nDURw3nfQiBOhJ27ja5G4kUFAfiJzHs0/yjEsl233Ovakl6GIiwvlkpk=
> =+QMf
> -----END PGP SIGNATURE-----
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 Aug 28 23:14:28 2013
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 28 Aug 2013 14:14:28 -0500
Subject: [Freeswitch-users] Implications of
 [inbound|outbound]-use-callid-as-uuid in sofia.xml
In-Reply-To: <CAAyX+KFMQg=b35ST+yx72Xuy_6e5F-HGqqDq+rxxoCLFZ5zG9g@mail.gmail.com>
References: <521DC828.5010609@retrospekt.dk>
	<CAAyX+KFMQg=b35ST+yx72Xuy_6e5F-HGqqDq+rxxoCLFZ5zG9g@mail.gmail.com>
Message-ID: <CAKbxfG_eG+0UsFM9=j1OV6vEuvEiQJdCfjHw-3LixchScNqXVA@mail.gmail.com>

Correct, you are at the mercy of the call-id being unique and uniqueness
cannot be guaranteed.
Many sip endpoints at least try pretty hard to make sure they are unique
but there is really nothing to enforce it.

Alternatively, you do have the sip_call_id variable present so its not that
hard to correlate them.



On Wed, Aug 28, 2013 at 1:40 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:

> I would imagine the issue is that an incoming INVITE CallID might not
> strictly be unique. I'm not sure if "use-callid-as-uuid" deals with this
> at all...
>
>
> On 28 August 2013 06:51, Kim Rostgaard Christensen <krc at retrospekt.dk>wrote:
>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> Hi list
>>
>> First post, so I would just like to give a big hat tip to the people
>> involved - extremely nice project. Well documented, nice community
>> We recently moved from Asterisk to FreeSWITCH, and have not looked
>> back since. Asterisk is a very nice project, but AMI is less suited
>> our needs than ESL.
>>
>> Today, while mucking about with PJSUA, I discovered the call_id to be
>> (entirely) different from the uuid assigned to the channel in FS.
>> While this appears to remedied by the *-use-callid-as-uuid options in
>> sofia.xml I still wonder if there are any potential
>> complications/slowdown associated with using the option?
>>
>> Best
>> - --
>> Kim Rostgaard Christensen | krc at retrospekt.dk | http://retrospekt.dk
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1.4.11 (GNU/Linux)
>> Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/
>>
>> iQEcBAEBAgAGBQJSHcgfAAoJEKRZG9Ehk6ybKZQH/3MvKC1QBmgX+6mH3fwnZWcI
>> ezJ2XxL6EX+09uqBTbXnk2i8AJHMhZbMAokKJuMcyZHfJvOcQiBnj66fK7njPSBX
>> A5/6ieXEzsxI+QHSu26t1pnhFLltgvkOGD2wVVf0e+bQhu027zDM6WW46IJtx3oa
>> NBK3gC/oFOw/by0ejeEHsWOwpUvtimEdFRgSJFOYA03L+SEQWQn34uoiCC0fMHmS
>> yKt1axCU04GDeN9nIEhrJTC+n/B5BiK4FcwbeMeUvj/VrK/+i57XCF+Wnr2qWJjy
>> kBo4Rg1nDURw3nfQiBOhJ27ja5G4kUFAfiJzHs0/yjEsl233Ovakl6GIiwvlkpk=
>> =+QMf
>> -----END PGP SIGNATURE-----
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
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From steveayre at gmail.com  Wed Aug 28 23:50:46 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 28 Aug 2013 20:50:46 +0100
Subject: [Freeswitch-users] Implications of
 [inbound|outbound]-use-callid-as-uuid in sofia.xml
In-Reply-To: <CAKbxfG_eG+0UsFM9=j1OV6vEuvEiQJdCfjHw-3LixchScNqXVA@mail.gmail.com>
References: <521DC828.5010609@retrospekt.dk>
	<CAAyX+KFMQg=b35ST+yx72Xuy_6e5F-HGqqDq+rxxoCLFZ5zG9g@mail.gmail.com>
	<CAKbxfG_eG+0UsFM9=j1OV6vEuvEiQJdCfjHw-3LixchScNqXVA@mail.gmail.com>
Message-ID: <CAFiqYu=EVz5jhZjHtWDk7RVUEu-r5dR1GD+bmmhzYxw7T16qGQ@mail.gmail.com>

Another thing is storage... if your DB structure is assuming UUIDs are of
the normal length, then longer Call-IDs are going to cause you truncation
problems.


On 28 August 2013 20:14, Anthony Minessale <anthony.minessale at gmail.com>wrote:

> Correct, you are at the mercy of the call-id being unique and uniqueness
> cannot be guaranteed.
> Many sip endpoints at least try pretty hard to make sure they are unique
> but there is really nothing to enforce it.
>
> Alternatively, you do have the sip_call_id variable present so its not
> that hard to correlate them.
>
>
>
> On Wed, Aug 28, 2013 at 1:40 PM, Ben Langfeld <ben at langfeld.co.uk> wrote:
>
>> I would imagine the issue is that an incoming INVITE CallID might not
>> strictly be unique. I'm not sure if "use-callid-as-uuid" deals with this
>> at all...
>>
>>
>> On 28 August 2013 06:51, Kim Rostgaard Christensen <krc at retrospekt.dk>wrote:
>>
>>> -----BEGIN PGP SIGNED MESSAGE-----
>>> Hash: SHA1
>>>
>>> Hi list
>>>
>>> First post, so I would just like to give a big hat tip to the people
>>> involved - extremely nice project. Well documented, nice community
>>> We recently moved from Asterisk to FreeSWITCH, and have not looked
>>> back since. Asterisk is a very nice project, but AMI is less suited
>>> our needs than ESL.
>>>
>>> Today, while mucking about with PJSUA, I discovered the call_id to be
>>> (entirely) different from the uuid assigned to the channel in FS.
>>> While this appears to remedied by the *-use-callid-as-uuid options in
>>> sofia.xml I still wonder if there are any potential
>>> complications/slowdown associated with using the option?
>>>
>>> Best
>>> - --
>>> Kim Rostgaard Christensen | krc at retrospekt.dk | http://retrospekt.dk
>>> -----BEGIN PGP SIGNATURE-----
>>> Version: GnuPG v1.4.11 (GNU/Linux)
>>> Comment: Using GnuPG with Thunderbird - http://www.enigmail.net/
>>>
>>> iQEcBAEBAgAGBQJSHcgfAAoJEKRZG9Ehk6ybKZQH/3MvKC1QBmgX+6mH3fwnZWcI
>>> ezJ2XxL6EX+09uqBTbXnk2i8AJHMhZbMAokKJuMcyZHfJvOcQiBnj66fK7njPSBX
>>> A5/6ieXEzsxI+QHSu26t1pnhFLltgvkOGD2wVVf0e+bQhu027zDM6WW46IJtx3oa
>>> NBK3gC/oFOw/by0ejeEHsWOwpUvtimEdFRgSJFOYA03L+SEQWQn34uoiCC0fMHmS
>>> yKt1axCU04GDeN9nIEhrJTC+n/B5BiK4FcwbeMeUvj/VrK/+i57XCF+Wnr2qWJjy
>>> kBo4Rg1nDURw3nfQiBOhJ27ja5G4kUFAfiJzHs0/yjEsl233Ovakl6GIiwvlkpk=
>>> =+QMf
>>> -----END PGP SIGNATURE-----
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveayre at gmail.com  Thu Aug 29 00:17:22 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Wed, 28 Aug 2013 21:17:22 +0100
Subject: [Freeswitch-users] Fw: SQL error on 'show calls count'
In-Reply-To: <1433076356.232024.1377686765651.JavaMail.apache@mail22.abv.bg>
References: <1433076356.232024.1377686765651.JavaMail.apache@mail22.abv.bg>
Message-ID: <CAFiqYumM6L4X8Vxd7+hD+O-kxXr+pasoVHohEGfJfPG0XWtk8g@mail.gmail.com>

I suspect the core.db file was corrupted by an incomplete write when FS2
crashed. Delete the file and restart FS and it'll be recreated.

If you use ODBC and MySQL/PostgreSQL/etc then they'll recover from crashes
better.




On 28 August 2013 11:46, Georgi Stefanov <georgi_mei at abv.bg> wrote:

>  Hello all,
>
> I have just found something strange
>
> Please see the following
>
> freeswitch at internal> status
> UP 0 years, 0 days, 0 hours, 0 minutes, 20 seconds, 669 milliseconds, 247
> microseconds
> FreeSWITCH (Version 1.4.0 ) is ready
> 0 session(s) since startup
> 0 session(s) - 0 out of max 2200 per sec
> 3000 session(s) max
> min idle cpu 0.00/94.00
> Current Stack Size/Max 240K/8192K
>
> freeswitch at internal> version
> FreeSWITCH Version 1.4.0 ()
>
> freeswitch at internal> show calls count
>
> 0 total.
>
> 2013-08-28 13:40:28.554849 [ERR] switch_core_sqldb.c:1126 SQL ERR: [select
> count(*) from basic_calls where hostname='my-domain-name-as-example']
> database disk image is malformed
> freeswitch at internal>
>
>
> How this happened?
> 1.I have two freeswitches (same version 1.4.0 installed yesterday)
> 2.2000 calls from FS1-> FS2
> 3.FS3 just died
> 4.I have launched FS2 again (freeswitch -nc)
> 5.Open fs_cli
> 6.Execute the commands shown in listing above
>
> Is this bug?
> Am I missing something?
>
> p.s. Forgive me if this is not the right list to post this message
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 gabe at gundy.org  Thu Aug 29 01:00:12 2013
From: gabe at gundy.org (Gabriel Gunderson)
Date: Wed, 28 Aug 2013 15:00:12 -0600
Subject: [Freeswitch-users] Recording Duration on EC2
In-Reply-To: <CAC5vAOQFMj4Oykxr-Adymc8HrzDjxKtcEx5bvoYJV24CE4uzwQ@mail.gmail.com>
References: <CAC5vAOQFMj4Oykxr-Adymc8HrzDjxKtcEx5bvoYJV24CE4uzwQ@mail.gmail.com>
Message-ID: <CAMwZ-t+qDQ_t0iLzWiCNR+dVkKfa-MrM6QjOTiVrLqNNzKNetw@mail.gmail.com>

On Mon, Aug 26, 2013 at 11:36 AM, Byron Clark <byron at theclarkfamily.name> wrote:
> I'm using FreeSWITCH on EC2 to call into a conference bridge and record the
> audio from the conference. It works well except for one thing: the recording
> contains all the audio from the call, but the duration of the file is
> shorter than the call. The duration is typically 3-5 seconds shorter than
> the call for each 5 minutes of call. On a recent test, wall clock showed a
> duration of 15:32.93 but the duration of the recording file was 15:19.96.
> The ugly part is that this only happens on EC2 instances where there is some
> CPU steal time (<5%) occurring.

Byron, do you see this in a regular bridge too? Or, is it only in a
conf. room? I know it would be hard to get, but do you know how long
the *actual* call was? Can you get back into the logs and see when
those events happen and give a time based on syslog?


Best,
Gabe


From jayachar88 at gmail.com  Thu Aug 29 01:09:49 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Thu, 29 Aug 2013 02:39:49 +0530
Subject: [Freeswitch-users] mod_gsmopen - extremely choppy announcement on
	placing call
Message-ID: <CAHDL-swmzhEZaGakio2LSgd=Kc7ZqAERObYEejhPSq_DU1GOUg@mail.gmail.com>

Hi,

For the very first call I place using my mobile phone, to the UMTS dongle
that is used as a mod_gsmopen endpoint, I distinctly hear an announcement,
but that is extremely choppy, and I am unable to understand what it says.
Soon after, I get a SEGV -- repeatable and already have a JIRA filed for
same.

What could be the reason for the extremely chopping announcements? Is it
the rather low-end Celeron 1.5GHz w/ 512MB RAM.

Thanks in anticipation for any hints / pointers.

cheers,
Jay
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From terry at digital-outpost.com  Thu Aug 29 05:03:30 2013
From: terry at digital-outpost.com (Terry Barnum)
Date: Wed, 28 Aug 2013 18:03:30 -0700
Subject: [Freeswitch-users] Google voice call fails
In-Reply-To: <006501cea2e0$26b95fe0$742c1fa0$@v10networks.ca>
References: <9665D20D-7235-4058-8666-14A654692503@digital-outpost.com>
	<006501cea2e0$26b95fe0$742c1fa0$@v10networks.ca>
Message-ID: <C782E830-FFB9-4DC9-9BFE-82CBDAB8EDD0@digital-outpost.com>

Sorry, I don't know what's considered "set correctly" in the dingaling profile. Here's what's in jingle_profiles/v_172.16.1.199_gtalk.xml. I've also tried setting ext-rtp-ip to $${external_rtp_ip}.

<include>
  <profile type="client">
    <param name="name" value="gtalk"/>
    <param name="login" value="USER at gmail.com/talk"/>
    <param name="password" value="MYPASS"/>
    <param name="dialplan" value="XML"/>
    <param name="context" value="public"/>
    <param name="message" value="Jingle all the way"/>
    <param name="rtp-ip" value="auto"/>
    <param name="ext-rtp-ip" value="auto-nat"/>
    <param name="auto-login" value="true"/>
    <param name="sasl" value="plain"/>
    <param name="server" value="talk.google.com"/>
    <param name="tls" value="true"/>
    <param name="use-rtp-timer" value="true"/>
    <param name="exten" value="102"/>
    <param name="vad" value="both"/>
    <param name="candidate-acl" value="wan.auto"/>
    <param name="local-network-acl" value="localnet.auto"/>
  </profile>
</include>

Thanks for any pointers.

-Terry

On Aug 26, 2013, at 9:44 PM, Jeff Leung wrote:

> Check to see if your IP addressing settings are set correctly within the
> dingaling profile.
> 
> Most of the times you'll need to have the profile to listen internally but
> have dingaling to specify an external IP address when it talks to the
> outside world.
> 
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
>> users-bounces at lists.freeswitch.org] On Behalf Of Terry Barnum
>> Sent: Monday, August 26, 2013 9:33 PM
>> To: FreeSWITCH Users Help
>> Subject: [Freeswitch-users] Google voice call fails
>> 
>> I've been reading the freeswitch book (great read!) and playing with my
> first
>> freeswitch install on a Mac mini host with freeswitch installed via git
> into a
>> Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones
> and
>> a Yealink T32G, internally between extensions, using a free DID to test
>> incoming calls and a free but limited iptel account to test outbound.
>> 
>> My next step was to try and play with real outbound calls so I followed
> these
>> instructions for Google voice:
>> <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and
>> are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just
>> PCMU but no change. xmpp appears to be connecting and authenticating to
>> google correctly.
>> 
>> The log of an attempted call (edited for privacy) is at
>> <http://pastebin.freeswitch.org/21338>
>> 
>> Snippets of the log that looks suspicious to a freeswitch noob:
>> 
>> 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0
>> rtp candidates.
>> 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0
>> rtcp candidates.
>> 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0
>> video_rtp candidates
>> 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0
>> video_rctp candidates
>> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing
>> session for 2155444888
>> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates
>> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate
>> 74.125.141.127:19305 PASS ACL wan.auto
>> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp
>> Candidate 74.125.141.127:19305
>> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp
>> candidates
>> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates
>> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp
>> candidates
>> 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing
>> session for 2155444888
>> 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup
>> dingaling/gtalk/+7609876543 at voice.google.com
>> 
>> Where should I be looking to try and fix? Freeswitch is behind the
> firewall
>> with a private IP but STUN says it's working (or at least it looks that
> way to
>> me).
>> 
>> Thanks,
>> -Terry
>> 
>> 
>> 
>> __________________________________________________________
>> _______________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>> 
>> 
>> 
>> 
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 byron at theclarkfamily.name  Thu Aug 29 04:11:27 2013
From: byron at theclarkfamily.name (Byron Clark)
Date: Wed, 28 Aug 2013 18:11:27 -0600
Subject: [Freeswitch-users] Recording Duration on EC2
In-Reply-To: <CAMwZ-t+qDQ_t0iLzWiCNR+dVkKfa-MrM6QjOTiVrLqNNzKNetw@mail.gmail.com>
References: <CAC5vAOQFMj4Oykxr-Adymc8HrzDjxKtcEx5bvoYJV24CE4uzwQ@mail.gmail.com>
	<CAMwZ-t+qDQ_t0iLzWiCNR+dVkKfa-MrM6QjOTiVrLqNNzKNetw@mail.gmail.com>
Message-ID: <CAC5vAORqR_0a3RXW8RLiPXoQZUQxXkieGtwjFMRMA4rpY2h4Xw@mail.gmail.com>

On Wed, Aug 28, 2013 at 3:00 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Mon, Aug 26, 2013 at 11:36 AM, Byron Clark <byron at theclarkfamily.name>
> wrote:
> > I'm using FreeSWITCH on EC2 to call into a conference bridge and record
> the
> > audio from the conference. It works well except for one thing: the
> recording
> > contains all the audio from the call, but the duration of the file is
> > shorter than the call. The duration is typically 3-5 seconds shorter than
> > the call for each 5 minutes of call. On a recent test, wall clock showed
> a
> > duration of 15:32.93 but the duration of the recording file was 15:19.96.
> > The ugly part is that this only happens on EC2 instances where there is
> some
> > CPU steal time (<5%) occurring.
>
> Byron, do you see this in a regular bridge too? Or, is it only in a
> conf. room? I know it would be hard to get, but do you know how long
> the *actual* call was? Can you get back into the logs and see when
> those events happen and give a time based on syslog?
>

Thanks, Gabe. I'm seeing this on any call I originate on the problematic
instances. Here are the things I've tried, to no avail, in order to improve
these results:
- Kernel CONFIG_HZ set to 1000 (Ubuntu 12.04 default on EC2 is 250)
- Using mod_timerfd


Here are the results of a recent test I ran where the recorded file was six
seconds short after about four minutes.

Command
---------------
freeswitch at internal> originate {record_waste_resources=true}sofia/external/
sip:music at iptel.org &record(/tmp/record1.wav) default default
+OK e223c3e0-103d-11e3-aeea-0779ec213503

I used uuid_kill to terminate the call after 00:03:57.

Duration of the resulting file
----------------------------------------
Input #0, wav, from '/tmp/record1.wav':
  Duration: 00:03:51.84, bitrate: 128 kb/s
    Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, 1
channels, s16, 128 kb/s

FreeSWITCH Console Output
-------------------------------------------
2013-08-28 23:59:32.417082 [DEBUG] switch_ivr_originate.c:2060 Parsing
global variables
2013-08-28 23:59:32.417082 [DEBUG] switch_event.c:1617 Parsing variable
[record_waste_resources]=[true]
2013-08-28 23:59:32.417082 [NOTICE] switch_channel.c:1030 New Channel
sofia/external/sip:music at iptel.org [e223c3e0-103d-11e3-aeea-0779ec213503]
2013-08-28 23:59:32.417082 [DEBUG] mod_sofia.c:5189 (sofia/external/
sip:music at iptel.org) State Change CS_NEW -> CS_INIT
2013-08-28 23:59:32.417082 [DEBUG] switch_core_session.c:1334 Send signal
sofia/external/sip:music at iptel.org [BREAK]
2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:415
(sofia/external/sip:music at iptel.org) Running State Change CS_INIT
2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:454
(sofia/external/sip:music at iptel.org) State INIT
2013-08-28 23:59:32.417082 [DEBUG] mod_sofia.c:87 sofia/external/
sip:music at iptel.org SOFIA INIT
2013-08-28 23:59:32.417082 [DEBUG] sofia_glue.c:2677 Local SDP:
v=0
o=FreeSWITCH 1377707312 1377707313 IN IP4 10.151.108.236
s=FreeSWITCH
c=IN IP4 10.151.108.236
t=0 0
m=audio 27060 RTP/AVP 0 8 101 13
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

2013-08-28 23:59:32.417082 [DEBUG] switch_core_session.c:999 Send signal
sofia/external/sip:music at iptel.org [BREAK]
2013-08-28 23:59:32.417082 [DEBUG] mod_sofia.c:127 (sofia/external/
sip:music at iptel.org) State Change CS_INIT -> CS_ROUTING
2013-08-28 23:59:32.417082 [DEBUG] switch_core_session.c:1334 Send signal
sofia/external/sip:music at iptel.org [BREAK]
2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:454
(sofia/external/sip:music at iptel.org) State INIT going to sleep
2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:415
(sofia/external/sip:music at iptel.org) Running State Change CS_ROUTING
2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:470
(sofia/external/sip:music at iptel.org) State ROUTING
2013-08-28 23:59:32.417082 [DEBUG] mod_sofia.c:150 sofia/external/
sip:music at iptel.org SOFIA ROUTING
2013-08-28 23:59:32.417082 [DEBUG] switch_ivr_originate.c:67
(sofia/external/sip:music at iptel.org) State Change CS_ROUTING ->
CS_CONSUME_MEDIA
2013-08-28 23:59:32.417082 [DEBUG] switch_core_session.c:1334 Send signal
sofia/external/sip:music at iptel.org [BREAK]
2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:470
(sofia/external/sip:music at iptel.org) State ROUTING going to sleep
2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:415
(sofia/external/sip:music at iptel.org) Running State Change CS_CONSUME_MEDIA
2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:489
(sofia/external/sip:music at iptel.org) State CONSUME_MEDIA
2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:489
(sofia/external/sip:music at iptel.org) State CONSUME_MEDIA going to sleep
2013-08-28 23:59:32.417082 [DEBUG] sofia.c:5720 Channel sofia/external/
sip:music at iptel.org entering state [calling][0]
2013-08-28 23:59:32.917126 [DEBUG] switch_core_session.c:999 Send signal
sofia/external/sip:music at iptel.org [BREAK]
2013-08-28 23:59:32.917126 [DEBUG] switch_core_session.c:999 Send signal
sofia/external/sip:music at iptel.org [BREAK]
2013-08-28 23:59:32.917126 [DEBUG] sofia.c:5720 Channel sofia/external/
sip:music at iptel.org entering state [completing][200]
2013-08-28 23:59:32.917126 [DEBUG] sofia.c:5733 Remote SDP:
v=0
o=- 308297691 1827516553 IN IP4 217.9.36.145
s=-
c=IN IP4 217.9.36.145
t=0 0
m=audio 37892 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=direction:both

2013-08-28 23:59:32.917126 [DEBUG] switch_core_session.c:999 Send signal
sofia/external/sip:music at iptel.org [BREAK]
2013-08-28 23:59:32.917126 [DEBUG] switch_core_session.c:999 Send signal
sofia/external/sip:music at iptel.org [BREAK]
2013-08-28 23:59:32.917126 [DEBUG] sofia.c:5720 Channel sofia/external/
sip:music at iptel.org entering state [ready][200]
2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:5181 Audio Codec Compare
[PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3124 Set Codec
sofia/external/sip:music at iptel.org PCMU/8000 20 ms 160 samples 64000 bits
2013-08-28 23:59:32.917126 [DEBUG] switch_core_codec.c:111 sofia/external/
sip:music at iptel.org Original read codec set to PCMU:0
2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:5303 Set 2833 dtmf send
payload to 101
2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3383 AUDIO RTP
[sofia/external/sip:music at iptel.org] 10.151.108.236 port 27060 ->
217.9.36.145 port 37892 codec: 0 ms: 20
2013-08-28 23:59:32.917126 [DEBUG] switch_rtp.c:1985 Starting timer [soft]
160 bytes per 20ms
2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3647 Set 2833 dtmf send
payload to 101
2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3653 Set 2833 dtmf receive
payload to 101
2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3680 sofia/external/
sip:music at iptel.org Set rtp dtmf delay to 40
2013-08-28 23:59:32.917126 [NOTICE] sofia.c:6476 Channel [sofia/external/
sip:music at iptel.org] has been answered
2013-08-28 23:59:32.917126 [DEBUG] switch_channel.c:3576 (sofia/external/
sip:music at iptel.org) Callstate Change DOWN -> ACTIVE
2013-08-28 23:59:32.937082 [DEBUG] switch_ivr_originate.c:3494 Originate
Resulted in Success: [sofia/external/sip:music at iptel.org]
2013-08-28 23:59:32.937082 [INFO] switch_channel.c:2973 sofia/external/
sip:music at iptel.org Flipping CID from "" <0000000000> to "Outbound Call"
<music>
2013-08-28 23:59:32.937082 [DEBUG] mod_commands.c:4088 (sofia/external/
sip:music at iptel.org) State Change CS_CONSUME_MEDIA -> CS_EXECUTE
2013-08-28 23:59:32.937082 [DEBUG] switch_core_session.c:1334 Send signal
sofia/external/sip:music at iptel.org [BREAK]
2013-08-28 23:59:32.937082 [DEBUG] switch_core_state_machine.c:415
(sofia/external/sip:music at iptel.org) Running State Change CS_EXECUTE
2013-08-28 23:59:32.937082 [DEBUG] switch_core_state_machine.c:477
(sofia/external/sip:music at iptel.org) State EXECUTE
2013-08-28 23:59:32.937082 [DEBUG] mod_sofia.c:243 sofia/external/
sip:music at iptel.org SOFIA EXECUTE
2013-08-28 23:59:32.937082 [DEBUG] switch_core_state_machine.c:209
sofia/external/sip:music at iptel.org Standard EXECUTE
EXECUTE sofia/external/sip:music at iptel.org record(/tmp/record1.wav)
2013-08-28 23:59:32.937082 [DEBUG] switch_ivr_play_say.c:475 Raw Codec
Activated, ready to waste resources!
2013-08-28 23:59:32.937082 [DEBUG] switch_ivr_play_say.c:599 Raw Codec
Activated
2013-08-28 23:59:32.937082 [DEBUG] switch_core_codec.c:219 sofia/external/
sip:music at iptel.org Push codec L16:70
2013-08-28 23:59:33.077079 [DEBUG] switch_rtp.c:3706 Correct ip/port
confirmed.
2013-08-29 00:03:29.537080 [NOTICE] switch_ivr.c:3675 Hangup sofia/external/
sip:music at iptel.org [CS_EXECUTE] [NORMAL_CLEARING]
2013-08-29 00:03:29.537080 [DEBUG] switch_channel.c:3130 Send signal
sofia/external/sip:music at iptel.org [KILL]
2013-08-29 00:03:29.537080 [DEBUG] switch_core_session.c:1334 Send signal
sofia/external/sip:music at iptel.org [BREAK]
2013-08-29 00:03:29.557080 [DEBUG] switch_core_codec.c:244 sofia/external/
sip:music at iptel.org Restore previous codec PCMU:0.
2013-08-29 00:03:29.557080 [DEBUG] switch_core_session.c:2740
sofia/external/sip:music at iptel.org skip receive message
[APPLICATION_EXEC_COMPLETE] (channel is hungup already)
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:477
(sofia/external/sip:music at iptel.org) State EXECUTE going to sleep
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:415
(sofia/external/sip:music at iptel.org) Running State Change CS_HANGUP
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:678
(sofia/external/sip:music at iptel.org) State HANGUP
2013-08-29 00:03:29.557080 [DEBUG] mod_sofia.c:504 Channel sofia/external/
sip:music at iptel.org hanging up, cause: NORMAL_CLEARING
2013-08-29 00:03:29.557080 [DEBUG] mod_sofia.c:556 Sending BYE to
sofia/external/sip:music at iptel.org
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:48
sofia/external/sip:music at iptel.org Standard HANGUP, cause: NORMAL_CLEARING
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:678
(sofia/external/sip:music at iptel.org) State HANGUP going to sleep
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:689
(sofia/external/sip:music at iptel.org) Callstate Change ACTIVE -> HANGUP
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:446
(sofia/external/sip:music at iptel.org) State Change CS_HANGUP -> CS_REPORTING
2013-08-29 00:03:29.557080 [DEBUG] switch_core_session.c:1334 Send signal
sofia/external/sip:music at iptel.org [BREAK]
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:415
(sofia/external/sip:music at iptel.org) Running State Change CS_REPORTING
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:761
(sofia/external/sip:music at iptel.org) State REPORTING
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:92
sofia/external/sip:music at iptel.org Standard REPORTING, cause:
NORMAL_CLEARING
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:761
(sofia/external/sip:music at iptel.org) State REPORTING going to sleep
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:440
(sofia/external/sip:music at iptel.org) State Change CS_REPORTING -> CS_DESTROY
2013-08-29 00:03:29.557080 [DEBUG] switch_core_session.c:1334 Send signal
sofia/external/sip:music at iptel.org [BREAK]
2013-08-29 00:03:29.557080 [DEBUG] switch_core_session.c:1542 Session 6
(sofia/external/sip:music at iptel.org) Locked, Waiting on external entities
2013-08-29 00:03:29.557080 [NOTICE] switch_core_session.c:1560 Session 6
(sofia/external/sip:music at iptel.org) Ended
2013-08-29 00:03:29.557080 [NOTICE] switch_core_session.c:1564 Close
Channel sofia/external/sip:music at iptel.org [CS_DESTROY]
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:565
(sofia/external/sip:music at iptel.org) Callstate Change HANGUP -> DOWN
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:568
(sofia/external/sip:music at iptel.org) Running State Change CS_DESTROY
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:578
(sofia/external/sip:music at iptel.org) State DESTROY
2013-08-29 00:03:29.557080 [DEBUG] mod_sofia.c:397 sofia/external/
sip:music at iptel.org SOFIA DESTROY
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:99
sofia/external/sip:music at iptel.org Standard DESTROY
2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:578
(sofia/external/sip:music at iptel.org) State DESTROY going to sleep
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From sdame at 207me.com  Thu Aug 29 03:30:04 2013
From: sdame at 207me.com (Stephen Dame)
Date: Wed, 28 Aug 2013 19:30:04 -0400
Subject: [Freeswitch-users] Recording Duration on EC2
In-Reply-To: <CAC5vAOQFMj4Oykxr-Adymc8HrzDjxKtcEx5bvoYJV24CE4uzwQ@mail.gmail.com>
References: <CAC5vAOQFMj4Oykxr-Adymc8HrzDjxKtcEx5bvoYJV24CE4uzwQ@mail.gmail.com>
Message-ID: <004201cea446$871d6680$95583380$@207me.com>

Byron,

 

We run a talk radio show based in EC2 with automated 55 minute shows,  I have the same issue and never could figure out why recordings where short?  You never lose full words,  but a trained radio person can hear the missing bits along the way.   Here are the last 20 shows, see below     I had tried both wav and mp3, and messed with settings for energy, write buffers, continuous transmit etc?   Never thought of trying it on bare metal server.    I?m running c1.medium with ubuntu10.04 with 1.06FS.   Overall works great, just need to fill the missing seconds in radio schedule when rebroadcasting.

 

Here is the length of last 20 shows, so how do you monitor steal time?

 

Length:      54:35

Length:      54:37

Length:      54:20

Length:      54:39

Length:      54:38

Length:      54:38

Length:      54:40

Length:      54:38

Length:      54:36

Length:      54:32

Length:      54:45

Length:      54:37

Length:      54:36

Length:      54:36

Length:      54:35

Length:      54:40

Length:      54:42

Length:      54:44

Length:      54:35

Length:      53:57

 

Regards,

Stephen

 

HostBBB ? Online Learning Solutions  http://www.hostbbb.com <http://www.hostbbb.com/> 

207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame

 

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Byron Clark
Sent: Monday, August 26, 2013 1:36 PM
To: FreeSWITCH Users
Subject: [Freeswitch-users] Recording Duration on EC2

 

I'm using FreeSWITCH on EC2 to call into a conference bridge and record the audio from the conference. It works well except for one thing: the recording contains all the audio from the call, but the duration of the file is shorter than the call. The duration is typically 3-5 seconds shorter than the call for each 5 minutes of call. On a recent test, wall clock showed a duration of 15:32.93 but the duration of the recording file was 15:19.96. The ugly part is that this only happens on EC2 instances where there is some CPU steal time (<5%) occurring. 

 

Here's my setup:

Operating System: Ubuntu 12.04

FreeSWITCH: 1.2.12

 

Command I'm using to start the call:

originate {record_waste_resources=true}sofia/external/SIPADDR &record(/tmp/record1.wav) default default

 

Any ideas on how to make the recording duration actually match up with the call duration? Or even what's really going wrong so I can work on fixing that?

 

--
Byron Clark

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From guga.salazar.loor at gmail.com  Thu Aug 29 06:02:09 2013
From: guga.salazar.loor at gmail.com (Gustavo Salazar)
Date: Wed, 28 Aug 2013 21:02:09 -0500
Subject: [Freeswitch-users] Secure Websocket Setup
In-Reply-To: <CAKbxfG9+OCmi0j6Vgf0_S_RdEzm6yHxAMb_SP9wSe37i7ghEnA@mail.gmail.com>
References: <2065652951.99.1377292193211.JavaMail.root@018000web.co>
	<AA86DB46-7D55-4804-B18E-5304E2907694@jerris.com>
	<CAKbxfG9+OCmi0j6Vgf0_S_RdEzm6yHxAMb_SP9wSe37i7ghEnA@mail.gmail.com>
Message-ID: <CAMKwYLCFj=++jwzd1e9B9Nv3NdC=FGA=ahvaUUePg56i26zVKw@mail.gmail.com>

If I want a valid cert/key pair, should I use these commands (steps 2 and
3)
http://wiki.freeswitch.org/wiki/SIP_TLS#Step_1_-_Generate_the_CA_.28Root.29_Certificate
to generate certificate and key?






2013/8/23 Anthony Minessale <anthony.minessale at gmail.com>

> And I don't think it actually works.  You need a valid cert/key pair that
> would work in apach catted together into a pem file.
>
>
>
> On Fri, Aug 23, 2013 at 4:33 PM, Michael Jerris <mike at jerris.com> wrote:
>
>>  We will self-gen a self signed wss.pem if you do not supply one.
>>
>> On Aug 23, 2013, at 5:09 PM, Ricardo Saavedra <rsaavedra at 018000web.co>
>> wrote:
>>
>> Hello,
>>
>> The webrtc setup instructions are:
>>
>> <!-- uncomment for sip over websocket support -->
>> <!--<param name="ws-binding" value=":5066"/>-->
>>
>> <!-- uncomment for sip over secure websocket support -->
>> <!-- You need wss.pem in /usr/local/freeswitch/certs for wss -->
>> <!--<param name="wss-binding" value=":7443"/>-->
>>
>> Now I have a test server for the Websocket and is working. ):
>>
>> For the secure websocket I have some questions:
>>
>> Freeswitch have any tool to generate a self-signed wss.pem file?
>>
>> Can I use the files: usr/local/freeswitch/certs/dtls-srtp.crt and
>> usr/local/freeswitch/certs/ dtls-srtp.key
>>
>> to generate the wss.pem that I need?
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Gustavo Salazar
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From jayachar88 at gmail.com  Thu Aug 29 08:34:44 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Thu, 29 Aug 2013 10:04:44 +0530
Subject: [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd -- preferred
	approach
Message-ID: <CAHDL-szR+_NFAQOmbaT-pc1jfkBWXBmSZLKS9SNU=37XxJF6fw@mail.gmail.com>

Assuming that one has full control over the kernel configuration and FS
build / config options, is there a preferred mechanism between the 2 for
letting FS have access to high resolution timers, i.e.

   - CONFIG_HZ=1000
   - mod_timerfd

This was one step I missed in my FS setup, and I think it might explain the
extremely choppy, broken, garbled announcement I hear when I call into my
gsmopen EP via phone.
J
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From jayachar88 at gmail.com  Thu Aug 29 08:44:23 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Thu, 29 Aug 2013 10:14:23 +0530
Subject: [Freeswitch-users] Recording Duration on EC2
In-Reply-To: <CAC5vAORqR_0a3RXW8RLiPXoQZUQxXkieGtwjFMRMA4rpY2h4Xw@mail.gmail.com>
References: <CAC5vAOQFMj4Oykxr-Adymc8HrzDjxKtcEx5bvoYJV24CE4uzwQ@mail.gmail.com>
	<CAMwZ-t+qDQ_t0iLzWiCNR+dVkKfa-MrM6QjOTiVrLqNNzKNetw@mail.gmail.com>
	<CAC5vAORqR_0a3RXW8RLiPXoQZUQxXkieGtwjFMRMA4rpY2h4Xw@mail.gmail.com>
Message-ID: <CAHDL-syMhd=CX+tE0LCn=uyrQ_6uxg+m0wAW07n5qPhsLUxHkg@mail.gmail.com>

Being a complete n00b with FS, I could be completely off the mark here, but
recently when I was searching around for my extremely poor audio quality
issues with mod_gsmopen, I found reference to timing issues, and then I saw
this thread. Coming from an embedded background, I believe I can extend the
logic that if you run too many instances with high-resolution software
timers on same processor, the resolution of all instances gets impacted,
since nobody would be getting "enough time" (or interrupts, fast enough)...
this could be a clue to chase, if your problem is unique to a particular
EC2 instance (however, I see follow-on posts indicating this to be a more
generic issue).

Here are 2 JIRA's that have interesting content:

http://jira.freeswitch.org/browse/FS-4256
http://jira.freeswitch.org/browse/FS-3290 (this one has a solution
that worked for someone)




On Thu, Aug 29, 2013 at 5:41 AM, Byron Clark <byron at theclarkfamily.name>wrote:

> On Wed, Aug 28, 2013 at 3:00 PM, Gabriel Gunderson <gabe at gundy.org> wrote:
>
>> On Mon, Aug 26, 2013 at 11:36 AM, Byron Clark <byron at theclarkfamily.name>
>> wrote:
>> > I'm using FreeSWITCH on EC2 to call into a conference bridge and record
>> the
>> > audio from the conference. It works well except for one thing: the
>> recording
>> > contains all the audio from the call, but the duration of the file is
>> > shorter than the call. The duration is typically 3-5 seconds shorter
>> than
>> > the call for each 5 minutes of call. On a recent test, wall clock
>> showed a
>> > duration of 15:32.93 but the duration of the recording file was
>> 15:19.96.
>> > The ugly part is that this only happens on EC2 instances where there is
>> some
>> > CPU steal time (<5%) occurring.
>>
>> Byron, do you see this in a regular bridge too? Or, is it only in a
>> conf. room? I know it would be hard to get, but do you know how long
>> the *actual* call was? Can you get back into the logs and see when
>> those events happen and give a time based on syslog?
>>
>
> Thanks, Gabe. I'm seeing this on any call I originate on the problematic
> instances. Here are the things I've tried, to no avail, in order to improve
> these results:
> - Kernel CONFIG_HZ set to 1000 (Ubuntu 12.04 default on EC2 is 250)
> - Using mod_timerfd
>
>
> Here are the results of a recent test I ran where the recorded file was
> six seconds short after about four minutes.
>
> Command
> ---------------
> freeswitch at internal> originate
> {record_waste_resources=true}sofia/external/sip:music at iptel.org&record(/tmp/record1.wav) default default
> +OK e223c3e0-103d-11e3-aeea-0779ec213503
>
> I used uuid_kill to terminate the call after 00:03:57.
>
> Duration of the resulting file
> ----------------------------------------
> Input #0, wav, from '/tmp/record1.wav':
>   Duration: 00:03:51.84, bitrate: 128 kb/s
>     Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, 1
> channels, s16, 128 kb/s
>
> FreeSWITCH Console Output
> -------------------------------------------
> 2013-08-28 23:59:32.417082 [DEBUG] switch_ivr_originate.c:2060 Parsing
> global variables
> 2013-08-28 23:59:32.417082 [DEBUG] switch_event.c:1617 Parsing variable
> [record_waste_resources]=[true]
> 2013-08-28 23:59:32.417082 [NOTICE] switch_channel.c:1030 New Channel
> sofia/external/sip:music at iptel.org [e223c3e0-103d-11e3-aeea-0779ec213503]
> 2013-08-28 23:59:32.417082 [DEBUG] mod_sofia.c:5189 (sofia/external/
> sip:music at iptel.org) State Change CS_NEW -> CS_INIT
> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_session.c:1334 Send signal
> sofia/external/sip:music at iptel.org [BREAK]
> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:415
> (sofia/external/sip:music at iptel.org) Running State Change CS_INIT
> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:454
> (sofia/external/sip:music at iptel.org) State INIT
> 2013-08-28 23:59:32.417082 [DEBUG] mod_sofia.c:87 sofia/external/
> sip:music at iptel.org SOFIA INIT
> 2013-08-28 23:59:32.417082 [DEBUG] sofia_glue.c:2677 Local SDP:
> v=0
> o=FreeSWITCH 1377707312 1377707313 IN IP4 10.151.108.236
> s=FreeSWITCH
> c=IN IP4 10.151.108.236
> t=0 0
> m=audio 27060 RTP/AVP 0 8 101 13
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=sendrecv
>
> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_session.c:999 Send signal
> sofia/external/sip:music at iptel.org [BREAK]
> 2013-08-28 23:59:32.417082 [DEBUG] mod_sofia.c:127 (sofia/external/
> sip:music at iptel.org) State Change CS_INIT -> CS_ROUTING
> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_session.c:1334 Send signal
> sofia/external/sip:music at iptel.org [BREAK]
> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:454
> (sofia/external/sip:music at iptel.org) State INIT going to sleep
> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:415
> (sofia/external/sip:music at iptel.org) Running State Change CS_ROUTING
> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:470
> (sofia/external/sip:music at iptel.org) State ROUTING
> 2013-08-28 23:59:32.417082 [DEBUG] mod_sofia.c:150 sofia/external/
> sip:music at iptel.org SOFIA ROUTING
> 2013-08-28 23:59:32.417082 [DEBUG] switch_ivr_originate.c:67
> (sofia/external/sip:music at iptel.org) State Change CS_ROUTING ->
> CS_CONSUME_MEDIA
> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_session.c:1334 Send signal
> sofia/external/sip:music at iptel.org [BREAK]
> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:470
> (sofia/external/sip:music at iptel.org) State ROUTING going to sleep
> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:415
> (sofia/external/sip:music at iptel.org) Running State Change CS_CONSUME_MEDIA
> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:489
> (sofia/external/sip:music at iptel.org) State CONSUME_MEDIA
> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:489
> (sofia/external/sip:music at iptel.org) State CONSUME_MEDIA going to sleep
> 2013-08-28 23:59:32.417082 [DEBUG] sofia.c:5720 Channel sofia/external/
> sip:music at iptel.org entering state [calling][0]
> 2013-08-28 23:59:32.917126 [DEBUG] switch_core_session.c:999 Send signal
> sofia/external/sip:music at iptel.org [BREAK]
> 2013-08-28 23:59:32.917126 [DEBUG] switch_core_session.c:999 Send signal
> sofia/external/sip:music at iptel.org [BREAK]
> 2013-08-28 23:59:32.917126 [DEBUG] sofia.c:5720 Channel sofia/external/
> sip:music at iptel.org entering state [completing][200]
> 2013-08-28 23:59:32.917126 [DEBUG] sofia.c:5733 Remote SDP:
> v=0
> o=- 308297691 1827516553 IN IP4 217.9.36.145
> s=-
> c=IN IP4 217.9.36.145
> t=0 0
> m=audio 37892 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=direction:both
>
> 2013-08-28 23:59:32.917126 [DEBUG] switch_core_session.c:999 Send signal
> sofia/external/sip:music at iptel.org [BREAK]
> 2013-08-28 23:59:32.917126 [DEBUG] switch_core_session.c:999 Send signal
> sofia/external/sip:music at iptel.org [BREAK]
> 2013-08-28 23:59:32.917126 [DEBUG] sofia.c:5720 Channel sofia/external/
> sip:music at iptel.org entering state [ready][200]
> 2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:5181 Audio Codec Compare
> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
> 2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3124 Set Codec
> sofia/external/sip:music at iptel.org PCMU/8000 20 ms 160 samples 64000 bits
> 2013-08-28 23:59:32.917126 [DEBUG] switch_core_codec.c:111 sofia/external/
> sip:music at iptel.org Original read codec set to PCMU:0
> 2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:5303 Set 2833 dtmf send
> payload to 101
> 2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3383 AUDIO RTP
> [sofia/external/sip:music at iptel.org] 10.151.108.236 port 27060 ->
> 217.9.36.145 port 37892 codec: 0 ms: 20
> 2013-08-28 23:59:32.917126 [DEBUG] switch_rtp.c:1985 Starting timer [soft]
> 160 bytes per 20ms
> 2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3647 Set 2833 dtmf send
> payload to 101
> 2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3653 Set 2833 dtmf receive
> payload to 101
> 2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3680 sofia/external/
> sip:music at iptel.org Set rtp dtmf delay to 40
> 2013-08-28 23:59:32.917126 [NOTICE] sofia.c:6476 Channel [sofia/external/
> sip:music at iptel.org] has been answered
> 2013-08-28 23:59:32.917126 [DEBUG] switch_channel.c:3576 (sofia/external/
> sip:music at iptel.org) Callstate Change DOWN -> ACTIVE
> 2013-08-28 23:59:32.937082 [DEBUG] switch_ivr_originate.c:3494 Originate
> Resulted in Success: [sofia/external/sip:music at iptel.org]
> 2013-08-28 23:59:32.937082 [INFO] switch_channel.c:2973 sofia/external/
> sip:music at iptel.org Flipping CID from "" <0000000000> to "Outbound Call"
> <music>
> 2013-08-28 23:59:32.937082 [DEBUG] mod_commands.c:4088 (sofia/external/
> sip:music at iptel.org) State Change CS_CONSUME_MEDIA -> CS_EXECUTE
> 2013-08-28 23:59:32.937082 [DEBUG] switch_core_session.c:1334 Send signal
> sofia/external/sip:music at iptel.org [BREAK]
> 2013-08-28 23:59:32.937082 [DEBUG] switch_core_state_machine.c:415
> (sofia/external/sip:music at iptel.org) Running State Change CS_EXECUTE
> 2013-08-28 23:59:32.937082 [DEBUG] switch_core_state_machine.c:477
> (sofia/external/sip:music at iptel.org) State EXECUTE
> 2013-08-28 23:59:32.937082 [DEBUG] mod_sofia.c:243 sofia/external/
> sip:music at iptel.org SOFIA EXECUTE
> 2013-08-28 23:59:32.937082 [DEBUG] switch_core_state_machine.c:209
> sofia/external/sip:music at iptel.org Standard EXECUTE
> EXECUTE sofia/external/sip:music at iptel.org record(/tmp/record1.wav)
> 2013-08-28 23:59:32.937082 [DEBUG] switch_ivr_play_say.c:475 Raw Codec
> Activated, ready to waste resources!
> 2013-08-28 23:59:32.937082 [DEBUG] switch_ivr_play_say.c:599 Raw Codec
> Activated
> 2013-08-28 23:59:32.937082 [DEBUG] switch_core_codec.c:219 sofia/external/
> sip:music at iptel.org Push codec L16:70
> 2013-08-28 23:59:33.077079 [DEBUG] switch_rtp.c:3706 Correct ip/port
> confirmed.
> 2013-08-29 00:03:29.537080 [NOTICE] switch_ivr.c:3675 Hangup
> sofia/external/sip:music at iptel.org [CS_EXECUTE] [NORMAL_CLEARING]
> 2013-08-29 00:03:29.537080 [DEBUG] switch_channel.c:3130 Send signal
> sofia/external/sip:music at iptel.org [KILL]
> 2013-08-29 00:03:29.537080 [DEBUG] switch_core_session.c:1334 Send signal
> sofia/external/sip:music at iptel.org [BREAK]
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_codec.c:244 sofia/external/
> sip:music at iptel.org Restore previous codec PCMU:0.
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_session.c:2740
> sofia/external/sip:music at iptel.org skip receive message
> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:477
> (sofia/external/sip:music at iptel.org) State EXECUTE going to sleep
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:415
> (sofia/external/sip:music at iptel.org) Running State Change CS_HANGUP
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:678
> (sofia/external/sip:music at iptel.org) State HANGUP
> 2013-08-29 00:03:29.557080 [DEBUG] mod_sofia.c:504 Channel sofia/external/
> sip:music at iptel.org hanging up, cause: NORMAL_CLEARING
> 2013-08-29 00:03:29.557080 [DEBUG] mod_sofia.c:556 Sending BYE to
> sofia/external/sip:music at iptel.org
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:48
> sofia/external/sip:music at iptel.org Standard HANGUP, cause: NORMAL_CLEARING
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:678
> (sofia/external/sip:music at iptel.org) State HANGUP going to sleep
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:689
> (sofia/external/sip:music at iptel.org) Callstate Change ACTIVE -> HANGUP
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:446
> (sofia/external/sip:music at iptel.org) State Change CS_HANGUP ->
> CS_REPORTING
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_session.c:1334 Send signal
> sofia/external/sip:music at iptel.org [BREAK]
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:415
> (sofia/external/sip:music at iptel.org) Running State Change CS_REPORTING
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:761
> (sofia/external/sip:music at iptel.org) State REPORTING
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:92
> sofia/external/sip:music at iptel.org Standard REPORTING, cause:
> NORMAL_CLEARING
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:761
> (sofia/external/sip:music at iptel.org) State REPORTING going to sleep
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:440
> (sofia/external/sip:music at iptel.org) State Change CS_REPORTING ->
> CS_DESTROY
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_session.c:1334 Send signal
> sofia/external/sip:music at iptel.org [BREAK]
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_session.c:1542 Session 6
> (sofia/external/sip:music at iptel.org) Locked, Waiting on external entities
> 2013-08-29 00:03:29.557080 [NOTICE] switch_core_session.c:1560 Session 6
> (sofia/external/sip:music at iptel.org) Ended
> 2013-08-29 00:03:29.557080 [NOTICE] switch_core_session.c:1564 Close
> Channel sofia/external/sip:music at iptel.org [CS_DESTROY]
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:565
> (sofia/external/sip:music at iptel.org) Callstate Change HANGUP -> DOWN
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:568
> (sofia/external/sip:music at iptel.org) Running State Change CS_DESTROY
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:578
> (sofia/external/sip:music at iptel.org) State DESTROY
> 2013-08-29 00:03:29.557080 [DEBUG] mod_sofia.c:397 sofia/external/
> sip:music at iptel.org SOFIA DESTROY
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:99
> sofia/external/sip:music at iptel.org Standard DESTROY
> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:578
> (sofia/external/sip:music at iptel.org) State DESTROY going to sleep
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jleung at v10networks.ca  Thu Aug 29 09:51:14 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Wed, 28 Aug 2013 22:51:14 -0700
Subject: [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd --
	preferred	approach
In-Reply-To: <CAHDL-szR+_NFAQOmbaT-pc1jfkBWXBmSZLKS9SNU=37XxJF6fw@mail.gmail.com>
References: <CAHDL-szR+_NFAQOmbaT-pc1jfkBWXBmSZLKS9SNU=37XxJF6fw@mail.gmail.com>
Message-ID: <A70AC0C1B7E7FD478A6BF19705461AA0C8DA@IMBER.smb.curriegrad2004.ca>

If I can recall correctly, timerfd will be used automatically regardless
in recent versions of FreeSWITCH if it's available.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Jayanth Acharya
Sent: Wednesday, August 28, 2013 9:35 PM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd -- preferred
approach

 

Assuming that one has full control over the kernel configuration and FS
build / config options, is there a preferred mechanism between the 2 for
letting FS have access to high resolution timers, i.e.

*	CONFIG_HZ=1000
*	mod_timerfd

This was one step I missed in my FS setup, and I think it might explain
the extremely choppy, broken, garbled announcement I hear when I call
into my gsmopen EP via phone.

J

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From jayachar88 at gmail.com  Thu Aug 29 11:00:06 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Thu, 29 Aug 2013 12:30:06 +0530
Subject: [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd -- preferred
	approach
In-Reply-To: <A70AC0C1B7E7FD478A6BF19705461AA0C8DA@IMBER.smb.curriegrad2004.ca>
References: <CAHDL-szR+_NFAQOmbaT-pc1jfkBWXBmSZLKS9SNU=37XxJF6fw@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C8DA@IMBER.smb.curriegrad2004.ca>
Message-ID: <CAHDL-sza2_KoL8OWswrYccbLEJXKXzpqGaYrza27vrE3LEepPQ@mail.gmail.com>

Thanks Jeff. Reading the mod_timerfd wiki page, I see that there are 2
approaches to using timerfd,

1. Using the loadable module i.e. mod_timerfd (requiring me to build this
module and then ensure it is loaded)
2. Using the support of timerfd now available in FS core itself (requiring
me to just enable it in the switch.conf file)

By default neither of those 2 seem to be true. I am on what as the Git HEAD
last week - 22nd Aug 2013, so pretty recent source. However, it when I
adopted the 2nd approach, i.e. enable timerfd use in FS core, by enabling
the config in switch.conf, it didn't have the desired effect. Perhaps
indicating that my problems might not be limited to just the timer
resolution !!

Using the method described here:
http://www.advenage.com/topics/linux-timer-interrupt-frequency.php

I tested and found that on my FS machine, the timer resolution available is
already pretty good -- in the ~3KHz range, so shouldn't be an issue, if FS
core is indeed correctly using timerfd.

---
kernel timer interrupt frequency is approx. 3401 Hz
jayachar at dabbian1:~$ ./timer_resolution_test
kernel timer interrupt frequency is approx. 3401 Hz
jayachar at dabbian1:~$ ./timer_resolution_test
kernel timer interrupt frequency is approx. 3448 Hz
jayachar at dabbian1:~$ ./timer_resolution_test
kernel timer interrupt frequency is approx. 3021 Hz
jayachar at dabbian1:~$ ./timer_resolution_test
kernel timer interrupt frequency is approx. 3067 Hz
---

I have already check that my kernel has the TIMERFD option enabled in the
build, which explains (perhaps) the ~3KHz resolution I see.

So the question now is, is FS really using the timerfd logic in it's core
properly, or do I need to try the mod_timerfd method as well !! (That's a
rhetorical question... I will go ahead and try anyway, disabling the FS
core option to use timerfd, as per the Wiki).



On Thu, Aug 29, 2013 at 11:21 AM, Jeff Leung <jleung at v10networks.ca> wrote:

> If I can recall correctly, timerfd will be used automatically regardless
> in recent versions of FreeSWITCH if it?s available.****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
> Acharya
> *Sent:* Wednesday, August 28, 2013 9:35 PM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd -- preferred
> approach****
>
> ** **
>
> Assuming that one has full control over the kernel configuration and FS
> build / config options, is there a preferred mechanism between the 2 for
> letting FS have access to high resolution timers, i.e.****
>
>    - CONFIG_HZ=1000****
>    - mod_timerfd****
>
> This was one step I missed in my FS setup, and I think it might explain
> the extremely choppy, broken, garbled announcement I hear when I call into
> my gsmopen EP via phone.****
>
> J****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Thu Aug 29 11:01:30 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Thu, 29 Aug 2013 12:31:30 +0530
Subject: [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd -- preferred
	approach
In-Reply-To: <CAHDL-sza2_KoL8OWswrYccbLEJXKXzpqGaYrza27vrE3LEepPQ@mail.gmail.com>
References: <CAHDL-szR+_NFAQOmbaT-pc1jfkBWXBmSZLKS9SNU=37XxJF6fw@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C8DA@IMBER.smb.curriegrad2004.ca>
	<CAHDL-sza2_KoL8OWswrYccbLEJXKXzpqGaYrza27vrE3LEepPQ@mail.gmail.com>
Message-ID: <CAHDL-sx_PRYRBokT+B-cvrHnaxsjo84KsiQwMJkd2U=seLn1tw@mail.gmail.com>

I didn't mean to say: "... By default neither of those 2 seem to be
true...."
but rather: "...By default neither of those 2 are enabled..."


On Thu, Aug 29, 2013 at 12:30 PM, Jayanth Acharya <jayachar88 at gmail.com>wrote:

> Thanks Jeff. Reading the mod_timerfd wiki page, I see that there are 2
> approaches to using timerfd,
>
> 1. Using the loadable module i.e. mod_timerfd (requiring me to build this
> module and then ensure it is loaded)
> 2. Using the support of timerfd now available in FS core itself (requiring
> me to just enable it in the switch.conf file)
>
> By default neither of those 2 seem to be true. I am on what as the Git
> HEAD last week - 22nd Aug 2013, so pretty recent source. However, it when I
> adopted the 2nd approach, i.e. enable timerfd use in FS core, by enabling
> the config in switch.conf, it didn't have the desired effect. Perhaps
> indicating that my problems might not be limited to just the timer
> resolution !!
>
> Using the method described here:
> http://www.advenage.com/topics/linux-timer-interrupt-frequency.php
>
> I tested and found that on my FS machine, the timer resolution available
> is already pretty good -- in the ~3KHz range, so shouldn't be an issue, if
> FS core is indeed correctly using timerfd.
>
> ---
> kernel timer interrupt frequency is approx. 3401 Hz
> jayachar at dabbian1:~$ ./timer_resolution_test
> kernel timer interrupt frequency is approx. 3401 Hz
> jayachar at dabbian1:~$ ./timer_resolution_test
> kernel timer interrupt frequency is approx. 3448 Hz
> jayachar at dabbian1:~$ ./timer_resolution_test
> kernel timer interrupt frequency is approx. 3021 Hz
> jayachar at dabbian1:~$ ./timer_resolution_test
> kernel timer interrupt frequency is approx. 3067 Hz
> ---
>
> I have already check that my kernel has the TIMERFD option enabled in the
> build, which explains (perhaps) the ~3KHz resolution I see.
>
> So the question now is, is FS really using the timerfd logic in it's core
> properly, or do I need to try the mod_timerfd method as well !! (That's a
> rhetorical question... I will go ahead and try anyway, disabling the FS
> core option to use timerfd, as per the Wiki).
>
>
>
> On Thu, Aug 29, 2013 at 11:21 AM, Jeff Leung <jleung at v10networks.ca>wrote:
>
>> If I can recall correctly, timerfd will be used automatically regardless
>> in recent versions of FreeSWITCH if it?s available.****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
>> Acharya
>> *Sent:* Wednesday, August 28, 2013 9:35 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd -- preferred
>> approach****
>>
>> ** **
>>
>> Assuming that one has full control over the kernel configuration and FS
>> build / config options, is there a preferred mechanism between the 2 for
>> letting FS have access to high resolution timers, i.e.****
>>
>>    - CONFIG_HZ=1000****
>>    - mod_timerfd****
>>
>> This was one step I missed in my FS setup, and I think it might explain
>> the extremely choppy, broken, garbled announcement I hear when I call into
>> my gsmopen EP via phone.****
>>
>> J****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 juanito1982 at gmail.com  Thu Aug 29 11:11:52 2013
From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=)
Date: Thu, 29 Aug 2013 09:11:52 +0200
Subject: [Freeswitch-users] sip-force-contact problem in FS 1.2.12
Message-ID: <CAHLYDCummijAhLmOnSytrf3tp2cpCfRFpdTAK5U6nZAYH1k+EQ@mail.gmail.com>

Hello,

   Do you know if there are any changes in the way of manage
sip-force-contact var in version FS 1.2.12? I have one NAT test endpoint
which registers as 'Registered(UDP-NAT)' in version 1.2.12 leaving contact
IP intact while registers as ''Registered(AUTO-NAT-2.0)' in version 1.2.10
forcing contact IP to its public IP. Tests done within the same machine and
same configuration.

Regards
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From peter at olssononline.se  Thu Aug 29 11:58:12 2013
From: peter at olssononline.se (Peter Olsson)
Date: Thu, 29 Aug 2013 09:58:12 +0200
Subject: [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd -- preferred
	approach
In-Reply-To: <CAHDL-sza2_KoL8OWswrYccbLEJXKXzpqGaYrza27vrE3LEepPQ@mail.gmail.com>
References: <CAHDL-szR+_NFAQOmbaT-pc1jfkBWXBmSZLKS9SNU=37XxJF6fw@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C8DA@IMBER.smb.curriegrad2004.ca>
	<CAHDL-sza2_KoL8OWswrYccbLEJXKXzpqGaYrza27vrE3LEepPQ@mail.gmail.com>
Message-ID: <CAPEdYTj=54WhSGSS-=2U5_bGQm1=opd=VDuY1J_tecauJ6OyKg@mail.gmail.com>

I believe that timerfd is used automatically in the core if it detects a
system supporting it. However, I don't know how to really be sure.. :) One
thing though, if you just startup FS, if it doesn't try to do clock
calibration during startup, I think it is using timerfd stuff, so that
might be a way to "know"...

Also, when running FS, you can try timer_test command (inside FS console)
and check out the results in there.

/Peter


2013/8/29 Jayanth Acharya <jayachar88 at gmail.com>

> Thanks Jeff. Reading the mod_timerfd wiki page, I see that there are 2
> approaches to using timerfd,
>
> 1. Using the loadable module i.e. mod_timerfd (requiring me to build this
> module and then ensure it is loaded)
> 2. Using the support of timerfd now available in FS core itself (requiring
> me to just enable it in the switch.conf file)
>
> By default neither of those 2 seem to be true. I am on what as the Git
> HEAD last week - 22nd Aug 2013, so pretty recent source. However, it when I
> adopted the 2nd approach, i.e. enable timerfd use in FS core, by enabling
> the config in switch.conf, it didn't have the desired effect. Perhaps
> indicating that my problems might not be limited to just the timer
> resolution !!
>
> Using the method described here:
> http://www.advenage.com/topics/linux-timer-interrupt-frequency.php
>
> I tested and found that on my FS machine, the timer resolution available
> is already pretty good -- in the ~3KHz range, so shouldn't be an issue, if
> FS core is indeed correctly using timerfd.
>
> ---
> kernel timer interrupt frequency is approx. 3401 Hz
> jayachar at dabbian1:~$ ./timer_resolution_test
> kernel timer interrupt frequency is approx. 3401 Hz
> jayachar at dabbian1:~$ ./timer_resolution_test
> kernel timer interrupt frequency is approx. 3448 Hz
> jayachar at dabbian1:~$ ./timer_resolution_test
> kernel timer interrupt frequency is approx. 3021 Hz
> jayachar at dabbian1:~$ ./timer_resolution_test
> kernel timer interrupt frequency is approx. 3067 Hz
> ---
>
> I have already check that my kernel has the TIMERFD option enabled in the
> build, which explains (perhaps) the ~3KHz resolution I see.
>
> So the question now is, is FS really using the timerfd logic in it's core
> properly, or do I need to try the mod_timerfd method as well !! (That's a
> rhetorical question... I will go ahead and try anyway, disabling the FS
> core option to use timerfd, as per the Wiki).
>
>
>
> On Thu, Aug 29, 2013 at 11:21 AM, Jeff Leung <jleung at v10networks.ca>wrote:
>
>> If I can recall correctly, timerfd will be used automatically regardless
>> in recent versions of FreeSWITCH if it?s available.****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
>> Acharya
>> *Sent:* Wednesday, August 28, 2013 9:35 PM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd -- preferred
>> approach****
>>
>> ** **
>>
>> Assuming that one has full control over the kernel configuration and FS
>> build / config options, is there a preferred mechanism between the 2 for
>> letting FS have access to high resolution timers, i.e.****
>>
>>    - CONFIG_HZ=1000****
>>    - mod_timerfd****
>>
>> This was one step I missed in my FS setup, and I think it might explain
>> the extremely choppy, broken, garbled announcement I hear when I call into
>> my gsmopen EP via phone.****
>>
>> J****
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 tahir at ictinnovations.com  Thu Aug 29 12:13:07 2013
From: tahir at ictinnovations.com (Tahir Almas)
Date: Thu, 29 Aug 2013 13:13:07 +0500
Subject: [Freeswitch-users] Recording Duration on EC2
In-Reply-To: <CAHDL-syMhd=CX+tE0LCn=uyrQ_6uxg+m0wAW07n5qPhsLUxHkg@mail.gmail.com>
References: <CAC5vAOQFMj4Oykxr-Adymc8HrzDjxKtcEx5bvoYJV24CE4uzwQ@mail.gmail.com>
	<CAMwZ-t+qDQ_t0iLzWiCNR+dVkKfa-MrM6QjOTiVrLqNNzKNetw@mail.gmail.com>
	<CAC5vAORqR_0a3RXW8RLiPXoQZUQxXkieGtwjFMRMA4rpY2h4Xw@mail.gmail.com>
	<CAHDL-syMhd=CX+tE0LCn=uyrQ_6uxg+m0wAW07n5qPhsLUxHkg@mail.gmail.com>
Message-ID: <CAOL0t+KwU_Pa-LVzEE9a4LXTRXfFN09viF9zRDjX9cazu6Eegg@mail.gmail.com>

interesting findings so far !

it resemble to travelling in black-hole

I came across another research work that show time get slow  around heavy
objects like travelling near  egypt's pyramids

*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT



On Thu, Aug 29, 2013 at 9:44 AM, Jayanth Acharya <jayachar88 at gmail.com>wrote:

> Being a complete n00b with FS, I could be completely off the mark here,
> but recently when I was searching around for my extremely poor audio
> quality issues with mod_gsmopen, I found reference to timing issues, and
> then I saw this thread. Coming from an embedded background, I believe I can
> extend the logic that if you run too many instances with high-resolution
> software timers on same processor, the resolution of all instances gets
> impacted, since nobody would be getting "enough time" (or interrupts, fast
> enough)... this could be a clue to chase, if your problem is unique to a
> particular EC2 instance (however, I see follow-on posts indicating this to
> be a more generic issue).
>
> Here are 2 JIRA's that have interesting content:
>
> http://jira.freeswitch.org/browse/FS-4256
> http://jira.freeswitch.org/browse/FS-3290 (this one has a solution that worked for someone)
>
>
>
>
> On Thu, Aug 29, 2013 at 5:41 AM, Byron Clark <byron at theclarkfamily.name>wrote:
>
>> On Wed, Aug 28, 2013 at 3:00 PM, Gabriel Gunderson <gabe at gundy.org>wrote:
>>
>>> On Mon, Aug 26, 2013 at 11:36 AM, Byron Clark <byron at theclarkfamily.name>
>>> wrote:
>>> > I'm using FreeSWITCH on EC2 to call into a conference bridge and
>>> record the
>>> > audio from the conference. It works well except for one thing: the
>>> recording
>>> > contains all the audio from the call, but the duration of the file is
>>> > shorter than the call. The duration is typically 3-5 seconds shorter
>>> than
>>> > the call for each 5 minutes of call. On a recent test, wall clock
>>> showed a
>>> > duration of 15:32.93 but the duration of the recording file was
>>> 15:19.96.
>>> > The ugly part is that this only happens on EC2 instances where there
>>> is some
>>> > CPU steal time (<5%) occurring.
>>>
>>> Byron, do you see this in a regular bridge too? Or, is it only in a
>>> conf. room? I know it would be hard to get, but do you know how long
>>> the *actual* call was? Can you get back into the logs and see when
>>> those events happen and give a time based on syslog?
>>>
>>
>> Thanks, Gabe. I'm seeing this on any call I originate on the problematic
>> instances. Here are the things I've tried, to no avail, in order to improve
>> these results:
>> - Kernel CONFIG_HZ set to 1000 (Ubuntu 12.04 default on EC2 is 250)
>> - Using mod_timerfd
>>
>>
>> Here are the results of a recent test I ran where the recorded file was
>> six seconds short after about four minutes.
>>
>> Command
>> ---------------
>> freeswitch at internal> originate
>> {record_waste_resources=true}sofia/external/sip:music at iptel.org&record(/tmp/record1.wav) default default
>> +OK e223c3e0-103d-11e3-aeea-0779ec213503
>>
>> I used uuid_kill to terminate the call after 00:03:57.
>>
>> Duration of the resulting file
>> ----------------------------------------
>> Input #0, wav, from '/tmp/record1.wav':
>>   Duration: 00:03:51.84, bitrate: 128 kb/s
>>     Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, 1
>> channels, s16, 128 kb/s
>>
>> FreeSWITCH Console Output
>> -------------------------------------------
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_ivr_originate.c:2060 Parsing
>> global variables
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_event.c:1617 Parsing variable
>> [record_waste_resources]=[true]
>> 2013-08-28 23:59:32.417082 [NOTICE] switch_channel.c:1030 New Channel
>> sofia/external/sip:music at iptel.org [e223c3e0-103d-11e3-aeea-0779ec213503]
>> 2013-08-28 23:59:32.417082 [DEBUG] mod_sofia.c:5189 (sofia/external/
>> sip:music at iptel.org) State Change CS_NEW -> CS_INIT
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_session.c:1334 Send signal
>> sofia/external/sip:music at iptel.org [BREAK]
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:415
>> (sofia/external/sip:music at iptel.org) Running State Change CS_INIT
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:454
>> (sofia/external/sip:music at iptel.org) State INIT
>> 2013-08-28 23:59:32.417082 [DEBUG] mod_sofia.c:87 sofia/external/
>> sip:music at iptel.org SOFIA INIT
>> 2013-08-28 23:59:32.417082 [DEBUG] sofia_glue.c:2677 Local SDP:
>> v=0
>> o=FreeSWITCH 1377707312 1377707313 IN IP4 10.151.108.236
>> s=FreeSWITCH
>> c=IN IP4 10.151.108.236
>> t=0 0
>> m=audio 27060 RTP/AVP 0 8 101 13
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>> a=sendrecv
>>
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_session.c:999 Send signal
>> sofia/external/sip:music at iptel.org [BREAK]
>> 2013-08-28 23:59:32.417082 [DEBUG] mod_sofia.c:127 (sofia/external/
>> sip:music at iptel.org) State Change CS_INIT -> CS_ROUTING
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_session.c:1334 Send signal
>> sofia/external/sip:music at iptel.org [BREAK]
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:454
>> (sofia/external/sip:music at iptel.org) State INIT going to sleep
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:415
>> (sofia/external/sip:music at iptel.org) Running State Change CS_ROUTING
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:470
>> (sofia/external/sip:music at iptel.org) State ROUTING
>> 2013-08-28 23:59:32.417082 [DEBUG] mod_sofia.c:150 sofia/external/
>> sip:music at iptel.org SOFIA ROUTING
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_ivr_originate.c:67
>> (sofia/external/sip:music at iptel.org) State Change CS_ROUTING ->
>> CS_CONSUME_MEDIA
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_session.c:1334 Send signal
>> sofia/external/sip:music at iptel.org [BREAK]
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:470
>> (sofia/external/sip:music at iptel.org) State ROUTING going to sleep
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:415
>> (sofia/external/sip:music at iptel.org) Running State Change
>> CS_CONSUME_MEDIA
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:489
>> (sofia/external/sip:music at iptel.org) State CONSUME_MEDIA
>> 2013-08-28 23:59:32.417082 [DEBUG] switch_core_state_machine.c:489
>> (sofia/external/sip:music at iptel.org) State CONSUME_MEDIA going to sleep
>> 2013-08-28 23:59:32.417082 [DEBUG] sofia.c:5720 Channel sofia/external/
>> sip:music at iptel.org entering state [calling][0]
>> 2013-08-28 23:59:32.917126 [DEBUG] switch_core_session.c:999 Send signal
>> sofia/external/sip:music at iptel.org [BREAK]
>> 2013-08-28 23:59:32.917126 [DEBUG] switch_core_session.c:999 Send signal
>> sofia/external/sip:music at iptel.org [BREAK]
>> 2013-08-28 23:59:32.917126 [DEBUG] sofia.c:5720 Channel sofia/external/
>> sip:music at iptel.org entering state [completing][200]
>> 2013-08-28 23:59:32.917126 [DEBUG] sofia.c:5733 Remote SDP:
>> v=0
>> o=- 308297691 1827516553 IN IP4 217.9.36.145
>> s=-
>> c=IN IP4 217.9.36.145
>> t=0 0
>> m=audio 37892 RTP/AVP 0 8 101
>> a=rtpmap:0 PCMU/8000
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-15
>> a=direction:both
>>
>> 2013-08-28 23:59:32.917126 [DEBUG] switch_core_session.c:999 Send signal
>> sofia/external/sip:music at iptel.org [BREAK]
>> 2013-08-28 23:59:32.917126 [DEBUG] switch_core_session.c:999 Send signal
>> sofia/external/sip:music at iptel.org [BREAK]
>> 2013-08-28 23:59:32.917126 [DEBUG] sofia.c:5720 Channel sofia/external/
>> sip:music at iptel.org entering state [ready][200]
>> 2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:5181 Audio Codec Compare
>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000]
>> 2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3124 Set Codec
>> sofia/external/sip:music at iptel.org PCMU/8000 20 ms 160 samples 64000 bits
>> 2013-08-28 23:59:32.917126 [DEBUG] switch_core_codec.c:111 sofia/external/
>> sip:music at iptel.org Original read codec set to PCMU:0
>> 2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:5303 Set 2833 dtmf send
>> payload to 101
>> 2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3383 AUDIO RTP
>> [sofia/external/sip:music at iptel.org] 10.151.108.236 port 27060 ->
>> 217.9.36.145 port 37892 codec: 0 ms: 20
>> 2013-08-28 23:59:32.917126 [DEBUG] switch_rtp.c:1985 Starting timer
>> [soft] 160 bytes per 20ms
>> 2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3647 Set 2833 dtmf send
>> payload to 101
>> 2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3653 Set 2833 dtmf
>> receive payload to 101
>> 2013-08-28 23:59:32.917126 [DEBUG] sofia_glue.c:3680 sofia/external/
>> sip:music at iptel.org Set rtp dtmf delay to 40
>> 2013-08-28 23:59:32.917126 [NOTICE] sofia.c:6476 Channel [sofia/external/
>> sip:music at iptel.org] has been answered
>> 2013-08-28 23:59:32.917126 [DEBUG] switch_channel.c:3576 (sofia/external/
>> sip:music at iptel.org) Callstate Change DOWN -> ACTIVE
>> 2013-08-28 23:59:32.937082 [DEBUG] switch_ivr_originate.c:3494 Originate
>> Resulted in Success: [sofia/external/sip:music at iptel.org]
>> 2013-08-28 23:59:32.937082 [INFO] switch_channel.c:2973 sofia/external/
>> sip:music at iptel.org Flipping CID from "" <0000000000> to "Outbound Call"
>> <music>
>> 2013-08-28 23:59:32.937082 [DEBUG] mod_commands.c:4088 (sofia/external/
>> sip:music at iptel.org) State Change CS_CONSUME_MEDIA -> CS_EXECUTE
>> 2013-08-28 23:59:32.937082 [DEBUG] switch_core_session.c:1334 Send signal
>> sofia/external/sip:music at iptel.org [BREAK]
>> 2013-08-28 23:59:32.937082 [DEBUG] switch_core_state_machine.c:415
>> (sofia/external/sip:music at iptel.org) Running State Change CS_EXECUTE
>> 2013-08-28 23:59:32.937082 [DEBUG] switch_core_state_machine.c:477
>> (sofia/external/sip:music at iptel.org) State EXECUTE
>> 2013-08-28 23:59:32.937082 [DEBUG] mod_sofia.c:243 sofia/external/
>> sip:music at iptel.org SOFIA EXECUTE
>> 2013-08-28 23:59:32.937082 [DEBUG] switch_core_state_machine.c:209
>> sofia/external/sip:music at iptel.org Standard EXECUTE
>> EXECUTE sofia/external/sip:music at iptel.org record(/tmp/record1.wav)
>> 2013-08-28 23:59:32.937082 [DEBUG] switch_ivr_play_say.c:475 Raw Codec
>> Activated, ready to waste resources!
>> 2013-08-28 23:59:32.937082 [DEBUG] switch_ivr_play_say.c:599 Raw Codec
>> Activated
>> 2013-08-28 23:59:32.937082 [DEBUG] switch_core_codec.c:219 sofia/external/
>> sip:music at iptel.org Push codec L16:70
>> 2013-08-28 23:59:33.077079 [DEBUG] switch_rtp.c:3706 Correct ip/port
>> confirmed.
>> 2013-08-29 00:03:29.537080 [NOTICE] switch_ivr.c:3675 Hangup
>> sofia/external/sip:music at iptel.org [CS_EXECUTE] [NORMAL_CLEARING]
>> 2013-08-29 00:03:29.537080 [DEBUG] switch_channel.c:3130 Send signal
>> sofia/external/sip:music at iptel.org [KILL]
>> 2013-08-29 00:03:29.537080 [DEBUG] switch_core_session.c:1334 Send signal
>> sofia/external/sip:music at iptel.org [BREAK]
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_codec.c:244 sofia/external/
>> sip:music at iptel.org Restore previous codec PCMU:0.
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_session.c:2740
>> sofia/external/sip:music at iptel.org skip receive message
>> [APPLICATION_EXEC_COMPLETE] (channel is hungup already)
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:477
>> (sofia/external/sip:music at iptel.org) State EXECUTE going to sleep
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:415
>> (sofia/external/sip:music at iptel.org) Running State Change CS_HANGUP
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:678
>> (sofia/external/sip:music at iptel.org) State HANGUP
>> 2013-08-29 00:03:29.557080 [DEBUG] mod_sofia.c:504 Channel sofia/external/
>> sip:music at iptel.org hanging up, cause: NORMAL_CLEARING
>> 2013-08-29 00:03:29.557080 [DEBUG] mod_sofia.c:556 Sending BYE to
>> sofia/external/sip:music at iptel.org
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:48
>> sofia/external/sip:music at iptel.org Standard HANGUP, cause:
>> NORMAL_CLEARING
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:678
>> (sofia/external/sip:music at iptel.org) State HANGUP going to sleep
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:689
>> (sofia/external/sip:music at iptel.org) Callstate Change ACTIVE -> HANGUP
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:446
>> (sofia/external/sip:music at iptel.org) State Change CS_HANGUP ->
>> CS_REPORTING
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_session.c:1334 Send signal
>> sofia/external/sip:music at iptel.org [BREAK]
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:415
>> (sofia/external/sip:music at iptel.org) Running State Change CS_REPORTING
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:761
>> (sofia/external/sip:music at iptel.org) State REPORTING
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:92
>> sofia/external/sip:music at iptel.org Standard REPORTING, cause:
>> NORMAL_CLEARING
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:761
>> (sofia/external/sip:music at iptel.org) State REPORTING going to sleep
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:440
>> (sofia/external/sip:music at iptel.org) State Change CS_REPORTING ->
>> CS_DESTROY
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_session.c:1334 Send signal
>> sofia/external/sip:music at iptel.org [BREAK]
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_session.c:1542 Session 6
>> (sofia/external/sip:music at iptel.org) Locked, Waiting on external entities
>> 2013-08-29 00:03:29.557080 [NOTICE] switch_core_session.c:1560 Session 6
>> (sofia/external/sip:music at iptel.org) Ended
>> 2013-08-29 00:03:29.557080 [NOTICE] switch_core_session.c:1564 Close
>> Channel sofia/external/sip:music at iptel.org [CS_DESTROY]
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:565
>> (sofia/external/sip:music at iptel.org) Callstate Change HANGUP -> DOWN
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:568
>> (sofia/external/sip:music at iptel.org) Running State Change CS_DESTROY
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:578
>> (sofia/external/sip:music at iptel.org) State DESTROY
>> 2013-08-29 00:03:29.557080 [DEBUG] mod_sofia.c:397 sofia/external/
>> sip:music at iptel.org SOFIA DESTROY
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:99
>> sofia/external/sip:music at iptel.org Standard DESTROY
>> 2013-08-29 00:03:29.557080 [DEBUG] switch_core_state_machine.c:578
>> (sofia/external/sip:music at iptel.org) State DESTROY going to sleep
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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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>
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From jayachar88 at gmail.com  Thu Aug 29 12:42:46 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Thu, 29 Aug 2013 14:12:46 +0530
Subject: [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd -- preferred
	approach
In-Reply-To: <CAPEdYTj=54WhSGSS-=2U5_bGQm1=opd=VDuY1J_tecauJ6OyKg@mail.gmail.com>
References: <CAHDL-szR+_NFAQOmbaT-pc1jfkBWXBmSZLKS9SNU=37XxJF6fw@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C8DA@IMBER.smb.curriegrad2004.ca>
	<CAHDL-sza2_KoL8OWswrYccbLEJXKXzpqGaYrza27vrE3LEepPQ@mail.gmail.com>
	<CAPEdYTj=54WhSGSS-=2U5_bGQm1=opd=VDuY1J_tecauJ6OyKg@mail.gmail.com>
Message-ID: <CAHDL-sxr+QeEzy9DPfxWWwK5iq4k5Svqzmegfg=DdeXc_JUy9w@mail.gmail.com>

On Thu, Aug 29, 2013 at 1:28 PM, Peter Olsson <peter at olssononline.se> wrote:

>
> I believe that timerfd is used automatically in the core if it detects a
> system supporting it. However, I don't know how to really be sure.. :) One
> thing though, if you just startup FS, if it doesn't try to do clock
> calibration during startup, I think it is using timerfd stuff, so that
> might be a way to "know"...
>
> Also, when running FS, you can try timer_test command (inside FS console)
> and check out the results in there.
>

Thanks Peter.

freeswitch at dabbian1> timer_test 10 3
2013-08-29 13:56:24.703028 [CONSOLE] mod_commands.c:827 Timer Test: 1 sleep
10 10019
2013-08-29 13:56:24.712967 [CONSOLE] mod_commands.c:827 Timer Test: 2 sleep
10 9930
2013-08-29 13:56:24.722996 [CONSOLE] mod_commands.c:827 Timer Test: 3 sleep
10 10040

Avg: 9.996ms Total Time: 29.989ms

However,  for the "time_test" command in Mod_commands Wiki page -- I see
that it assumes units as "microseconds" (not milliseconds, which is usually
what we mean by "ms").. or is it a typo ??

*... bad timer jitter is. It runs the test count times (default 10) and
tries to sleep for mss microseconds. It returns the actual timer duration
along with an average.
*

Anyhow, not sure how to interpret if the output of timer_test indicates a
healthy resolution or not.

/Peter
>
>
> 2013/8/29 Jayanth Acharya <jayachar88 at gmail.com>
>
>> Thanks Jeff. Reading the mod_timerfd wiki page, I see that there are 2
>> approaches to using timerfd,
>>
>> 1. Using the loadable module i.e. mod_timerfd (requiring me to build this
>> module and then ensure it is loaded)
>> 2. Using the support of timerfd now available in FS core itself
>> (requiring me to just enable it in the switch.conf file)
>>
>> By default neither of those 2 seem to be true. I am on what as the Git
>> HEAD last week - 22nd Aug 2013, so pretty recent source. However, it when I
>> adopted the 2nd approach, i.e. enable timerfd use in FS core, by enabling
>> the config in switch.conf, it didn't have the desired effect. Perhaps
>> indicating that my problems might not be limited to just the timer
>> resolution !!
>>
>> Using the method described here:
>> http://www.advenage.com/topics/linux-timer-interrupt-frequency.php
>>
>> I tested and found that on my FS machine, the timer resolution available
>> is already pretty good -- in the ~3KHz range, so shouldn't be an issue, if
>> FS core is indeed correctly using timerfd.
>>
>> ---
>> kernel timer interrupt frequency is approx. 3401 Hz
>> jayachar at dabbian1:~$ ./timer_resolution_test
>> kernel timer interrupt frequency is approx. 3401 Hz
>> jayachar at dabbian1:~$ ./timer_resolution_test
>> kernel timer interrupt frequency is approx. 3448 Hz
>> jayachar at dabbian1:~$ ./timer_resolution_test
>> kernel timer interrupt frequency is approx. 3021 Hz
>> jayachar at dabbian1:~$ ./timer_resolution_test
>> kernel timer interrupt frequency is approx. 3067 Hz
>> ---
>>
>> I have already check that my kernel has the TIMERFD option enabled in the
>> build, which explains (perhaps) the ~3KHz resolution I see.
>>
>> So the question now is, is FS really using the timerfd logic in it's core
>> properly, or do I need to try the mod_timerfd method as well !! (That's a
>> rhetorical question... I will go ahead and try anyway, disabling the FS
>> core option to use timerfd, as per the Wiki).
>>
>>
>>
>> On Thu, Aug 29, 2013 at 11:21 AM, Jeff Leung <jleung at v10networks.ca>wrote:
>>
>>> If I can recall correctly, timerfd will be used automatically regardless
>>> in recent versions of FreeSWITCH if it?s available.****
>>>
>>> ** **
>>>
>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
>>> Acharya
>>> *Sent:* Wednesday, August 28, 2013 9:35 PM
>>> *To:* FreeSWITCH Users Help
>>> *Subject:* [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd --
>>> preferred approach****
>>>
>>> ** **
>>>
>>> Assuming that one has full control over the kernel configuration and FS
>>> build / config options, is there a preferred mechanism between the 2 for
>>> letting FS have access to high resolution timers, i.e.****
>>>
>>>    - CONFIG_HZ=1000****
>>>    - mod_timerfd****
>>>
>>> This was one step I missed in my FS setup, and I think it might explain
>>> the extremely choppy, broken, garbled announcement I hear when I call into
>>> my gsmopen EP via phone.****
>>>
>>> J****
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 peter at olssononline.se  Thu Aug 29 13:06:33 2013
From: peter at olssononline.se (Peter Olsson)
Date: Thu, 29 Aug 2013 11:06:33 +0200
Subject: [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd -- preferred
	approach
In-Reply-To: <CAHDL-sxr+QeEzy9DPfxWWwK5iq4k5Svqzmegfg=DdeXc_JUy9w@mail.gmail.com>
References: <CAHDL-szR+_NFAQOmbaT-pc1jfkBWXBmSZLKS9SNU=37XxJF6fw@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C8DA@IMBER.smb.curriegrad2004.ca>
	<CAHDL-sza2_KoL8OWswrYccbLEJXKXzpqGaYrza27vrE3LEepPQ@mail.gmail.com>
	<CAPEdYTj=54WhSGSS-=2U5_bGQm1=opd=VDuY1J_tecauJ6OyKg@mail.gmail.com>
	<CAHDL-sxr+QeEzy9DPfxWWwK5iq4k5Svqzmegfg=DdeXc_JUy9w@mail.gmail.com>
Message-ID: <CAPEdYTgyra0E+G-VAWeQV0+muBuKUM5hNPk2-WOrCs0dqjWFgg@mail.gmail.com>

Yes, it might be a typo. IIRC it should be usec for time_test. The small
sample of timer_test you provided seems ok to me, +/- 40 usec is probably
good enough.

I think you're good to go :)

/Peter

2013/8/29 Jayanth Acharya <jayachar88 at gmail.com>

> On Thu, Aug 29, 2013 at 1:28 PM, Peter Olsson <peter at olssononline.se>wrote:
>
>>
>> I believe that timerfd is used automatically in the core if it detects a
>> system supporting it. However, I don't know how to really be sure.. :) One
>> thing though, if you just startup FS, if it doesn't try to do clock
>> calibration during startup, I think it is using timerfd stuff, so that
>> might be a way to "know"...
>>
>> Also, when running FS, you can try timer_test command (inside FS console)
>> and check out the results in there.
>>
>
> Thanks Peter.
>
> freeswitch at dabbian1> timer_test 10 3
> 2013-08-29 13:56:24.703028 [CONSOLE] mod_commands.c:827 Timer Test: 1
> sleep 10 10019
> 2013-08-29 13:56:24.712967 [CONSOLE] mod_commands.c:827 Timer Test: 2
> sleep 10 9930
> 2013-08-29 13:56:24.722996 [CONSOLE] mod_commands.c:827 Timer Test: 3
> sleep 10 10040
>
> Avg: 9.996ms Total Time: 29.989ms
>
> However,  for the "time_test" command in Mod_commands Wiki page -- I see
> that it assumes units as "microseconds" (not milliseconds, which is usually
> what we mean by "ms").. or is it a typo ??
>
> *... bad timer jitter is. It runs the test count times (default 10) and
> tries to sleep for mss microseconds. It returns the actual timer duration
> along with an average.
> *
>
> Anyhow, not sure how to interpret if the output of timer_test indicates a
> healthy resolution or not.
>
> /Peter
>>
>>
>> 2013/8/29 Jayanth Acharya <jayachar88 at gmail.com>
>>
>>> Thanks Jeff. Reading the mod_timerfd wiki page, I see that there are 2
>>> approaches to using timerfd,
>>>
>>> 1. Using the loadable module i.e. mod_timerfd (requiring me to build
>>> this module and then ensure it is loaded)
>>> 2. Using the support of timerfd now available in FS core itself
>>> (requiring me to just enable it in the switch.conf file)
>>>
>>> By default neither of those 2 seem to be true. I am on what as the Git
>>> HEAD last week - 22nd Aug 2013, so pretty recent source. However, it when I
>>> adopted the 2nd approach, i.e. enable timerfd use in FS core, by enabling
>>> the config in switch.conf, it didn't have the desired effect. Perhaps
>>> indicating that my problems might not be limited to just the timer
>>> resolution !!
>>>
>>> Using the method described here:
>>> http://www.advenage.com/topics/linux-timer-interrupt-frequency.php
>>>
>>> I tested and found that on my FS machine, the timer resolution available
>>> is already pretty good -- in the ~3KHz range, so shouldn't be an issue, if
>>> FS core is indeed correctly using timerfd.
>>>
>>> ---
>>> kernel timer interrupt frequency is approx. 3401 Hz
>>> jayachar at dabbian1:~$ ./timer_resolution_test
>>> kernel timer interrupt frequency is approx. 3401 Hz
>>> jayachar at dabbian1:~$ ./timer_resolution_test
>>> kernel timer interrupt frequency is approx. 3448 Hz
>>> jayachar at dabbian1:~$ ./timer_resolution_test
>>> kernel timer interrupt frequency is approx. 3021 Hz
>>> jayachar at dabbian1:~$ ./timer_resolution_test
>>> kernel timer interrupt frequency is approx. 3067 Hz
>>> ---
>>>
>>> I have already check that my kernel has the TIMERFD option enabled in
>>> the build, which explains (perhaps) the ~3KHz resolution I see.
>>>
>>> So the question now is, is FS really using the timerfd logic in it's
>>> core properly, or do I need to try the mod_timerfd method as well !!
>>> (That's a rhetorical question... I will go ahead and try anyway, disabling
>>> the FS core option to use timerfd, as per the Wiki).
>>>
>>>
>>>
>>> On Thu, Aug 29, 2013 at 11:21 AM, Jeff Leung <jleung at v10networks.ca>wrote:
>>>
>>>> If I can recall correctly, timerfd will be used automatically
>>>> regardless in recent versions of FreeSWITCH if it?s available.****
>>>>
>>>> ** **
>>>>
>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
>>>> Acharya
>>>> *Sent:* Wednesday, August 28, 2013 9:35 PM
>>>> *To:* FreeSWITCH Users Help
>>>> *Subject:* [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd --
>>>> preferred approach****
>>>>
>>>> ** **
>>>>
>>>> Assuming that one has full control over the kernel configuration and FS
>>>> build / config options, is there a preferred mechanism between the 2 for
>>>> letting FS have access to high resolution timers, i.e.****
>>>>
>>>>    - CONFIG_HZ=1000****
>>>>    - mod_timerfd****
>>>>
>>>> This was one step I missed in my FS setup, and I think it might explain
>>>> the extremely choppy, broken, garbled announcement I hear when I call into
>>>> my gsmopen EP via phone.****
>>>>
>>>> J****
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 jayachar88 at gmail.com  Thu Aug 29 14:18:32 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Thu, 29 Aug 2013 15:48:32 +0530
Subject: [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd -- preferred
	approach
In-Reply-To: <CAPEdYTgyra0E+G-VAWeQV0+muBuKUM5hNPk2-WOrCs0dqjWFgg@mail.gmail.com>
References: <CAHDL-szR+_NFAQOmbaT-pc1jfkBWXBmSZLKS9SNU=37XxJF6fw@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C8DA@IMBER.smb.curriegrad2004.ca>
	<CAHDL-sza2_KoL8OWswrYccbLEJXKXzpqGaYrza27vrE3LEepPQ@mail.gmail.com>
	<CAPEdYTj=54WhSGSS-=2U5_bGQm1=opd=VDuY1J_tecauJ6OyKg@mail.gmail.com>
	<CAHDL-sxr+QeEzy9DPfxWWwK5iq4k5Svqzmegfg=DdeXc_JUy9w@mail.gmail.com>
	<CAPEdYTgyra0E+G-VAWeQV0+muBuKUM5hNPk2-WOrCs0dqjWFgg@mail.gmail.com>
Message-ID: <CAHDL-syLc89-+cLMPbCNfmHbi7nt04=+7dyiHoYjRiWdFkY6CQ@mail.gmail.com>

Thanks again Peter.
So my theory attributing timer issue to the terrible audio quality goes
away. Need to find what else could be the cause !!


On Thu, Aug 29, 2013 at 2:36 PM, Peter Olsson <peter at olssononline.se> wrote:

> Yes, it might be a typo. IIRC it should be usec for time_test. The small
> sample of timer_test you provided seems ok to me, +/- 40 usec is probably
> good enough.
>
> I think you're good to go :)
>
> /Peter
>
> 2013/8/29 Jayanth Acharya <jayachar88 at gmail.com>
>
>> On Thu, Aug 29, 2013 at 1:28 PM, Peter Olsson <peter at olssononline.se>wrote:
>>
>>>
>>> I believe that timerfd is used automatically in the core if it detects a
>>> system supporting it. However, I don't know how to really be sure.. :) One
>>> thing though, if you just startup FS, if it doesn't try to do clock
>>> calibration during startup, I think it is using timerfd stuff, so that
>>> might be a way to "know"...
>>>
>>> Also, when running FS, you can try timer_test command (inside FS
>>> console) and check out the results in there.
>>>
>>
>> Thanks Peter.
>>
>> freeswitch at dabbian1> timer_test 10 3
>> 2013-08-29 13:56:24.703028 [CONSOLE] mod_commands.c:827 Timer Test: 1
>> sleep 10 10019
>> 2013-08-29 13:56:24.712967 [CONSOLE] mod_commands.c:827 Timer Test: 2
>> sleep 10 9930
>> 2013-08-29 13:56:24.722996 [CONSOLE] mod_commands.c:827 Timer Test: 3
>> sleep 10 10040
>>
>> Avg: 9.996ms Total Time: 29.989ms
>>
>> However,  for the "time_test" command in Mod_commands Wiki page -- I see
>> that it assumes units as "microseconds" (not milliseconds, which is usually
>> what we mean by "ms").. or is it a typo ??
>>
>> *... bad timer jitter is. It runs the test count times (default 10) and
>> tries to sleep for mss microseconds. It returns the actual timer
>> duration along with an average.
>> *
>>
>> Anyhow, not sure how to interpret if the output of timer_test indicates a
>> healthy resolution or not.
>>
>> /Peter
>>>
>>>
>>> 2013/8/29 Jayanth Acharya <jayachar88 at gmail.com>
>>>
>>>> Thanks Jeff. Reading the mod_timerfd wiki page, I see that there are 2
>>>> approaches to using timerfd,
>>>>
>>>> 1. Using the loadable module i.e. mod_timerfd (requiring me to build
>>>> this module and then ensure it is loaded)
>>>> 2. Using the support of timerfd now available in FS core itself
>>>> (requiring me to just enable it in the switch.conf file)
>>>>
>>>> By default neither of those 2 seem to be true. I am on what as the Git
>>>> HEAD last week - 22nd Aug 2013, so pretty recent source. However, it when I
>>>> adopted the 2nd approach, i.e. enable timerfd use in FS core, by enabling
>>>> the config in switch.conf, it didn't have the desired effect. Perhaps
>>>> indicating that my problems might not be limited to just the timer
>>>> resolution !!
>>>>
>>>> Using the method described here:
>>>> http://www.advenage.com/topics/linux-timer-interrupt-frequency.php
>>>>
>>>> I tested and found that on my FS machine, the timer resolution
>>>> available is already pretty good -- in the ~3KHz range, so shouldn't be an
>>>> issue, if FS core is indeed correctly using timerfd.
>>>>
>>>> ---
>>>> kernel timer interrupt frequency is approx. 3401 Hz
>>>> jayachar at dabbian1:~$ ./timer_resolution_test
>>>> kernel timer interrupt frequency is approx. 3401 Hz
>>>> jayachar at dabbian1:~$ ./timer_resolution_test
>>>> kernel timer interrupt frequency is approx. 3448 Hz
>>>> jayachar at dabbian1:~$ ./timer_resolution_test
>>>> kernel timer interrupt frequency is approx. 3021 Hz
>>>> jayachar at dabbian1:~$ ./timer_resolution_test
>>>> kernel timer interrupt frequency is approx. 3067 Hz
>>>> ---
>>>>
>>>> I have already check that my kernel has the TIMERFD option enabled in
>>>> the build, which explains (perhaps) the ~3KHz resolution I see.
>>>>
>>>> So the question now is, is FS really using the timerfd logic in it's
>>>> core properly, or do I need to try the mod_timerfd method as well !!
>>>> (That's a rhetorical question... I will go ahead and try anyway, disabling
>>>> the FS core option to use timerfd, as per the Wiki).
>>>>
>>>>
>>>>
>>>> On Thu, Aug 29, 2013 at 11:21 AM, Jeff Leung <jleung at v10networks.ca>wrote:
>>>>
>>>>> If I can recall correctly, timerfd will be used automatically
>>>>> regardless in recent versions of FreeSWITCH if it?s available.****
>>>>>
>>>>> ** **
>>>>>
>>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
>>>>> Acharya
>>>>> *Sent:* Wednesday, August 28, 2013 9:35 PM
>>>>> *To:* FreeSWITCH Users Help
>>>>> *Subject:* [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd --
>>>>> preferred approach****
>>>>>
>>>>> ** **
>>>>>
>>>>> Assuming that one has full control over the kernel configuration and
>>>>> FS build / config options, is there a preferred mechanism between the 2 for
>>>>> letting FS have access to high resolution timers, i.e.****
>>>>>
>>>>>    - CONFIG_HZ=1000****
>>>>>    - mod_timerfd****
>>>>>
>>>>> This was one step I missed in my FS setup, and I think it might
>>>>> explain the extremely choppy, broken, garbled announcement I hear when I
>>>>> call into my gsmopen EP via phone.****
>>>>>
>>>>> J****
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 fs.user at fordior.net  Thu Aug 29 14:23:12 2013
From: fs.user at fordior.net (EL)
Date: Thu, 29 Aug 2013 12:23:12 +0200
Subject: [Freeswitch-users] Recording Duration on EC2
In-Reply-To: <004201cea446$871d6680$95583380$@207me.com>
References: <CAC5vAOQFMj4Oykxr-Adymc8HrzDjxKtcEx5bvoYJV24CE4uzwQ@mail.gmail.com>
	<004201cea446$871d6680$95583380$@207me.com>
Message-ID: <20130829102312.GF31829@0rdior.com>

Somehow this all reminds me about the importance of using the right
software configuration (and hardware) for handling real-time audio.

See for example:
http://manual.ardour.org/setting-up-your-system/the-right-computer-system-for-digital-audio/

And since you're using ubuntu:
https://help.ubuntu.com/community/UbuntuStudio/RealTimeKernel

As the ubuntu page describes, the normal ubuntu stock kernel is
different compared to the studio version.

More real-time related information, links, articles and such:
http://wiki.linuxaudio.org/wiki/real_time_info

I hope this information may assist you into solving your issue.
-- 
EL


From peter at olssononline.se  Thu Aug 29 14:54:29 2013
From: peter at olssononline.se (Peter Olsson)
Date: Thu, 29 Aug 2013 12:54:29 +0200
Subject: [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd -- preferred
	approach
In-Reply-To: <CAHDL-syLc89-+cLMPbCNfmHbi7nt04=+7dyiHoYjRiWdFkY6CQ@mail.gmail.com>
References: <CAHDL-szR+_NFAQOmbaT-pc1jfkBWXBmSZLKS9SNU=37XxJF6fw@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C8DA@IMBER.smb.curriegrad2004.ca>
	<CAHDL-sza2_KoL8OWswrYccbLEJXKXzpqGaYrza27vrE3LEepPQ@mail.gmail.com>
	<CAPEdYTj=54WhSGSS-=2U5_bGQm1=opd=VDuY1J_tecauJ6OyKg@mail.gmail.com>
	<CAHDL-sxr+QeEzy9DPfxWWwK5iq4k5Svqzmegfg=DdeXc_JUy9w@mail.gmail.com>
	<CAPEdYTgyra0E+G-VAWeQV0+muBuKUM5hNPk2-WOrCs0dqjWFgg@mail.gmail.com>
	<CAHDL-syLc89-+cLMPbCNfmHbi7nt04=+7dyiHoYjRiWdFkY6CQ@mail.gmail.com>
Message-ID: <CAPEdYTiejz+vNW=6FJxDrFfgjp4AzE66puVMYo8ZOJg9af73cA@mail.gmail.com>

I usually end up using Wireshark to trace the actual RTP stream. There are
many reasons for bad audio, timing is one, network's another one. I have no
experience of gsmopen endpoints though, so I really don't know how to take
that part any further.

/Peter


2013/8/29 Jayanth Acharya <jayachar88 at gmail.com>

> Thanks again Peter.
> So my theory attributing timer issue to the terrible audio quality goes
> away. Need to find what else could be the cause !!
>
>
> On Thu, Aug 29, 2013 at 2:36 PM, Peter Olsson <peter at olssononline.se>wrote:
>
>> Yes, it might be a typo. IIRC it should be usec for time_test. The small
>> sample of timer_test you provided seems ok to me, +/- 40 usec is probably
>> good enough.
>>
>> I think you're good to go :)
>>
>> /Peter
>>
>> 2013/8/29 Jayanth Acharya <jayachar88 at gmail.com>
>>
>>> On Thu, Aug 29, 2013 at 1:28 PM, Peter Olsson <peter at olssononline.se>wrote:
>>>
>>>>
>>>> I believe that timerfd is used automatically in the core if it detects
>>>> a system supporting it. However, I don't know how to really be sure.. :)
>>>> One thing though, if you just startup FS, if it doesn't try to do clock
>>>> calibration during startup, I think it is using timerfd stuff, so that
>>>> might be a way to "know"...
>>>>
>>>> Also, when running FS, you can try timer_test command (inside FS
>>>> console) and check out the results in there.
>>>>
>>>
>>> Thanks Peter.
>>>
>>> freeswitch at dabbian1> timer_test 10 3
>>> 2013-08-29 13:56:24.703028 [CONSOLE] mod_commands.c:827 Timer Test: 1
>>> sleep 10 10019
>>> 2013-08-29 13:56:24.712967 [CONSOLE] mod_commands.c:827 Timer Test: 2
>>> sleep 10 9930
>>> 2013-08-29 13:56:24.722996 [CONSOLE] mod_commands.c:827 Timer Test: 3
>>> sleep 10 10040
>>>
>>> Avg: 9.996ms Total Time: 29.989ms
>>>
>>> However,  for the "time_test" command in Mod_commands Wiki page -- I see
>>> that it assumes units as "microseconds" (not milliseconds, which is usually
>>> what we mean by "ms").. or is it a typo ??
>>>
>>> *... bad timer jitter is. It runs the test count times (default 10) and
>>> tries to sleep for mss microseconds. It returns the actual timer
>>> duration along with an average.
>>> *
>>>
>>> Anyhow, not sure how to interpret if the output of timer_test indicates
>>> a healthy resolution or not.
>>>
>>> /Peter
>>>>
>>>>
>>>> 2013/8/29 Jayanth Acharya <jayachar88 at gmail.com>
>>>>
>>>>> Thanks Jeff. Reading the mod_timerfd wiki page, I see that there are 2
>>>>> approaches to using timerfd,
>>>>>
>>>>> 1. Using the loadable module i.e. mod_timerfd (requiring me to build
>>>>> this module and then ensure it is loaded)
>>>>> 2. Using the support of timerfd now available in FS core itself
>>>>> (requiring me to just enable it in the switch.conf file)
>>>>>
>>>>> By default neither of those 2 seem to be true. I am on what as the Git
>>>>> HEAD last week - 22nd Aug 2013, so pretty recent source. However, it when I
>>>>> adopted the 2nd approach, i.e. enable timerfd use in FS core, by enabling
>>>>> the config in switch.conf, it didn't have the desired effect. Perhaps
>>>>> indicating that my problems might not be limited to just the timer
>>>>> resolution !!
>>>>>
>>>>> Using the method described here:
>>>>> http://www.advenage.com/topics/linux-timer-interrupt-frequency.php
>>>>>
>>>>> I tested and found that on my FS machine, the timer resolution
>>>>> available is already pretty good -- in the ~3KHz range, so shouldn't be an
>>>>> issue, if FS core is indeed correctly using timerfd.
>>>>>
>>>>> ---
>>>>> kernel timer interrupt frequency is approx. 3401 Hz
>>>>> jayachar at dabbian1:~$ ./timer_resolution_test
>>>>> kernel timer interrupt frequency is approx. 3401 Hz
>>>>> jayachar at dabbian1:~$ ./timer_resolution_test
>>>>> kernel timer interrupt frequency is approx. 3448 Hz
>>>>> jayachar at dabbian1:~$ ./timer_resolution_test
>>>>> kernel timer interrupt frequency is approx. 3021 Hz
>>>>> jayachar at dabbian1:~$ ./timer_resolution_test
>>>>> kernel timer interrupt frequency is approx. 3067 Hz
>>>>> ---
>>>>>
>>>>> I have already check that my kernel has the TIMERFD option enabled in
>>>>> the build, which explains (perhaps) the ~3KHz resolution I see.
>>>>>
>>>>> So the question now is, is FS really using the timerfd logic in it's
>>>>> core properly, or do I need to try the mod_timerfd method as well !!
>>>>> (That's a rhetorical question... I will go ahead and try anyway, disabling
>>>>> the FS core option to use timerfd, as per the Wiki).
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Aug 29, 2013 at 11:21 AM, Jeff Leung <jleung at v10networks.ca>wrote:
>>>>>
>>>>>> If I can recall correctly, timerfd will be used automatically
>>>>>> regardless in recent versions of FreeSWITCH if it?s available.****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>>>>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
>>>>>> Acharya
>>>>>> *Sent:* Wednesday, August 28, 2013 9:35 PM
>>>>>> *To:* FreeSWITCH Users Help
>>>>>> *Subject:* [Freeswitch-users] CONFIG_HZ=1000 vs mod_timerfd --
>>>>>> preferred approach****
>>>>>>
>>>>>> ** **
>>>>>>
>>>>>> Assuming that one has full control over the kernel configuration and
>>>>>> FS build / config options, is there a preferred mechanism between the 2 for
>>>>>> letting FS have access to high resolution timers, i.e.****
>>>>>>
>>>>>>    - CONFIG_HZ=1000****
>>>>>>    - mod_timerfd****
>>>>>>
>>>>>> This was one step I missed in my FS setup, and I think it might
>>>>>> explain the extremely choppy, broken, garbled announcement I hear when I
>>>>>> call into my gsmopen EP via phone.****
>>>>>>
>>>>>> J****
>>>>>>
>>>>>>
>>>>>> _________________________________________________________________________
>>>>>> Professional FreeSWITCH Consulting Services:
>>>>>> consulting at freeswitch.org
>>>>>> http://www.freeswitchsolutions.com
>>>>>>
>>>>>> 
>>>>>> 
>>>>>>
>>>>>> Official FreeSWITCH Sites
>>>>>> http://www.freeswitch.org
>>>>>> http://wiki.freeswitch.org
>>>>>> http://www.cluecon.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
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> _________________________________________________________________________
>>>>> Professional FreeSWITCH Consulting Services:
>>>>> consulting at freeswitch.org
>>>>> http://www.freeswitchsolutions.com
>>>>>
>>>>> 
>>>>> 
>>>>>
>>>>> Official FreeSWITCH Sites
>>>>> http://www.freeswitch.org
>>>>> http://wiki.freeswitch.org
>>>>> http://www.cluecon.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
>>>>>
>>>>>
>>>>
>>>>
>>>> _________________________________________________________________________
>>>> Professional FreeSWITCH Consulting Services:
>>>> consulting at freeswitch.org
>>>> http://www.freeswitchsolutions.com
>>>>
>>>> 
>>>> 
>>>>
>>>> Official FreeSWITCH Sites
>>>> http://www.freeswitch.org
>>>> http://wiki.freeswitch.org
>>>> http://www.cluecon.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
>>>>
>>>>
>>>
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>>
>>> 
>>> 
>>>
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>>
>>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 pm_zefman_r at mail.ru  Thu Aug 29 16:54:42 2013
From: pm_zefman_r at mail.ru (=?UTF-8?B?RG1pdHJpeSBTaHVtYWV2?=)
Date: Thu, 29 Aug 2013 16:54:42 +0400
Subject: [Freeswitch-users] =?utf-8?q?Speex_codec_in_low_bandwidth_mode_?=
 =?utf-8?q?=28narrowband=2C_mode_=3D_8=2C_quality_=3D_1=2C_bit-rate_=3D_3?=
 =?utf-8?b?Ljk1IGticHMp?=
Message-ID: <1377780882.634466895@f323.i.mail.ru>

 FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git a424765 2013-01-04 15:45:59Z).

I'm trying to use speex codec in mode = 8?(quality = 1, bit-rate = 3.95 kbps), but it seems that it only works with 24.6, 42.2, 44.0 kbps (quality = 10).
Can FS work with other modes/quality (I'm only interested in mode = 8)? If so, what should I do to configure it to use specific mode / quality (i.e. 8 / 1)?

I've modified file <speex.conf.xml> as follows:
<configuration name="speex.conf" description="Speex codec configuration">
? <settings name="default">

??? <param name="quality" value="1"/>??? <!-- !!! -->
??? <param name="complexity" value="10"/>??? <!-- !!! -->
??? <param name="enhancement" value="false"/>
??? <param name="vad" value="false"/>
??? <param name="vbr" value="false"/>
??? <param name="vbr-quality" value="4.0"/>
??? <param name="abr" value="0"/>
??? <param name="dtx" value="false"/>
??? ...
? </settings>
</configuration>
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From tahir at ictinnovations.com  Thu Aug 29 17:40:44 2013
From: tahir at ictinnovations.com (Tahir Almas)
Date: Thu, 29 Aug 2013 18:40:44 +0500
Subject: [Freeswitch-users] Recording Duration on EC2
In-Reply-To: <20130829102312.GF31829@0rdior.com>
References: <CAC5vAOQFMj4Oykxr-Adymc8HrzDjxKtcEx5bvoYJV24CE4uzwQ@mail.gmail.com>
	<004201cea446$871d6680$95583380$@207me.com>
	<20130829102312.GF31829@0rdior.com>
Message-ID: <CAOL0t+L9_d8Vvufh-v15sbXYfyJU3ecxSBHSgSTVBOo0s7io6g@mail.gmail.com>

Yes, you are right

I always recommend our customer to use dedicated hardware for  real time
communications

*Tahir Almas*

Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT



On Thu, Aug 29, 2013 at 3:23 PM, EL <fs.user at fordior.net> wrote:

> Somehow this all reminds me about the importance of using the right
> software configuration (and hardware) for handling real-time audio.
>
> See for example:
>
> http://manual.ardour.org/setting-up-your-system/the-right-computer-system-for-digital-audio/
>
> And since you're using ubuntu:
> https://help.ubuntu.com/community/UbuntuStudio/RealTimeKernel
>
> As the ubuntu page describes, the normal ubuntu stock kernel is
> different compared to the studio version.
>
> More real-time related information, links, articles and such:
> http://wiki.linuxaudio.org/wiki/real_time_info
>
> I hope this information may assist you into solving your issue.
> --
> EL
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 mehroz.ashraf85 at gmail.com  Thu Aug 29 17:49:12 2013
From: mehroz.ashraf85 at gmail.com (mehroz)
Date: Thu, 29 Aug 2013 06:49:12 -0700 (PDT)
Subject: [Freeswitch-users] Have someone zrtp with correct sas ?
In-Reply-To: <45A8C155-CAC4-4C87-BA8D-768559BCB9F6@freeswitch.org>
References: <BAY164-ds10EADF64CEECCFCDCF0F29C16E0@phx.gbl>
	<BANLkTintbY_pUpgEZQvyqW1o=ByH3x8Fdw@mail.gmail.com>
	<BAY164-ds8731F6DE9ECCDAD9DAFC6C16E0@phx.gbl>
	<45A8C155-CAC4-4C87-BA8D-768559BCB9F6@freeswitch.org>
Message-ID: <1377784152751-7594420.post@n2.nabble.com>

Hi Brian, 

This is somewhat confusing what does it mean by "who wins". According to my
understanding, when FS is declared as trusted MiTM, it should pass the same
SAS to called party, showing everything is OK in between after knowing same
SAS on both ends.

We are trusting on FS that he wont deceive us ! And so on you :)



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Have-someone-zrtp-with-correct-sas-tp6496129p7594420.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From akostenko at broadvox.com  Thu Aug 29 18:09:58 2013
From: akostenko at broadvox.com (Kostenko, Alex)
Date: Thu, 29 Aug 2013 17:09:58 +0300
Subject: [Freeswitch-users] mod_xml_curl load balancing
Message-ID: <521F5636.7030502@broadvox.com>

Hi list,
Could somebody tell me  mod_xml_curl has ability to make load balancing 
in last master git or I need to use additional patch for this?
I can't find info about this.

Thanks.


From jmesquita at freeswitch.org  Thu Aug 29 18:21:18 2013
From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=)
Date: Thu, 29 Aug 2013 11:21:18 -0300
Subject: [Freeswitch-users] mod_xml_curl load balancing
In-Reply-To: <521F5636.7030502@broadvox.com>
References: <521F5636.7030502@broadvox.com>
Message-ID: <D9456349-2DD4-4404-B0FA-6E99BBD2F6D0@freeswitch.org>

I guess you would normally use regular web infrastructure for that such as nginx or other sort. 

Sent from my iPhone

On Aug 29, 2013, at 11:09 AM, "Kostenko, Alex" <akostenko at broadvox.com> wrote:

> Hi list,
> Could somebody tell me  mod_xml_curl has ability to make load balancing 
> in last master git or I need to use additional patch for this?
> I can't find info about this.
> 
> Thanks.
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 fs.user at fordior.net  Thu Aug 29 19:03:48 2013
From: fs.user at fordior.net (EL)
Date: Thu, 29 Aug 2013 17:03:48 +0200
Subject: [Freeswitch-users] Recording Duration on EC2
In-Reply-To: <CAOL0t+L9_d8Vvufh-v15sbXYfyJU3ecxSBHSgSTVBOo0s7io6g@mail.gmail.com>
References: <CAC5vAOQFMj4Oykxr-Adymc8HrzDjxKtcEx5bvoYJV24CE4uzwQ@mail.gmail.com>
	<004201cea446$871d6680$95583380$@207me.com>
	<20130829102312.GF31829@0rdior.com>
	<CAOL0t+L9_d8Vvufh-v15sbXYfyJU3ecxSBHSgSTVBOo0s7io6g@mail.gmail.com>
Message-ID: <20130829150348.GG31829@0rdior.com>

Tahir,

> I always recommend our customer to use dedicated hardware for  real time
> communications

It's not only using dedicated hardware, although it's better. I you use
dedicated hardware with a standard linux kernel, you still can have
trouble. I you look at the third provided URL, there are several
references to other usefull links that provide additional information
which is relevant if Real-Time processing is important to you.

For example:
https://rt.wiki.kernel.org/index.php/Frequently_Asked_Questions

So I'm curious if the problems persist when the kernel/distro is
replaced/customized with/to an RT version (even when it's still a
virtual machine). I would love to get feedback on possible test
results...

-- 
EL


From david.villasmil.work at gmail.com  Thu Aug 29 19:10:32 2013
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 29 Aug 2013 17:10:32 +0200
Subject: [Freeswitch-users] Random not working?
Message-ID: <CAFGRPVq1uzQ2G=VU4Jt6RtUW77y2gcMx1JPR=DpQr-N=VbBc1A@mail.gmail.com>

Hello guys,

I've got this in my dialplan:

<action application="set" data="rand_val=${expr
randomize(&x);ceil(random(0,100,&x))}"/>
 <action application="log" data="INFO Random dialplan ${expr
randomize(&x);ceil(random(0,100,&x))}"/>
<action application="log" data="INFO Random value is ${rand_val}"/>

and this is what i see in my cli:

Dialplan: sofia/external/12345 at 1.2.3.4 Regex (PASS) [respond]
destination_number(1234) =~ /^.*/ break=on-false
Dialplan: sofia/external/12345 at 1.2.3.4 Action set(rand_val=${expr
randomize(&x);ceil(random(0,100,&x))})
Dialplan: sofia/external/12345 at 1.2.3.4 Action log(INFO Random dialplan
${expr randomize(&x);ceil(random(0,100,&x))})
Dialplan: sofia/external/12345 at 1.2.3.4 Action log(INFO Random value is
${rand_val})
2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:167
(sofia/external/12345 at 1.2.3.4) State Change CS_ROUTING -> CS_EXECUTE
2013-08-29 16:40:48.834592 [DEBUG] switch_core_session.c:1341 Send signal
sofia/external/12345 at 1.2.3.4 [BREAK]
2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:471
(sofia/external/12345 at 1.2.3.4) State ROUTING going to sleep
2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:416
(sofia/external/12345 at 1.2.3.4) Running State Change CS_EXECUTE
2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:478
(sofia/external/12345 at 1.2.3.4) State EXECUTE
2013-08-29 16:40:48.834592 [DEBUG] mod_sofia.c:230 sofia/external/
12345 at 1.2.3.4 SOFIA EXECUTE
2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:209
sofia/external/12345 at 1.2.3.4 Standard EXECUTE
EXECUTE sofia/external/12345 at 1.2.3.4 log(INFO ASR 40 we got )
2013-08-29 16:40:48.854572 [INFO] mod_dptools.c:1567 ASR 40 we got


Why is the "${rand_val}" not being parsed?

Thanks,

David
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From byron at theclarkfamily.name  Thu Aug 29 08:27:14 2013
From: byron at theclarkfamily.name (Byron Clark)
Date: Wed, 28 Aug 2013 22:27:14 -0600
Subject: [Freeswitch-users] Recording Duration on EC2
In-Reply-To: <004201cea446$871d6680$95583380$@207me.com>
References: <CAC5vAOQFMj4Oykxr-Adymc8HrzDjxKtcEx5bvoYJV24CE4uzwQ@mail.gmail.com>
	<004201cea446$871d6680$95583380$@207me.com>
Message-ID: <CAC5vAOQ+rBkJCBhPSmDN9VabArUg9DdXeikZmMFTHhK+JjQh3Q@mail.gmail.com>

On Wed, Aug 28, 2013 at 5:30 PM, Stephen Dame <sdame at 207me.com> wrote:

> Here is the length of last 20 shows, so how do you monitor steal time?


top shows the current steal time percentage. We've been using New Relic to
track it over time.

--
Byron Clark
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From krc at retrospekt.dk  Thu Aug 29 13:01:50 2013
From: krc at retrospekt.dk (Kim Rostgaard Christensen)
Date: Thu, 29 Aug 2013 11:01:50 +0200
Subject: [Freeswitch-users] Implications of
 [inbound|outbound]-use-callid-as-uuid in sofia.xml
In-Reply-To: <CAFiqYu=EVz5jhZjHtWDk7RVUEu-r5dR1GD+bmmhzYxw7T16qGQ@mail.gmail.com>
References: <521DC828.5010609@retrospekt.dk>
	<CAAyX+KFMQg=b35ST+yx72Xuy_6e5F-HGqqDq+rxxoCLFZ5zG9g@mail.gmail.com>
	<CAKbxfG_eG+0UsFM9=j1OV6vEuvEiQJdCfjHw-3LixchScNqXVA@mail.gmail.com>
	<CAFiqYu=EVz5jhZjHtWDk7RVUEu-r5dR1GD+bmmhzYxw7T16qGQ@mail.gmail.com>
Message-ID: <521F0DFE.1080704@retrospekt.dk>

Thank you for your answers. I'll try to see whether it is worth to
maintain a transition table for traceability instead of using the
option then.

Best
Kim

On 2013-08-28 21:50, Steven Ayre wrote:
> Another thing is storage... if your DB structure is assuming UUIDs
> are of the normal length, then longer Call-IDs are going to cause
> you truncation problems.
> 
> 
> On 28 August 2013 20:14, Anthony Minessale
> <anthony.minessale at gmail.com <mailto:anthony.minessale at gmail.com>>
> wrote:
> 
> Correct, you are at the mercy of the call-id being unique and 
> uniqueness cannot be guaranteed. Many sip endpoints at least try
> pretty hard to make sure they are unique but there is really
> nothing to enforce it.
> 
> Alternatively, you do have the sip_call_id variable present so its 
> not that hard to correlate them.
> 
> 
> 
> On Wed, Aug 28, 2013 at 1:40 PM, Ben Langfeld <ben at langfeld.co.uk 
> <mailto:ben at langfeld.co.uk>> wrote:
> 
> I would imagine the issue is that an incoming INVITE CallID might
> not strictly be unique. I'm not sure if "use-callid-as-uuid" deals
> with this at all...
> 
> 
> On 28 August 2013 06:51, Kim Rostgaard Christensen 
> <krc at retrospekt.dk <mailto:krc at retrospekt.dk>> wrote:
> 
> Hi list
> 
> First post, so I would just like to give a big hat tip to the
> people involved - extremely nice project. Well documented, nice 
> community We recently moved from Asterisk to FreeSWITCH, and have
> not looked back since. Asterisk is a very nice project, but AMI is
> less suited our needs than ESL.
> 
> Today, while mucking about with PJSUA, I discovered the call_id to
> be (entirely) different from the uuid assigned to the channel in
> FS. While this appears to remedied by the *-use-callid-as-uuid 
> options in sofia.xml I still wonder if there are any potential 
> complications/slowdown associated with using the option?
> 
> Best


-- 
Kim Rostgaard Christensen | krc at retrospekt.dk | http://retrospekt.dk


From kworm at sofnet.com  Thu Aug 29 20:28:53 2013
From: kworm at sofnet.com (Kevin Wormington)
Date: Thu, 29 Aug 2013 11:28:53 -0500
Subject: [Freeswitch-users] Random not working?
In-Reply-To: <CAFGRPVq1uzQ2G=VU4Jt6RtUW77y2gcMx1JPR=DpQr-N=VbBc1A@mail.gmail.com>
References: <CAFGRPVq1uzQ2G=VU4Jt6RtUW77y2gcMx1JPR=DpQr-N=VbBc1A@mail.gmail.com>
Message-ID: <521F76C5.6020306@sofnet.com>

I believe you would need to add inline="true" to your action setting the 
variable...see http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions

Kevin

On 08/29/2013 10:10 AM, David Villasmil wrote:
> Hello guys,
>
> I've got this in my dialplan:
>
> <action application="set" data="rand_val=${expr
> randomize(&x);ceil(random(0,100,&x))}"/>
> <action application="log" data="INFO Random dialplan ${expr
> randomize(&x);ceil(random(0,100,&x))}"/>
> <action application="log" data="INFO Random value is ${rand_val}"/>
>
> and this is what i see in my cli:
>
> Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Regex
> (PASS) [respond] destination_number(1234) =~ /^.*/ break=on-false
> Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Action
> set(rand_val=${expr randomize(&x);ceil(random(0,100,&x))})
> Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Action
> log(INFO Random dialplan ${expr randomize(&x);ceil(random(0,100,&x))})
> Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Action
> log(INFO Random value is ${rand_val})
> 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:167
> (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>) State Change
> CS_ROUTING -> CS_EXECUTE
> 2013-08-29 16:40:48.834592 [DEBUG] switch_core_session.c:1341 Send
> signal sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> [BREAK]
> 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:471
> (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>) State ROUTING
> going to sleep
> 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:416
> (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>) Running State
> Change CS_EXECUTE
> 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:478
> (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>) State EXECUTE
> 2013-08-29 16:40:48.834592 [DEBUG] mod_sofia.c:230
> sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> SOFIA EXECUTE
> 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:209
> sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Standard EXECUTE
> EXECUTE sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> log(INFO ASR
> 40 we got )
> 2013-08-29 16:40:48.854572 [INFO] mod_dptools.c:1567 ASR 40 we got
>
>
> Why is the "${rand_val}" not being parsed?
>
> Thanks,
>
> David
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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.villasmil.work at gmail.com  Thu Aug 29 23:06:10 2013
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Thu, 29 Aug 2013 21:06:10 +0200
Subject: [Freeswitch-users] Random not working?
In-Reply-To: <521F76C5.6020306@sofnet.com>
References: <CAFGRPVq1uzQ2G=VU4Jt6RtUW77y2gcMx1JPR=DpQr-N=VbBc1A@mail.gmail.com>
	<521F76C5.6020306@sofnet.com>
Message-ID: <CAFGRPVpsjqboAE4HT+Ow_4E4-bdE0Xy0jRStZttpqVU9eDsXRw@mail.gmail.com>

You the man! thanks!! I had never heard (read) or that "inline"...

David


On Thu, Aug 29, 2013 at 6:28 PM, Kevin Wormington <kworm at sofnet.com> wrote:

> I believe you would need to add inline="true" to your action setting the
> variable...see http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>
> Kevin
>
> On 08/29/2013 10:10 AM, David Villasmil wrote:
> > Hello guys,
> >
> > I've got this in my dialplan:
> >
> > <action application="set" data="rand_val=${expr
> > randomize(&x);ceil(random(0,100,&x))}"/>
> > <action application="log" data="INFO Random dialplan ${expr
> > randomize(&x);ceil(random(0,100,&x))}"/>
> > <action application="log" data="INFO Random value is ${rand_val}"/>
> >
> > and this is what i see in my cli:
> >
> > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Regex
> > (PASS) [respond] destination_number(1234) =~ /^.*/ break=on-false
> > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Action
> > set(rand_val=${expr randomize(&x);ceil(random(0,100,&x))})
> > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Action
> > log(INFO Random dialplan ${expr randomize(&x);ceil(random(0,100,&x))})
> > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Action
> > log(INFO Random value is ${rand_val})
> > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:167
> > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>) State Change
> > CS_ROUTING -> CS_EXECUTE
> > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_session.c:1341 Send
> > signal sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> [BREAK]
> > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:471
> > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>) State ROUTING
> > going to sleep
> > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:416
> > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>) Running State
> > Change CS_EXECUTE
> > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:478
> > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>) State EXECUTE
> > 2013-08-29 16:40:48.834592 [DEBUG] mod_sofia.c:230
> > sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> SOFIA EXECUTE
> > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:209
> > sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Standard EXECUTE
> > EXECUTE sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> log(INFO ASR
> > 40 we got )
> > 2013-08-29 16:40:48.854572 [INFO] mod_dptools.c:1567 ASR 40 we got
> >
> >
> > Why is the "${rand_val}" not being parsed?
> >
> > Thanks,
> >
> > David
> >
> >
> >
> > _________________________________________________________________________
> > Professional FreeSWITCH Consulting Services:
> > consulting at freeswitch.org
> > http://www.freeswitchsolutions.com
> >
> > 
> > 
> >
> > Official FreeSWITCH Sites
> > http://www.freeswitch.org
> > http://wiki.freeswitch.org
> > http://www.cluecon.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
> >
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 aseelye-lists at eltopia.com  Thu Aug 29 22:26:20 2013
From: aseelye-lists at eltopia.com (Aaron Seelye)
Date: Thu, 29 Aug 2013 11:26:20 -0700
Subject: [Freeswitch-users] Detected energy & floor in mod_conference
Message-ID: <521F924C.8030208@eltopia.com>

With callers in a conference, if I do a 'conference list' on the fs_cli, 
I can see the members in a conference and who is talking, but the 
'energy detected' field is always at 0, even though the minimum energy 
is set at 300 (changing the energy threshold doesn't seem to affect 
detected energy, it's always at 0).  Is this something that has gone by 
the wayside, is slated for implementation later, or am I just reading 
the wiki wrong?  Is it possible to detect the volume at which individual 
members are talking?

Also, what is the significance of 'floor'?  All I can find via google is 
some references to video conferences.  Does this have any effect on 
audio-only conferencing?

TIA,

-Aaron


From ssinyagin at yahoo.com  Thu Aug 29 23:44:46 2013
From: ssinyagin at yahoo.com (Stanislav Sinyagin)
Date: Thu, 29 Aug 2013 12:44:46 -0700 (PDT)
Subject: [Freeswitch-users]
 =?utf-8?q?=EF=BB=BFSpeex_codec_in_low_bandwidt?=
 =?utf-8?q?h_mode_=28narrowband=2C_mode_=3D_8=2C_quality_=3D_1=2C_bit-rate?=
 =?utf-8?b?ID0gM++7vy45NSBrYnBzKQ==?=
In-Reply-To: <1377780882.634466895@f323.i.mail.ru>
References: <1377780882.634466895@f323.i.mail.ru>
Message-ID: <1377805486.76333.YahooMailNeo@web126202.mail.ne1.yahoo.com>

?????? :)

The Speex preferred mode in each direction is announced by the receiving party:
http://tools.ietf.org/html/rfc5574#section-4.1.1

So I guess you need to use Wireshark and see what's going on between the parties.








________________________________
 From: Dmitriy Shumaev <pm_zefman_r at mail.ru>
To: freeswitch-users at lists.freeswitch.org 
Sent: Thursday, August 29, 2013 2:54 PM
Subject: [Freeswitch-users] ?Speex codec in low bandwidth mode (narrowband, mode = 8, quality = 1, bit-rate = 3?.95 kbps)
 


FreeSWITCH Version 1.2.6+git~20130104T154559Z~a4247651ca (git a424765 2013-01-04 15:45:59Z).

I'm trying to use speex codec in mode = 8?(quality = 1, bit-rate = 3.95 kbps), but it seems that it only works with 24.6, 42.2, 44.0 kbps (quality = 10).
Can FS work with other modes/quality (I'm only interested in mode = 8)? If so, what should I do to configure it to use specific mode / quality (i.e. 8 / 1)?

I've modified file <speex.conf.xml> as follows:
<configuration name="speex.conf" description="Speex codec configuration">
? <settings name="default">

??? <param name="quality" value="1"/>??? <!-- !!! -->
??? <param name="complexity" value="10"/>??? <!-- !!! -->
??? <param name="enhancement" value="false"/>
??? <param name="vad" value="false"/>
??? <param name="vbr" value="false"/>
??? <param name="vbr-quality" value="4.0"/>
??? <param name="abr" value="0"/>
??? <param name="dtx" value="false"/>
??? ...
? </settings>
</configuration>

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.com

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From msc at freeswitch.org  Fri Aug 30 03:30:56 2013
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 29 Aug 2013 16:30:56 -0700
Subject: [Freeswitch-users] Random not working?
In-Reply-To: <CAFGRPVpsjqboAE4HT+Ow_4E4-bdE0Xy0jRStZttpqVU9eDsXRw@mail.gmail.com>
References: <CAFGRPVq1uzQ2G=VU4Jt6RtUW77y2gcMx1JPR=DpQr-N=VbBc1A@mail.gmail.com>
	<521F76C5.6020306@sofnet.com>
	<CAFGRPVpsjqboAE4HT+Ow_4E4-bdE0Xy0jRStZttpqVU9eDsXRw@mail.gmail.com>
Message-ID: <CAKzWOxWQhFUQLn0bLz3uaHNX7RRvCBR_TxGzUMrhHMZ=3vh8pQ@mail.gmail.com>

tsk tsk! you should be reading about that in chapters 5 and 8 of the
FreeSWITCH book! ;) Also this page has some extra info you might appreciate:

https://wiki.freeswitch.org/wiki/Dialplan_XML#Availability_of_Variables

-MC


On Thu, Aug 29, 2013 at 12:06 PM, David Villasmil <
david.villasmil.work at gmail.com> wrote:

> You the man! thanks!! I had never heard (read) or that "inline"...
>
> David
>
>
> On Thu, Aug 29, 2013 at 6:28 PM, Kevin Wormington <kworm at sofnet.com>wrote:
>
>> I believe you would need to add inline="true" to your action setting the
>> variable...see
>> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>>
>> Kevin
>>
>> On 08/29/2013 10:10 AM, David Villasmil wrote:
>> > Hello guys,
>> >
>> > I've got this in my dialplan:
>> >
>> > <action application="set" data="rand_val=${expr
>> > randomize(&x);ceil(random(0,100,&x))}"/>
>> > <action application="log" data="INFO Random dialplan ${expr
>> > randomize(&x);ceil(random(0,100,&x))}"/>
>> > <action application="log" data="INFO Random value is ${rand_val}"/>
>> >
>> > and this is what i see in my cli:
>> >
>> > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Regex
>> > (PASS) [respond] destination_number(1234) =~ /^.*/ break=on-false
>> > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Action
>> > set(rand_val=${expr randomize(&x);ceil(random(0,100,&x))})
>> > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Action
>> > log(INFO Random dialplan ${expr randomize(&x);ceil(random(0,100,&x))})
>> > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Action
>> > log(INFO Random value is ${rand_val})
>> > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:167
>> > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>) State Change
>> > CS_ROUTING -> CS_EXECUTE
>> > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_session.c:1341 Send
>> > signal sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> [BREAK]
>> > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:471
>> > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>) State ROUTING
>> > going to sleep
>> > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:416
>> > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>) Running State
>> > Change CS_EXECUTE
>> > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:478
>> > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>) State EXECUTE
>> > 2013-08-29 16:40:48.834592 [DEBUG] mod_sofia.c:230
>> > sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> SOFIA EXECUTE
>> > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:209
>> > sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> Standard EXECUTE
>> > EXECUTE sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> log(INFO
>> ASR
>> > 40 we got )
>> > 2013-08-29 16:40:48.854572 [INFO] mod_dptools.c:1567 ASR 40 we got
>> >
>> >
>> > Why is the "${rand_val}" not being parsed?
>> >
>> > Thanks,
>> >
>> > David
>> >
>> >
>> >
>> >
>> _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
>> > http://www.freeswitch.org
>> > http://wiki.freeswitch.org
>> > http://www.cluecon.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
>> >
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Michael S Collins
Twitter: @mercutioviz
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
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From terry at digital-outpost.com  Fri Aug 30 07:43:40 2013
From: terry at digital-outpost.com (Terry Barnum)
Date: Thu, 29 Aug 2013 20:43:40 -0700
Subject: [Freeswitch-users] Google voice call fails
In-Reply-To: <C782E830-FFB9-4DC9-9BFE-82CBDAB8EDD0@digital-outpost.com>
References: <9665D20D-7235-4058-8666-14A654692503@digital-outpost.com>
	<006501cea2e0$26b95fe0$742c1fa0$@v10networks.ca>
	<C782E830-FFB9-4DC9-9BFE-82CBDAB8EDD0@digital-outpost.com>
Message-ID: <DD6DEED2-CE6A-4996-B8A5-405898D53DC3@digital-outpost.com>

Hi FS gurus,

Would someone be so kind as to post an example that shows what Jeff is describing about IP address settings in the dingaling profile for a machine with a private IP? Are we talking about the profile in conf/jingle_profiles? The rtp-ip and ext-rtp-ip settings?

Here's an excerpt of the log. As a freeswitch noob I'm unclear about what exactly is failing.
...
Dialplan: sofia/internal/102 at 172.16.1.199 parsing [default->gvoice_out] continue=false
Dialplan: sofia/internal/102 at 172.16.1.199 Regex (PASS) [gvoice_out] destination_number(18581234567) =~ /^1(\d{10})$/ break=on-false
Dialplan: sofia/internal/102 at 172.16.1.199 Action set(call_direction=outbound)
Dialplan: sofia/internal/102 at 172.16.1.199 Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/102 at 172.16.1.199 Action set(effective_caller_id_name=${outbound_caller_id_name})
Dialplan: sofia/internal/102 at 172.16.1.199 Action set(effective_caller_id_number=${outbound_caller_id_number})
Dialplan: sofia/internal/102 at 172.16.1.199 Action set(inherit_codec=true)
Dialplan: sofia/internal/102 at 172.16.1.199 Action bridge(dingaling/gtalk/+7609876543 at voice.google.com)
2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:167 (sofia/internal/102 at 172.16.1.199) State Change CS_ROUTING -> CS_EXECUTE
2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal sofia/internal/102 at 172.16.1.199 [BREAK]
2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:471 (sofia/internal/102 at 172.16.1.199) State ROUTING going to sleep
2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (sofia/internal/102 at 172.16.1.199) Running State Change CS_EXECUTE
2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:478 (sofia/internal/102 at 172.16.1.199) State EXECUTE
2013-08-26 20:45:32.847979 [DEBUG] mod_sofia.c:230 sofia/internal/102 at 172.16.1.199 SOFIA EXECUTE
2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:209 sofia/internal/102 at 172.16.1.199 Standard EXECUTE
EXECUTE sofia/internal/102 at 172.16.1.199 hash(insert/172.16.1.199-spymap/102/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391)
EXECUTE sofia/internal/102 at 172.16.1.199 hash(insert/172.16.1.199-last_dial/102/18581234567)
EXECUTE sofia/internal/102 at 172.16.1.199 hash(insert/172.16.1.199-last_dial/global/1ffd9854-0ecb-11e3-bfcb-8d6f93e4c391)
EXECUTE sofia/internal/102 at 172.16.1.199 export(RFC2822_DATE=Mon, 26 Aug 2013 20:45:32 -0700)
2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1222 EXPORT (export_vars) [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32 -0700]
EXECUTE sofia/internal/102 at 172.16.1.199 set(call_direction=outbound)
2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102 at 172.16.1.199 SET [call_direction]=[outbound]
EXECUTE sofia/internal/102 at 172.16.1.199 set(hangup_after_bridge=true)
2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102 at 172.16.1.199 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/102 at 172.16.1.199 set(effective_caller_id_name=Terry)
2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102 at 172.16.1.199 SET [effective_caller_id_name]=[Terry]
EXECUTE sofia/internal/102 at 172.16.1.199 set(effective_caller_id_number=760-999-9999)
2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102 at 172.16.1.199 SET [effective_caller_id_number]=[760-999-9999]
EXECUTE sofia/internal/102 at 172.16.1.199 set(inherit_codec=true)
2013-08-26 20:45:32.847979 [DEBUG] mod_dptools.c:1393 sofia/internal/102 at 172.16.1.199 SET [inherit_codec]=[true]
EXECUTE sofia/internal/102 at 172.16.1.199 bridge(dingaling/gtalk/+7609876543 at voice.google.com)
2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102 at 172.16.1.199 EXPORTING[export_vars] [domain_name]=[172.16.1.199] to event
2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:1176 sofia/internal/102 at 172.16.1.199 EXPORTING[export_vars] [RFC2822_DATE]=[Mon, 26 Aug 2013 20:45:32-0700] to event
2013-08-26 20:45:32.847979 [DEBUG] switch_ivr_originate.c:2060 Parsing global variables
2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:7FJY7u0J8fngzGK3msd2SKzr/LMLZTkJ+QRamvwG]
2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1028 Set Local Key [1 AES_CM_128_HMAC_SHA1_80 inline:dP4OPlIvkj7VWF0evrM1gjHC1nBS1Y/YKMfrfa7N]
2013-08-26 20:45:32.847979 [NOTICE] switch_channel.c:1030 New Channel dingaling/gtalk/+7609876543 at voice.google.com [200e0388-0ecb-11e3-bfdb-8d6f93e4c391]
2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2665 (dingaling/gtalk/+7609876543 at voice.google.com) State Change CS_NEW -> CS_INIT
2013-08-26 20:45:32.847979 [DEBUG] switch_core_session.c:1341 Send signal dingaling/gtalk/+7609876543 at voice.google.com [BREAK]
2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:2095 dingaling/gtalk/+7609876543 at voice.google.com CHANNEL KILL
2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:416 (dingaling/gtalk/+7609876543 at voice.google.com) Running State Change CS_INIT
2013-08-26 20:45:32.847979 [DEBUG] switch_core_state_machine.c:455 (dingaling/gtalk/+7609876543 at voice.google.com) State INIT
2013-08-26 20:45:32.847979 [NOTICE] mod_dingaling.c:1816 Ring-Ready dingaling/gtalk/+7609876543 at voice.google.com!
2013-08-26 20:45:32.847979 [DEBUG] switch_channel.c:3180 (dingaling/gtalk/+7609876543 at voice.google.com) Callstate Change DOWN -> RINGING
2013-08-26 20:45:32.847979 [DEBUG] mod_dingaling.c:1692 Don't have my audio codec yet here's one
2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1738 Don't have video codec.
2013-08-26 20:45:32.867959 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU at 8000]
2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888
2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1738 Don't have video codec.
2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1749 Send Describe [PCMU at 8000]
2013-08-26 20:45:33.288612 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32116
2013-08-26 20:45:33.408000 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:33430
2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1584 Send rtp Candidate 72.197.53.144:33430 [3ToneGgvu6KLidpo]
2013-08-26 20:45:33.408000 [DEBUG] mod_dingaling.c:1560 Stun Lookup Local 172.16.1.199:32117
2013-08-26 20:45:33.568012 [INFO] mod_dingaling.c:1570 Stun Success 72.197.53.144:39390
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1584 Send rtcp Candidate 72.197.53.144:39390 [IYN0LV3N3sDDCJws]
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates.
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates.
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates
2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates
2013-08-26 20:45:34.427964 [DEBUG] sofia_reg.c:2232 Changing expire time to 90 by request of proxy sip:callcentric.com
2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888
2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates
2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto
2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305
2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates
2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates
2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates
2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888
2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543 at voice.google.com
...

Thanks for any help,
-Terry

On Aug 28, 2013, at 6:03 PM, Terry Barnum wrote:

> Sorry, I don't know what's considered "set correctly" in the dingaling profile. Here's what's in jingle_profiles/v_172.16.1.199_gtalk.xml. I've also tried setting ext-rtp-ip to $${external_rtp_ip}.
> 
> <include>
>  <profile type="client">
>    <param name="name" value="gtalk"/>
>    <param name="login" value="USER at gmail.com/talk"/>
>    <param name="password" value="MYPASS"/>
>    <param name="dialplan" value="XML"/>
>    <param name="context" value="public"/>
>    <param name="message" value="Jingle all the way"/>
>    <param name="rtp-ip" value="auto"/>
>    <param name="ext-rtp-ip" value="auto-nat"/>
>    <param name="auto-login" value="true"/>
>    <param name="sasl" value="plain"/>
>    <param name="server" value="talk.google.com"/>
>    <param name="tls" value="true"/>
>    <param name="use-rtp-timer" value="true"/>
>    <param name="exten" value="102"/>
>    <param name="vad" value="both"/>
>    <param name="candidate-acl" value="wan.auto"/>
>    <param name="local-network-acl" value="localnet.auto"/>
>  </profile>
> </include>
> 
> Thanks for any pointers.
> 
> -Terry
> 
> On Aug 26, 2013, at 9:44 PM, Jeff Leung wrote:
> 
>> Check to see if your IP addressing settings are set correctly within the
>> dingaling profile.
>> 
>> Most of the times you'll need to have the profile to listen internally but
>> have dingaling to specify an external IP address when it talks to the
>> outside world.
>> 
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
>>> users-bounces at lists.freeswitch.org] On Behalf Of Terry Barnum
>>> Sent: Monday, August 26, 2013 9:33 PM
>>> To: FreeSWITCH Users Help
>>> Subject: [Freeswitch-users] Google voice call fails
>>> 
>>> I've been reading the freeswitch book (great read!) and playing with my
>> first
>>> freeswitch install on a Mac mini host with freeswitch installed via git
>> into a
>>> Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones
>> and
>>> a Yealink T32G, internally between extensions, using a free DID to test
>>> incoming calls and a free but limited iptel account to test outbound.
>>> 
>>> My next step was to try and play with real outbound calls so I followed
>> these
>>> instructions for Google voice:
>>> <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and
>>> are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just
>>> PCMU but no change. xmpp appears to be connecting and authenticating to
>>> google correctly.
>>> 
>>> The log of an attempted call (edited for privacy) is at
>>> <http://pastebin.freeswitch.org/21338>
>>> 
>>> Snippets of the log that looks suspicious to a freeswitch noob:
>>> 
>>> 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0
>>> rtp candidates.
>>> 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0
>>> rtcp candidates.
>>> 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0
>>> video_rtp candidates
>>> 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0
>>> video_rctp candidates
>>> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing
>>> session for 2155444888
>>> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates
>>> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate
>>> 74.125.141.127:19305 PASS ACL wan.auto
>>> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp
>>> Candidate 74.125.141.127:19305
>>> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp
>>> candidates
>>> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates
>>> 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp
>>> candidates
>>> 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing
>>> session for 2155444888
>>> 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup
>>> dingaling/gtalk/+7609876543 at voice.google.com
>>> 
>>> Where should I be looking to try and fix? Freeswitch is behind the
>> firewall
>>> with a private IP but STUN says it's working (or at least it looks that
>> way to
>>> me).



From rajsaini at gmail.com  Fri Aug 30 08:18:02 2013
From: rajsaini at gmail.com (Raj Saini)
Date: Fri, 30 Aug 2013 09:48:02 +0530
Subject: [Freeswitch-users] Roaming Extensions?
In-Reply-To: <20130828104652.GA31829@0rdior.com>
References: <934ED40B-C55D-48A2-8730-6F3B9FF1CFE3@newzealand.co.nz>
	<BA6A05D48E282C4C9989496521266BA102987A2AEF@ex2.connex.local>
	<20130827233148.GB28980@0rdior.com> <521D9ACF.4010000@aerea.de>
	<CAMbW9oKAECac-Mbs6w7NzU-+YT97V_wB99jo5hdw6rBXOSN8Nw@mail.gmail.com>
	<1377679036.25915.YahooMailNeo@web126204.mail.ne1.yahoo.com>
	<20130828104652.GA31829@0rdior.com>
Message-ID: <52201CFA.5030308@gmail.com>

Kazoo (2600hz.org) have implemented Hotdesking. You can enable this in 
the UI with feature codes.

Raj

On Wednesday 28 August 2013 04:16 PM, EL wrote:
> Thanks for the input Soeren, Andrew and Stanislav.
>
> I'm kind of wondering that nobody has implemented a clean method of
> hotdesking yet and/or documented to relevant configuration. Something
> about reinventing the wheel so to say... ;)
>
> I'm currently not in an urgent need of hotdesking, but that might change
> in the near future.



From GB at cm.nl  Fri Aug 30 09:44:47 2013
From: GB at cm.nl (Grant Bagdasarian)
Date: Fri, 30 Aug 2013 07:44:47 +0200
Subject: [Freeswitch-users] Get B-Leg Call-ID from Dialplan
In-Reply-To: <FB7D97A214987F458242ACBDF87614073B1BD1A274@clubvirtual40.ClubMessage.local>
References: <FB7D97A214987F458242ACBDF87614073B1BD1A274@clubvirtual40.ClubMessage.local>
Message-ID: <FB7D97A214987F458242ACBDF87614073B1BD1A401@clubvirtual40.ClubMessage.local>

Anyone have an idea how to do this?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Grant Bagdasarian
Sent: Wednesday, August 28, 2013 3:03 PM
To: FreeSWITCH Users Help (freeswitch-users at lists.freeswitch.org)
Subject: [Freeswitch-users] Get B-Leg Call-ID from Dialplan

Hello,

I'm using the below extension to export the A-Leg Call-ID to the B-Leg. The A-Leg  Call-ID is visible in the B-Leg INVITE as the header X-OCallID.

    <extension name="external_to_internal">
      <condition field="destination_number" expression="(\d+)$">
        <action application="set" data="hangup_after_bridge=false"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="export" data="sip_h_X-OCallID=${sip_call_id}" />
        <action application="bridge" data="sofia/internal/${destination_number}@$${sip_proxy_ip}:$${sip_proxy_port}"/>
      </condition>
    </extension>

Next to this, I also want to be able to append the Call-ID of the B-Leg to a custom header of the A-Leg.
The B-Leg Call-ID would be appended to a response message, like 100, 200 or 4xx, 5xx, 6xx.
Is this possible? If so, how? I took a look at using Lua hooks in lua.xml.conf, but I can't get the script to work.
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From jayachar88 at gmail.com  Fri Aug 30 12:16:27 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Fri, 30 Aug 2013 13:46:27 +0530
Subject: [Freeswitch-users] JIRA closes unfixed problems ??
Message-ID: <CAHDL-szw04McW_0AUu3-5YEhMvk-2LP2VT4uht+4GsAzK7g69g@mail.gmail.com>

Was browsing through the JIRA database, trying to identify any discussion
regarding the audio distortion, and I found several tickets that have been
closed (at least, that is how I interpret the closure), because they have
been inactive for some time (15 days??). Which effectively means that the
bug-closure is not by solving/fixing, but simply timing-out, in the hope
that the bug "somehow magically" got fixed in the latest release ?? I hope
I am terribly mistaken in my interpretation.

J
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From andrew at cassidywebservices.co.uk  Fri Aug 30 12:28:27 2013
From: andrew at cassidywebservices.co.uk (Andrew Cassidy)
Date: Fri, 30 Aug 2013 09:28:27 +0100
Subject: [Freeswitch-users] JIRA closes unfixed problems ??
In-Reply-To: <CAHDL-szw04McW_0AUu3-5YEhMvk-2LP2VT4uht+4GsAzK7g69g@mail.gmail.com>
References: <CAHDL-szw04McW_0AUu3-5YEhMvk-2LP2VT4uht+4GsAzK7g69g@mail.gmail.com>
Message-ID: <CAMbW9oKgArBxPJ3cjJypWzCXsFBZDovQbkASZ1Zy=XPOU586TA@mail.gmail.com>

I may be wrong, but this is my understanding:

In a way it is a timeout, but usually where they dev has been waiting for a
response from the reporter to say whether or not the fix worked.

So, in most cases, it's been fixed but the fix has not been verified by the
reporter.


On 30 August 2013 09:16, Jayanth Acharya <jayachar88 at gmail.com> wrote:

> Was browsing through the JIRA database, trying to identify any discussion
> regarding the audio distortion, and I found several tickets that have been
> closed (at least, that is how I interpret the closure), because they have
> been inactive for some time (15 days??). Which effectively means that the
> bug-closure is not by solving/fixing, but simply timing-out, in the hope
> that the bug "somehow magically" got fixed in the latest release ?? I hope
> I am terribly mistaken in my interpretation.
>
> J
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
*Andrew Cassidy BSc (Hons) MBCS SSCA*
Managing Director


*T <info at cassidywebservices.co.uk> *03300 100 960
*F<info at cassidywebservices.co.uk>
 *03300 100 961
*E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
*W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
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From bob.mccarthy at experient.com  Fri Aug 30 13:34:31 2013
From: bob.mccarthy at experient.com (Bob McCarthy)
Date: Fri, 30 Aug 2013 03:34:31 -0600
Subject: [Freeswitch-users] Conferencing with SLA
Message-ID: <01e001cea564$2285a6a0$6790f3e0$@experient.com>

I am trying to see if there is a way to keep a conference alive in SLA when
the Shared Lines drop out of a call, i.e. call comes in,  a shared line
answers and then conferences out to a gateway or some other non-shared line
using i.e. bgapi expand originate  sofia/internal/2001@${somewhere}
&sofia_sla(uuid_sharedLine) . (now three channels on the call)

 

Presently when the shared line hangs up the entire call ends.  I would like
the two or more non-shared entities to remain conferenced.  

 

 

Bob McCarthy

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From jayachar88 at gmail.com  Fri Aug 30 16:27:53 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Fri, 30 Aug 2013 17:57:53 +0530
Subject: [Freeswitch-users] JIRA closes unfixed problems ??
In-Reply-To: <CAMbW9oKgArBxPJ3cjJypWzCXsFBZDovQbkASZ1Zy=XPOU586TA@mail.gmail.com>
References: <CAHDL-szw04McW_0AUu3-5YEhMvk-2LP2VT4uht+4GsAzK7g69g@mail.gmail.com>
	<CAMbW9oKgArBxPJ3cjJypWzCXsFBZDovQbkASZ1Zy=XPOU586TA@mail.gmail.com>
Message-ID: <CAHDL-sxbrwaa-o3E2-GHkWqTPV640z6X_c3vfLeZn33uyYYs2Q@mail.gmail.com>

Thanks Andrew. You are right, and I was wrong. FS project's usage of JIRA
is somewhat different than how we use JIRA at work, and thus got confused,
as I didn't find the change-management information at the place I was
looking. Sorry about the confusion folks.


On Fri, Aug 30, 2013 at 1:58 PM, Andrew Cassidy <
andrew at cassidywebservices.co.uk> wrote:

> I may be wrong, but this is my understanding:
>
> In a way it is a timeout, but usually where they dev has been waiting for
> a response from the reporter to say whether or not the fix worked.
>
> So, in most cases, it's been fixed but the fix has not been verified by
> the reporter.
>
>
> On 30 August 2013 09:16, Jayanth Acharya <jayachar88 at gmail.com> wrote:
>
>> Was browsing through the JIRA database, trying to identify any discussion
>> regarding the audio distortion, and I found several tickets that have been
>> closed (at least, that is how I interpret the closure), because they have
>> been inactive for some time (15 days??). Which effectively means that the
>> bug-closure is not by solving/fixing, but simply timing-out, in the hope
>> that the bug "somehow magically" got fixed in the latest release ?? I hope
>> I am terribly mistaken in my interpretation.
>>
>> J
>>
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> *Andrew Cassidy BSc (Hons) MBCS SSCA*
> Managing Director
>
>
> *T <info at cassidywebservices.co.uk> *03300 100 960  *F<info at cassidywebservices.co.uk>
>  *03300 100 961
> *E <info at cassidywebservices.co.uk> *andrew at cassidywebservices.co.uk
> *W <info at cassidywebservices.co.uk> *www.cassidywebservices.co.uk
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Fri Aug 30 16:39:27 2013
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 30 Aug 2013 07:39:27 -0500
Subject: [Freeswitch-users] JIRA closes unfixed problems ??
In-Reply-To: <CAHDL-sxbrwaa-o3E2-GHkWqTPV640z6X_c3vfLeZn33uyYYs2Q@mail.gmail.com>
Message-ID: <CE45FCAF.79D7E%krice@freeswitch.org>

Also don?t forget, we do manually close unresolved bugs on a regular
basis... This is only done after multiple attempts to get more information
from a reporter, and we can not figure out how to duplicate the issue. I
would say that most of these issues are user error but we have no way to
really know that because well, the reporters need to provide more
information... 

So for everyone else reading this, if you open a Jira, please, follow up
when people ask for more info... It just helps make the world a better
place!


On 8/30/13 7:27 AM, "Jayanth Acharya" <jayachar88 at gmail.com> wrote:

> Thanks Andrew. You are right, and I was wrong. FS project's usage of JIRA is
> somewhat different than how we use JIRA at work, and thus got confused, as I
> didn't find the change-management information at the place I was looking.
> Sorry about the confusion folks.
> 
> 
> On Fri, Aug 30, 2013 at 1:58 PM, Andrew Cassidy
> <andrew at cassidywebservices.co.uk> wrote:
>> I may be wrong, but this is my understanding:
>> 
>> In a way it is a timeout, but usually where they dev has been waiting for a
>> response from the reporter to say whether or not the fix worked.?
>> 
>> So, in most cases, it's been fixed but the fix has not been verified by the
>> reporter.
>> 
>> 
>> On 30 August 2013 09:16, Jayanth Acharya <jayachar88 at gmail.com> wrote:
>>> Was browsing through the JIRA database, trying to identify any discussion
>>> regarding the audio distortion, and I found several tickets that have been
>>> closed (at least, that is how I interpret the closure), because they have
>>> been inactive for some time (15 days??). Which effectively means that the
>>> bug-closure is not by solving/fixing, but simply timing-out, in the hope
>>> that the bug "somehow magically" got fixed in the latest release ?? I hope I
>>> am terribly mistaken in my interpretation.
>>> 
>>> J
>>> 
>>> 
>>> 
>>> _________________________________________________________________________
>>> Professional FreeSWITCH Consulting Services:
>>> consulting at freeswitch.org
>>> http://www.freeswitchsolutions.com
>>> 
>>> 
>>> 
>>> 
>>> Official FreeSWITCH Sites
>>> http://www.freeswitch.org
>>> http://wiki.freeswitch.org
>>> http://www.cluecon.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
>>> 
>> 
>> 

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From nneul at mst.edu  Fri Aug 30 16:55:48 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Fri, 30 Aug 2013 07:55:48 -0500
Subject: [Freeswitch-users] Random not working?
In-Reply-To: <CAKzWOxWQhFUQLn0bLz3uaHNX7RRvCBR_TxGzUMrhHMZ=3vh8pQ@mail.gmail.com>
References: <CAFGRPVq1uzQ2G=VU4Jt6RtUW77y2gcMx1JPR=DpQr-N=VbBc1A@mail.gmail.com>
	<521F76C5.6020306@sofnet.com>
	<CAFGRPVpsjqboAE4HT+Ow_4E4-bdE0Xy0jRStZttpqVU9eDsXRw@mail.gmail.com>
	<CAKzWOxWQhFUQLn0bLz3uaHNX7RRvCBR_TxGzUMrhHMZ=3vh8pQ@mail.gmail.com>
Message-ID: <52209654.5040506@mst.edu>

FYI, even the book isn't completely consistent - I know there was at least one place in it where it said something 
explicitly about not being able to use a set variable to make further dialplan decisions with no mention of inline.

I can try and find it again if it's something you want to update for a future rev.

-- Nathan

On 08/29/2013 06:30 PM, Michael Collins wrote:
> tsk tsk! you should be reading about that in chapters 5 and 8 of the FreeSWITCH book! ;) Also this page has some extra
> info you might appreciate:
>
> https://wiki.freeswitch.org/wiki/Dialplan_XML#Availability_of_Variables
>
> -MC
>
>
> On Thu, Aug 29, 2013 at 12:06 PM, David Villasmil <david.villasmil.work at gmail.com
> <mailto:david.villasmil.work at gmail.com>> wrote:
>
>     You the man! thanks!! I had never heard (read) or that "inline"...
>
>     David
>
>
>     On Thu, Aug 29, 2013 at 6:28 PM, Kevin Wormington <kworm at sofnet.com <mailto:kworm at sofnet.com>> wrote:
>
>         I believe you would need to add inline="true" to your action setting the
>         variable...see http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
>
>         Kevin
>
>         On 08/29/2013 10:10 AM, David Villasmil wrote:
>          > Hello guys,
>          >
>          > I've got this in my dialplan:
>          >
>          > <action application="set" data="rand_val=${expr
>          > randomize(&x);ceil(random(0,100,&x))}"/>
>          > <action application="log" data="INFO Random dialplan ${expr
>          > randomize(&x);ceil(random(0,100,&x))}"/>
>          > <action application="log" data="INFO Random value is ${rand_val}"/>
>          >
>          > and this is what i see in my cli:
>          >
>          > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>> Regex
>          > (PASS) [respond] destination_number(1234) =~ /^.*/ break=on-false
>          > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>
>         Action
>          > set(rand_val=${expr randomize(&x);ceil(random(0,100,&x))})
>          > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>
>         Action
>          > log(INFO Random dialplan ${expr randomize(&x);ceil(random(0,100,&x))})
>          > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>
>         Action
>          > log(INFO Random value is ${rand_val})
>          > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:167
>          > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>) State Change
>          > CS_ROUTING -> CS_EXECUTE
>          > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_session.c:1341 Send
>          > signal sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>> [BREAK]
>          > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:471
>          > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>) State ROUTING
>          > going to sleep
>          > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:416
>          > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>) Running State
>          > Change CS_EXECUTE
>          > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:478
>          > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>) State EXECUTE
>          > 2013-08-29 16:40:48.834592 [DEBUG] mod_sofia.c:230
>          > sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>> SOFIA EXECUTE
>          > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_state_machine.c:209
>          > sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>> Standard
>         EXECUTE
>          > EXECUTE sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>
>         log(INFO ASR
>          > 40 we got )
>          > 2013-08-29 16:40:48.854572 [INFO] mod_dptools.c:1567 ASR 40 we got
>          >
>          >
>          > Why is the "${rand_val}" not being parsed?
>          >
>          > Thanks,
>          >
>          > David
>          >
>          >
>          >
>          > _________________________________________________________________________
>          > Professional FreeSWITCH Consulting Services:
>          > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>          > http://www.freeswitchsolutions.com
>          >
>          > 
>          > 
>          >
>          > Official FreeSWITCH Sites
>          > http://www.freeswitch.org
>          > http://wiki.freeswitch.org
>          > http://www.cluecon.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
>          >
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.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
>
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
>
> --
> Michael S Collins
> Twitter: @mercutioviz
> http://www.FreeSWITCH.org
> http://www.ClueCon.com
> http://www.OSTAG.org
>

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From david.villasmil.work at gmail.com  Fri Aug 30 17:01:16 2013
From: david.villasmil.work at gmail.com (David Villasmil)
Date: Fri, 30 Aug 2013 15:01:16 +0200
Subject: [Freeswitch-users] Random not working?
In-Reply-To: <52209654.5040506@mst.edu>
References: <CAFGRPVq1uzQ2G=VU4Jt6RtUW77y2gcMx1JPR=DpQr-N=VbBc1A@mail.gmail.com>
	<521F76C5.6020306@sofnet.com>
	<CAFGRPVpsjqboAE4HT+Ow_4E4-bdE0Xy0jRStZttpqVU9eDsXRw@mail.gmail.com>
	<CAKzWOxWQhFUQLn0bLz3uaHNX7RRvCBR_TxGzUMrhHMZ=3vh8pQ@mail.gmail.com>
	<52209654.5040506@mst.edu>
Message-ID: <CAFGRPVpPUKBKOJy6DSoRhuDv9Nj04cDWuVsePU4_s9E2wVYbuQ@mail.gmail.com>

I do have the book :)


On Fri, Aug 30, 2013 at 2:55 PM, Nathan Neulinger <nneul at mst.edu> wrote:

> FYI, even the book isn't completely consistent - I know there was at least
> one place in it where it said something
> explicitly about not being able to use a set variable to make further
> dialplan decisions with no mention of inline.
>
> I can try and find it again if it's something you want to update for a
> future rev.
>
> -- Nathan
>
> On 08/29/2013 06:30 PM, Michael Collins wrote:
> > tsk tsk! you should be reading about that in chapters 5 and 8 of the
> FreeSWITCH book! ;) Also this page has some extra
> > info you might appreciate:
> >
> > https://wiki.freeswitch.org/wiki/Dialplan_XML#Availability_of_Variables
> >
> > -MC
> >
> >
> > On Thu, Aug 29, 2013 at 12:06 PM, David Villasmil <
> david.villasmil.work at gmail.com
> > <mailto:david.villasmil.work at gmail.com>> wrote:
> >
> >     You the man! thanks!! I had never heard (read) or that "inline"...
> >
> >     David
> >
> >
> >     On Thu, Aug 29, 2013 at 6:28 PM, Kevin Wormington <kworm at sofnet.com<mailto:
> kworm at sofnet.com>> wrote:
> >
> >         I believe you would need to add inline="true" to your action
> setting the
> >         variable...see
> http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions
> >
> >         Kevin
> >
> >         On 08/29/2013 10:10 AM, David Villasmil wrote:
> >          > Hello guys,
> >          >
> >          > I've got this in my dialplan:
> >          >
> >          > <action application="set" data="rand_val=${expr
> >          > randomize(&x);ceil(random(0,100,&x))}"/>
> >          > <action application="log" data="INFO Random dialplan ${expr
> >          > randomize(&x);ceil(random(0,100,&x))}"/>
> >          > <action application="log" data="INFO Random value is
> ${rand_val}"/>
> >          >
> >          > and this is what i see in my cli:
> >          >
> >          > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>
> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>> Regex
> >          > (PASS) [respond] destination_number(1234) =~ /^.*/
> break=on-false
> >          > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>
> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>
> >         Action
> >          > set(rand_val=${expr randomize(&x);ceil(random(0,100,&x))})
> >          > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>
> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>
> >         Action
> >          > log(INFO Random dialplan ${expr
> randomize(&x);ceil(random(0,100,&x))})
> >          > Dialplan: sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>
> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>
> >         Action
> >          > log(INFO Random value is ${rand_val})
> >          > 2013-08-29 16:40:48.834592 [DEBUG]
> switch_core_state_machine.c:167
> >          > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:
> 12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>) State Change
> >          > CS_ROUTING -> CS_EXECUTE
> >          > 2013-08-29 16:40:48.834592 [DEBUG] switch_core_session.c:1341
> Send
> >          > signal sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>
> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>> [BREAK]
> >          > 2013-08-29 16:40:48.834592 [DEBUG]
> switch_core_state_machine.c:471
> >          > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:
> 12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>) State ROUTING
> >          > going to sleep
> >          > 2013-08-29 16:40:48.834592 [DEBUG]
> switch_core_state_machine.c:416
> >          > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:
> 12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>) Running State
> >          > Change CS_EXECUTE
> >          > 2013-08-29 16:40:48.834592 [DEBUG]
> switch_core_state_machine.c:478
> >          > (sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:
> 12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>) State EXECUTE
> >          > 2013-08-29 16:40:48.834592 [DEBUG] mod_sofia.c:230
> >          > sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:
> 12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>> SOFIA EXECUTE
> >          > 2013-08-29 16:40:48.834592 [DEBUG]
> switch_core_state_machine.c:209
> >          > sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4> <mailto:
> 12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>> Standard
> >         EXECUTE
> >          > EXECUTE sofia/external/12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>
> <mailto:12345 at 1.2.3.4 <mailto:12345 at 1.2.3.4>>
> >         log(INFO ASR
> >          > 40 we got )
> >          > 2013-08-29 16:40:48.854572 [INFO] mod_dptools.c:1567 ASR 40
> we got
> >          >
> >          >
> >          > Why is the "${rand_val}" not being parsed?
> >          >
> >          > Thanks,
> >          >
> >          > David
> >          >
> >          >
> >          >
> >          >
> _________________________________________________________________________
> >          > Professional FreeSWITCH Consulting Services:
> >          > consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >          > http://www.freeswitchsolutions.com
> >          >
> >          > 
> >          > 
> >          >
> >          > Official FreeSWITCH Sites
> >          > http://www.freeswitch.org
> >          > http://wiki.freeswitch.org
> >          > http://www.cluecon.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
> >          >
> >
> >
> _________________________________________________________________________
> >         Professional FreeSWITCH Consulting Services:
> >         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >         http://www.freeswitchsolutions.com
> >
> >         
> >         
> >
> >         Official FreeSWITCH Sites
> >         http://www.freeswitch.org
> >         http://wiki.freeswitch.org
> >         http://www.cluecon.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
> >
> >
> >
> >
> _________________________________________________________________________
> >     Professional FreeSWITCH Consulting Services:
> >     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
> >     http://www.freeswitchsolutions.com
> >
> >     
> >     
> >
> >     Official FreeSWITCH Sites
> >     http://www.freeswitch.org
> >     http://wiki.freeswitch.org
> >     http://www.cluecon.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
> >
> >
> >
> >
> > --
> > Michael S Collins
> > Twitter: @mercutioviz
> > http://www.FreeSWITCH.org
> > http://www.ClueCon.com
> > http://www.OSTAG.org
> >
>
> --
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nneul at mst.edu  Fri Aug 30 19:20:13 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Fri, 30 Aug 2013 10:20:13 -0500
Subject: [Freeswitch-users] Suggestion for additional syntax for
	"continue_on_fail"
Message-ID: <5220B82D.1020502@mst.edu>

It seems like there are a bunch of codes that would typically indicate a problem with a gateway, as opposed to a valid 
return - like no answer or busy.

My current list that I'm working with that I'm treating as "upstream failed to handle it for some reason" is:

CALL_REJECTED,GATEWAY_DOWN,RECOVERY_ON_TIMER_EXPIRE,NORMAL_TEMPORARY_FAILURE,DESTINATION_OUT_OF_ORDER,
NORMAL_UNSPECIFIED,UNALLOCATED_NUMBER,NO_ROUTE_DESTINATION

It would be nice to be able to both use negation type syntax, as well as 'collection's.

I was thinking something like this:

     ALL,!USER_BUSY,!NO_ANSWER

Example from OpenSSL:

	SSLCipherSuite ALL:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:NULL-MD5:NULL-SHA


For now, I'm just using a variable for it defined centrally.

-- Nathan

------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From jayachar88 at gmail.com  Fri Aug 30 19:25:41 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Fri, 30 Aug 2013 20:55:41 +0530
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
Message-ID: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>

Can anyone having the book let me know if it has material on mod_gsmopen ?
It doesn't seem to be a module that many people use, and not sure the book
would cover all modules, but it is what I am very interested in.

I am looking at the general architecture, design principles, configuration
etc. Basically something, that is not there in the Wiki.
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From sdame at 207me.com  Fri Aug 30 19:32:18 2013
From: sdame at 207me.com (Stephen Dame)
Date: Fri, 30 Aug 2013 11:32:18 -0400
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
In-Reply-To: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>
References: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>
Message-ID: <00fe01cea596$1d3cbbc0$57b63340$@207me.com>

Latest freeswitch 1.2 index

 

mod_gsmopen module

about 346

GSM with 348, 349

Skype 347, 348

 

Regards,

Stephen

 

HostBBB - Online Learning Solutions  http://www.hostbbb.com
<http://www.hostbbb.com/> 

207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jayanth
Acharya
Sent: Friday, August 30, 2013 11:26 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book

 

Can anyone having the book let me know if it has material on mod_gsmopen ?
It doesn't seem to be a module that many people use, and not sure the book
would cover all modules, but it is what I am very interested in.

I am looking at the general architecture, design principles, configuration
etc. Basically something, that is not there in the Wiki.

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From sdame at 207me.com  Fri Aug 30 19:35:12 2013
From: sdame at 207me.com (Stephen Dame)
Date: Fri, 30 Aug 2013 11:35:12 -0400
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
In-Reply-To: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>
References: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>
Message-ID: <010301cea596$84c036f0$8e40a4d0$@207me.com>

I would suggest picking up all three books as pdf.  Mine are left open on my
desktop all the time for reference

 

Original 1.2 book

Alternative endpoints 264

Skype and GSM endpoints 264

Skype with mod_skypopen 265

GSM with mod_gsmopen 266

TDM with OpenZAP and FreeTDM 267

 

Regards,

Stephen

 

HostBBB - Online Learning Solutions  http://www.hostbbb.com
<http://www.hostbbb.com/> 

207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jayanth
Acharya
Sent: Friday, August 30, 2013 11:26 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book

 

Can anyone having the book let me know if it has material on mod_gsmopen ?
It doesn't seem to be a module that many people use, and not sure the book
would cover all modules, but it is what I am very interested in.

I am looking at the general architecture, design principles, configuration
etc. Basically something, that is not there in the Wiki.

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From gmaruzz at gmail.com  Fri Aug 30 19:39:01 2013
From: gmaruzz at gmail.com (Giovanni Maruzzelli)
Date: Fri, 30 Aug 2013 17:39:01 +0200
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
In-Reply-To: <010301cea596$84c036f0$8e40a4d0$@207me.com>
References: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>
	<010301cea596$84c036f0$8e40a4d0$@207me.com>
Message-ID: <CALXCt0r5vhS8kW-s=ZeR9DAHJ4YsnCvMrRKWuUubJLoez1yX_w@mail.gmail.com>

The only authoritative source for gsmopen is the wiki page at the moment.

http://wiki.freeswitch.org/wiki/GSMopen


-giovanni




On Fri, Aug 30, 2013 at 5:35 PM, Stephen Dame <sdame at 207me.com> wrote:

> *I would suggest picking up all three books as pdf.  Mine are left open
> on my desktop all the time for reference*
>
> * *
>
> *Original 1.2 book*
>
> *Alternative endpoints 264*****
>
> Skype and GSM endpoints 264****
>
> Skype with mod_skypopen 265****
>
> GSM with mod_gsmopen 266****
>
> TDM with OpenZAP and FreeTDM 267****
>
> ** **
>
> Regards,****
>
> Stephen****
>
> ** **
>
> HostBBB ? Online Learning Solutions  http://www.hostbbb.com****
>
> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
> Acharya
> *Sent:* Friday, August 30, 2013 11:26 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Question about the latest FreeSWITCH book***
> *
>
> ** **
>
> Can anyone having the book let me know if it has material on mod_gsmopen ?
> It doesn't seem to be a module that many people use, and not sure the book
> would cover all modules, but it is what I am very interested in.
>
>
> I am looking at the general architecture, design principles, configuration
> etc. Basically something, that is not there in the Wiki.****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618
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From jayachar88 at gmail.com  Fri Aug 30 19:56:01 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Fri, 30 Aug 2013 21:26:01 +0530
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
In-Reply-To: <CALXCt0r5vhS8kW-s=ZeR9DAHJ4YsnCvMrRKWuUubJLoez1yX_w@mail.gmail.com>
References: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>
	<010301cea596$84c036f0$8e40a4d0$@207me.com>
	<CALXCt0r5vhS8kW-s=ZeR9DAHJ4YsnCvMrRKWuUubJLoez1yX_w@mail.gmail.com>
Message-ID: <CAHDL-sx+iU+qJrUezLeqcZ4rx3a38tXutdojKAeE7LEYXX_OCw@mail.gmail.com>

Thanks folks... and thanks Giovanni.

Anyhow, I am still going to order the book... before the great deal on the
eBook goes away.

If anyone has purchased the FreeSWITCH 1.2 book in ebook format, can they
confirm if it is PDF that I can read on my PC and Android tablet ? Or does
it require the special App/client to read it ? Will order it for sure if it
doesn't require special reader and not locked to a single device.


On Fri, Aug 30, 2013 at 9:09 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>wrote:

> The only authoritative source for gsmopen is the wiki page at the moment.
>
> http://wiki.freeswitch.org/wiki/GSMopen
>
>
> -giovanni
>
>
>
>
> On Fri, Aug 30, 2013 at 5:35 PM, Stephen Dame <sdame at 207me.com> wrote:
>
>> *I would suggest picking up all three books as pdf.  Mine are left open
>> on my desktop all the time for reference*
>>
>> * *
>>
>> *Original 1.2 book*
>>
>> *Alternative endpoints 264*****
>>
>> Skype and GSM endpoints 264****
>>
>> Skype with mod_skypopen 265****
>>
>> GSM with mod_gsmopen 266****
>>
>> TDM with OpenZAP and FreeTDM 267****
>>
>> ** **
>>
>> Regards,****
>>
>> Stephen****
>>
>> ** **
>>
>> HostBBB ? Online Learning Solutions  http://www.hostbbb.com****
>>
>> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
>> Acharya
>> *Sent:* Friday, August 30, 2013 11:26 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] Question about the latest FreeSWITCH book**
>> **
>>
>> ** **
>>
>> Can anyone having the book let me know if it has material on mod_gsmopen
>> ? It doesn't seem to be a module that many people use, and not sure the
>> book would cover all modules, but it is what I am very interested in.
>>
>>
>> I am looking at the general architecture, design principles,
>> configuration etc. Basically something, that is not there in the Wiki.***
>> *
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>>
>
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 7799099 at gmail.com  Fri Aug 30 19:41:43 2013
From: 7799099 at gmail.com (Alex S)
Date: Fri, 30 Aug 2013 08:41:43 -0700 (PDT)
Subject: [Freeswitch-users] Can't make inbound call to webrtc-registered
	user (sipml5)
Message-ID: <1377877303126-7594447.post@n2.nabble.com>

Hi,
i'm trying to make inbound call to webrtc-registered user, which use sipml5
client. But i have an error (USER_NOT_REGISTERED)

=======================================
2013-08-30 17:30:02.016741 [NOTICE] switch_ivr_originate.c:2661 Cannot
create outgoing channel of type [error] cause: [USER_NOT_REGISTERED]
2013-08-30 17:30:02.016741 [DEBUG] switch_ivr_originate.c:3632 Originate
Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2013-08-30 17:30:02.016741 [NOTICE] switch_ivr_originate.c:2661 Cannot
create outgoing channel of type [user] cause: [USER_NOT_REGISTERED]
2013-08-30 17:30:02.016741 [DEBUG] switch_ivr_originate.c:3632 Originate
Resulted in Error Cause: 606 [USER_NOT_REGISTERED]
2013-08-30 17:30:02.016741 [INFO] mod_dptools.c:3194 Originate Failed. 
Cause: USER_NOT_REGISTERED
=======================================

list_users
=======================================
fs_user_ext_0YY|ext_incoming|YYY.YYY.YYY.YYY|default|error/user_not_registered||FS
User External 0XX|990XX
=======================================

I can't see user in users list, BUT i can see him at registrations list

sofia status profile INTERFACE reg
=======================================
Call-ID:        d0xxf526-9zz2-2db0-0yy4-f666abdd4d92
User:           990XX at YYY.YYY.YYY.YYY
Contact:        "FS EXT User 0XX"
<sip:990XX at dxxjal23ls0d.invalid;rtcweb-breaker=yes;transport=ws;fs_nat=yes;fs_path=sip%3A990XX%40XXX.XXX.XXX.XXX%3A56251%3Brtcweb-breaker%3Dyes%3Btransport%3Dws>
Agent:          IM-client/OMA1.0 sipML5-v1.2013.08.10B
Status:         Registered(WS-NAT)(unknown) EXP(2013-08-30 17:31:02)
EXPSECS(246)
Host:           serverZZZ
IP:             XXX.XXX.XXX.XXX
Port:           56251
Auth-User:      990XX
Auth-Realm:     YYY.YYY.YYY.YYY
MWI-Account:    990XX at YYY.YYY.YYY.YYY
=======================================

dilaplan
=======================================
  <context name="ext_group_0XX_outgoing">

    <extension name="ext_group_0XX_outgoing" continue="true">
      <condition field="destination_number" expression="^(99(0[0-9]{2}))$">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="vm_auto_play=false"/>
        <action application="set" data="call_timeout=60"/>
        <action application="set" data="ringback=${ru-ring}"/>
        <action application="set" data="transfer_ringback=${ru-ring}"/>

        <action application="set"
data="account_id=${user_data(${destination_number}@${domain_name} attr
id)}"/>

        <action application="export" data="sip_invite_to_params=exten=$1"/>
        <action application="export" data="sip_callee_id_name=$1"/>
        <action application="export" data="sip_callee_id_number=$1"/>

        <action application="bridge"
data="user/${account_id}@$${location_1}"/>

        <action application="answer"/>
        <action application="voicemail" data="default voicemail_1 99$2"/>
        <action application="hangup"/>
      </condition>
    </extension>

  </context>
=======================================

How it possible to make inbound call to this user?

With Great Regards



--
View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Can-t-make-inbound-call-to-webrtc-registered-user-sipml5-tp7594447.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From sdame at 207me.com  Fri Aug 30 20:05:39 2013
From: sdame at 207me.com (Stephen Dame)
Date: Fri, 30 Aug 2013 12:05:39 -0400
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
In-Reply-To: <CAHDL-sx+iU+qJrUezLeqcZ4rx3a38tXutdojKAeE7LEYXX_OCw@mail.gmail.com>
References: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>	<010301cea596$84c036f0$8e40a4d0$@207me.com>	<CALXCt0r5vhS8kW-s=ZeR9DAHJ4YsnCvMrRKWuUubJLoez1yX_w@mail.gmail.com>
	<CAHDL-sx+iU+qJrUezLeqcZ4rx3a38tXutdojKAeE7LEYXX_OCw@mail.gmail.com>
Message-ID: <012301cea59a$c5d30290$517907b0$@207me.com>

When you order from http://www.packtpub.com/  you can download PDFs that
work on android and pcs, it embeds your name in pdf to discourage sharing.

 

Also I subscribe to the while library for 21 month.  and you can view them
all online all the time, and 1 book a month is free to download included in
this price.

 

Regards,

Stephen

 

HostBBB - Online Learning Solutions   <http://www.hostbbb.com/>
http://www.hostbbb.com

207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jayanth
Acharya
Sent: Friday, August 30, 2013 11:56 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Question about the latest FreeSWITCH book

 

Thanks folks... and thanks Giovanni.

Anyhow, I am still going to order the book... before the great deal on the
eBook goes away.

If anyone has purchased the FreeSWITCH 1.2 book in ebook format, can they
confirm if it is PDF that I can read on my PC and Android tablet ? Or does
it require the special App/client to read it ? Will order it for sure if it
doesn't require special reader and not locked to a single device.

 

On Fri, Aug 30, 2013 at 9:09 PM, Giovanni Maruzzelli <gmaruzz at gmail.com
<mailto:gmaruzz at gmail.com> > wrote:

The only authoritative source for gsmopen is the wiki page at the moment.

 

http://wiki.freeswitch.org/wiki/GSMopen

 

 

-giovanni

 

 

 

On Fri, Aug 30, 2013 at 5:35 PM, Stephen Dame <sdame at 207me.com
<mailto:sdame at 207me.com> > wrote:

I would suggest picking up all three books as pdf.  Mine are left open on my
desktop all the time for reference

 

Original 1.2 book

Alternative endpoints 264

Skype and GSM endpoints 264

Skype with mod_skypopen 265

GSM with mod_gsmopen 266

TDM with OpenZAP and FreeTDM 267

 

Regards,

Stephen

 

HostBBB - Online Learning Solutions  http://www.hostbbb.com
<http://www.hostbbb.com/> 

207 Technology Group Inc.   1-888-229-9756 <tel:1-888-229-9756>   skype:
Stephen_Dame

 

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
Jayanth Acharya
Sent: Friday, August 30, 2013 11:26 AM
To: FreeSWITCH Users Help
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book

 

Can anyone having the book let me know if it has material on mod_gsmopen ?
It doesn't seem to be a module that many people use, and not sure the book
would cover all modules, but it is what I am very interested in.



I am looking at the general architecture, design principles, configuration
etc. Basically something, that is not there in the Wiki.

 

_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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





 

-- 
Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org <mailto:consulting at freeswitch.org> 
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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

 

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From jayachar88 at gmail.com  Fri Aug 30 20:27:26 2013
From: jayachar88 at gmail.com (Jayanth Acharya)
Date: Fri, 30 Aug 2013 21:57:26 +0530
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
In-Reply-To: <012301cea59a$c5d30290$517907b0$@207me.com>
References: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>
	<010301cea596$84c036f0$8e40a4d0$@207me.com>
	<CALXCt0r5vhS8kW-s=ZeR9DAHJ4YsnCvMrRKWuUubJLoez1yX_w@mail.gmail.com>
	<CAHDL-sx+iU+qJrUezLeqcZ4rx3a38tXutdojKAeE7LEYXX_OCw@mail.gmail.com>
	<012301cea59a$c5d30290$517907b0$@207me.com>
Message-ID: <CAHDL-sxs--_Oe7PpqAVGXcDO4x+GSNTDYpzZajqEB-rUDuZhjA@mail.gmail.com>

Thanks Stephen. Good to hear that the PDF can be used on multiple devices.


On Fri, Aug 30, 2013 at 9:35 PM, Stephen Dame <sdame at 207me.com> wrote:

> When you order from http://www.packtpub.com/  you can download PDFs that
> work on android and pcs, it embeds your name in pdf to discourage sharing.
> ****
>
> ** **
>
> Also I subscribe to the while library for 21 month?  and you can view them
> all online all the time, and 1 book a month is free to download included in
> this price.****
>
> ** **
>
> Regards,****
>
> Stephen****
>
> ** **
>
> HostBBB ? Online Learning Solutions  http://www.hostbbb.com****
>
> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
> Acharya
> *Sent:* Friday, August 30, 2013 11:56 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Question about the latest FreeSWITCH
> book****
>
> ** **
>
> Thanks folks... and thanks Giovanni.****
>
> Anyhow, I am still going to order the book... before the great deal on the
> eBook goes away.****
>
> If anyone has purchased the FreeSWITCH 1.2 book in ebook format, can they
> confirm if it is PDF that I can read on my PC and Android tablet ? Or does
> it require the special App/client to read it ? Will order it for sure if it
> doesn't require special reader and not locked to a single device.****
>
> ** **
>
> On Fri, Aug 30, 2013 at 9:09 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:****
>
> The only authoritative source for gsmopen is the wiki page at the moment.*
> ***
>
> ** **
>
> http://wiki.freeswitch.org/wiki/GSMopen****
>
> ** **
>
> ** **
>
> -giovanni****
>
> ** **
>
> ** **
>
> ** **
>
> On Fri, Aug 30, 2013 at 5:35 PM, Stephen Dame <sdame at 207me.com> wrote:****
>
> *I would suggest picking up all three books as pdf.  Mine are left open
> on my desktop all the time for reference*****
>
> * *****
>
> *Original 1.2 book*****
>
> *Alternative endpoints 264*****
>
> Skype and GSM endpoints 264****
>
> Skype with mod_skypopen 265****
>
> GSM with mod_gsmopen 266****
>
> TDM with OpenZAP and FreeTDM 267****
>
>  ****
>
> Regards,****
>
> Stephen****
>
>  ****
>
> HostBBB ? Online Learning Solutions  http://www.hostbbb.com****
>
> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
> Acharya
> *Sent:* Friday, August 30, 2013 11:26 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Question about the latest FreeSWITCH book***
> *
>
>  ****
>
> Can anyone having the book let me know if it has material on mod_gsmopen ?
> It doesn't seem to be a module that many people use, and not sure the book
> would cover all modules, but it is what I am very interested in.****
>
>
>
> I am looking at the general architecture, design principles, configuration
> etc. Basically something, that is not there in the Wiki.****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
>
>
> ****
>
> ** **
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Fri Aug 30 20:30:09 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Fri, 30 Aug 2013 22:00:09 +0530
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
In-Reply-To: <012301cea59a$c5d30290$517907b0$@207me.com>
References: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>
	<010301cea596$84c036f0$8e40a4d0$@207me.com>
	<CALXCt0r5vhS8kW-s=ZeR9DAHJ4YsnCvMrRKWuUubJLoez1yX_w@mail.gmail.com>
	<CAHDL-sx+iU+qJrUezLeqcZ4rx3a38tXutdojKAeE7LEYXX_OCw@mail.gmail.com>
	<012301cea59a$c5d30290$517907b0$@207me.com>
Message-ID: <CAMvhjJt7FGg6xGxJYqaOUKJF8ag2h5_Uxe8dOEasV9F1jPj12g@mail.gmail.com>

Jayanth i will mail you the book...i have the pdf format....if you have not
orderd it yet...!!!
Regards
Ashish Mishra
On Aug 30, 2013 9:37 PM, "Stephen Dame" <sdame at 207me.com> wrote:

> When you order from http://www.packtpub.com/  you can download PDFs that
> work on android and pcs, it embeds your name in pdf to discourage sharing.
> ****
>
> ** **
>
> Also I subscribe to the while library for 21 month?  and you can view them
> all online all the time, and 1 book a month is free to download included in
> this price.****
>
> ** **
>
> Regards,****
>
> Stephen****
>
> ** **
>
> HostBBB ? Online Learning Solutions  http://www.hostbbb.com****
>
> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>
> ** **
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
> Acharya
> *Sent:* Friday, August 30, 2013 11:56 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Question about the latest FreeSWITCH
> book****
>
> ** **
>
> Thanks folks... and thanks Giovanni.****
>
> Anyhow, I am still going to order the book... before the great deal on the
> eBook goes away.****
>
> If anyone has purchased the FreeSWITCH 1.2 book in ebook format, can they
> confirm if it is PDF that I can read on my PC and Android tablet ? Or does
> it require the special App/client to read it ? Will order it for sure if it
> doesn't require special reader and not locked to a single device.****
>
> ** **
>
> On Fri, Aug 30, 2013 at 9:09 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:****
>
> The only authoritative source for gsmopen is the wiki page at the moment.*
> ***
>
> ** **
>
> http://wiki.freeswitch.org/wiki/GSMopen****
>
> ** **
>
> ** **
>
> -giovanni****
>
> ** **
>
> ** **
>
> ** **
>
> On Fri, Aug 30, 2013 at 5:35 PM, Stephen Dame <sdame at 207me.com> wrote:****
>
> *I would suggest picking up all three books as pdf.  Mine are left open
> on my desktop all the time for reference*****
>
> * *****
>
> *Original 1.2 book*****
>
> *Alternative endpoints 264*****
>
> Skype and GSM endpoints 264****
>
> Skype with mod_skypopen 265****
>
> GSM with mod_gsmopen 266****
>
> TDM with OpenZAP and FreeTDM 267****
>
>  ****
>
> Regards,****
>
> Stephen****
>
>  ****
>
> HostBBB ? Online Learning Solutions  http://www.hostbbb.com****
>
> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>
>  ****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
> Acharya
> *Sent:* Friday, August 30, 2013 11:26 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Question about the latest FreeSWITCH book***
> *
>
>  ****
>
> Can anyone having the book let me know if it has material on mod_gsmopen ?
> It doesn't seem to be a module that many people use, and not sure the book
> would cover all modules, but it is what I am very interested in.****
>
>
>
> I am looking at the general architecture, design principles, configuration
> etc. Basically something, that is not there in the Wiki.****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
>
>
> ****
>
> ** **
>
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
> ** **
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 itsme.kunnu at gmail.com  Fri Aug 30 20:31:26 2013
From: itsme.kunnu at gmail.com (Ashish Mishra)
Date: Fri, 30 Aug 2013 22:01:26 +0530
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
In-Reply-To: <CAMvhjJt7FGg6xGxJYqaOUKJF8ag2h5_Uxe8dOEasV9F1jPj12g@mail.gmail.com>
References: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>
	<010301cea596$84c036f0$8e40a4d0$@207me.com>
	<CALXCt0r5vhS8kW-s=ZeR9DAHJ4YsnCvMrRKWuUubJLoez1yX_w@mail.gmail.com>
	<CAHDL-sx+iU+qJrUezLeqcZ4rx3a38tXutdojKAeE7LEYXX_OCw@mail.gmail.com>
	<012301cea59a$c5d30290$517907b0$@207me.com>
	<CAMvhjJt7FGg6xGxJYqaOUKJF8ag2h5_Uxe8dOEasV9F1jPj12g@mail.gmail.com>
Message-ID: <CAMvhjJuPtQzvQk7o24QUEnTNV8bbv+1x3nbbqXVU51pOG8PT8g@mail.gmail.com>

Kindly let me know if you are interested.

Regards
Ashish Mishra
On Aug 30, 2013 10:00 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:

> Jayanth i will mail you the book...i have the pdf format....if you have
> not orderd it yet...!!!
> Regards
> Ashish Mishra
> On Aug 30, 2013 9:37 PM, "Stephen Dame" <sdame at 207me.com> wrote:
>
>> When you order from http://www.packtpub.com/  you can download PDFs that
>> work on android and pcs, it embeds your name in pdf to discourage sharing.
>> ****
>>
>> ** **
>>
>> Also I subscribe to the while library for 21 month?  and you can view
>> them all online all the time, and 1 book a month is free to download
>> included in this price.****
>>
>> ** **
>>
>> Regards,****
>>
>> Stephen****
>>
>> ** **
>>
>> HostBBB ? Online Learning Solutions  http://www.hostbbb.com****
>>
>> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>>
>> ** **
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
>> Acharya
>> *Sent:* Friday, August 30, 2013 11:56 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* Re: [Freeswitch-users] Question about the latest FreeSWITCH
>> book****
>>
>> ** **
>>
>> Thanks folks... and thanks Giovanni.****
>>
>> Anyhow, I am still going to order the book... before the great deal on
>> the eBook goes away.****
>>
>> If anyone has purchased the FreeSWITCH 1.2 book in ebook format, can they
>> confirm if it is PDF that I can read on my PC and Android tablet ? Or does
>> it require the special App/client to read it ? Will order it for sure if it
>> doesn't require special reader and not locked to a single device.****
>>
>> ** **
>>
>> On Fri, Aug 30, 2013 at 9:09 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
>> wrote:****
>>
>> The only authoritative source for gsmopen is the wiki page at the moment.
>> ****
>>
>> ** **
>>
>> http://wiki.freeswitch.org/wiki/GSMopen****
>>
>> ** **
>>
>> ** **
>>
>> -giovanni****
>>
>> ** **
>>
>> ** **
>>
>> ** **
>>
>> On Fri, Aug 30, 2013 at 5:35 PM, Stephen Dame <sdame at 207me.com> wrote:***
>> *
>>
>> *I would suggest picking up all three books as pdf.  Mine are left open
>> on my desktop all the time for reference*****
>>
>> * *****
>>
>> *Original 1.2 book*****
>>
>> *Alternative endpoints 264*****
>>
>> Skype and GSM endpoints 264****
>>
>> Skype with mod_skypopen 265****
>>
>> GSM with mod_gsmopen 266****
>>
>> TDM with OpenZAP and FreeTDM 267****
>>
>>  ****
>>
>> Regards,****
>>
>> Stephen****
>>
>>  ****
>>
>> HostBBB ? Online Learning Solutions  http://www.hostbbb.com****
>>
>> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>>
>>  ****
>>
>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
>> Acharya
>> *Sent:* Friday, August 30, 2013 11:26 AM
>> *To:* FreeSWITCH Users Help
>> *Subject:* [Freeswitch-users] Question about the latest FreeSWITCH book**
>> **
>>
>>  ****
>>
>> Can anyone having the book let me know if it has material on mod_gsmopen
>> ? It doesn't seem to be a module that many people use, and not sure the
>> book would cover all modules, but it is what I am very interested in.****
>>
>>
>>
>> I am looking at the general architecture, design principles,
>> configuration etc. Basically something, that is not there in the Wiki.***
>> *
>>
>> ** **
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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****
>>
>>
>>
>> ****
>>
>> ** **
>>
>> --
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618****
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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****
>>
>> ** **
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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 jleung at v10networks.ca  Fri Aug 30 21:30:24 2013
From: jleung at v10networks.ca (Jeff Leung)
Date: Fri, 30 Aug 2013 10:30:24 -0700
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
In-Reply-To: <CAMvhjJuPtQzvQk7o24QUEnTNV8bbv+1x3nbbqXVU51pOG8PT8g@mail.gmail.com>
References: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>	<010301cea596$84c036f0$8e40a4d0$@207me.com>	<CALXCt0r5vhS8kW-s=ZeR9DAHJ4YsnCvMrRKWuUubJLoez1yX_w@mail.gmail.com>	<CAHDL-sx+iU+qJrUezLeqcZ4rx3a38tXutdojKAeE7LEYXX_OCw@mail.gmail.com>	<012301cea59a$c5d30290$517907b0$@207me.com>	<CAMvhjJt7FGg6xGxJYqaOUKJF8ag2h5_Uxe8dOEasV9F1jPj12g@mail.gmail.com>
	<CAMvhjJuPtQzvQk7o24QUEnTNV8bbv+1x3nbbqXVU51pOG8PT8g@mail.gmail.com>
Message-ID: <A70AC0C1B7E7FD478A6BF19705461AA0C8DC@IMBER.smb.curriegrad2004.ca>

Please do not offer a PDF of the book on this list. I'm not quite sure
if the publisher is going to like this at all.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Ashish Mishra
Sent: Friday, August 30, 2013 9:31 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Question about the latest FreeSWITCH
book

 

Kindly let me know if you are interested.

Regards
Ashish Mishra

On Aug 30, 2013 10:00 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:

Jayanth i will mail you the book...i have the pdf format....if you have
not orderd it yet...!!!
Regards
Ashish Mishra

On Aug 30, 2013 9:37 PM, "Stephen Dame" <sdame at 207me.com> wrote:

When you order from http://www.packtpub.com/  you can download PDFs that
work on android and pcs, it embeds your name in pdf to discourage
sharing.

 

Also I subscribe to the while library for 21 month...  and you can view
them all online all the time, and 1 book a month is free to download
included in this price.

 

Regards,

Stephen

 

HostBBB - Online Learning Solutions  http://www.hostbbb.com
<http://www.hostbbb.com/> 

207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Jayanth Acharya
Sent: Friday, August 30, 2013 11:56 AM
To: FreeSWITCH Users Help
Subject: Re: [Freeswitch-users] Question about the latest FreeSWITCH
book

 

Thanks folks... and thanks Giovanni.

Anyhow, I am still going to order the book... before the great deal on
the eBook goes away.

If anyone has purchased the FreeSWITCH 1.2 book in ebook format, can
they confirm if it is PDF that I can read on my PC and Android tablet ?
Or does it require the special App/client to read it ? Will order it for
sure if it doesn't require special reader and not locked to a single
device.

 

On Fri, Aug 30, 2013 at 9:09 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
wrote:

	The only authoritative source for gsmopen is the wiki page at
the moment.

	 

	http://wiki.freeswitch.org/wiki/GSMopen

	 

	 

	-giovanni

	 

	 

	 

	On Fri, Aug 30, 2013 at 5:35 PM, Stephen Dame <sdame at 207me.com>
wrote:

		I would suggest picking up all three books as pdf.  Mine
are left open on my desktop all the time for reference

		 

		Original 1.2 book

		Alternative endpoints 264

		Skype and GSM endpoints 264

		Skype with mod_skypopen 265

		GSM with mod_gsmopen 266

		TDM with OpenZAP and FreeTDM 267

		 

		Regards,

		Stephen

		 

		HostBBB - Online Learning Solutions
http://www.hostbbb.com <http://www.hostbbb.com/> 

		207 Technology Group Inc.   1-888-229-9756  skype:
Stephen_Dame

		 

		From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Jayanth Acharya
		Sent: Friday, August 30, 2013 11:26 AM
		To: FreeSWITCH Users Help
		Subject: [Freeswitch-users] Question about the latest
FreeSWITCH book

		 

		Can anyone having the book let me know if it has
material on mod_gsmopen ? It doesn't seem to be a module that many
people use, and not sure the book would cover all modules, but it is
what I am very interested in.

		
		
		I am looking at the general architecture, design
principles, configuration etc. Basically something, that is not there in
the Wiki.

		 

	
________________________________________________________________________
_
		Professional FreeSWITCH Consulting Services:
		consulting at freeswitch.org
		http://www.freeswitchsolutions.com
		
		FreeSWITCH-powered IP PBX: The CudaTel Communication
Server
		
		
		Official FreeSWITCH Sites
		http://www.freeswitch.org
		http://wiki.freeswitch.org
		http://www.cluecon.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

	
	
	

	 

	-- 
	Sincerely,
	
	Giovanni Maruzzelli
	Cell : +39-347-2665618

	
	
________________________________________________________________________
_
	Professional FreeSWITCH Consulting Services:
	consulting at freeswitch.org
	http://www.freeswitchsolutions.com
	
	
	
	
	Official FreeSWITCH Sites
	http://www.freeswitch.org
	http://wiki.freeswitch.org
	http://www.cluecon.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

 


________________________________________________________________________
_
Professional FreeSWITCH Consulting Services:
consulting at freeswitch.org
http://www.freeswitchsolutions.com




Official FreeSWITCH Sites
http://www.freeswitch.org
http://wiki.freeswitch.org
http://www.cluecon.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 mike at jerris.com  Fri Aug 30 21:48:40 2013
From: mike at jerris.com (Michael Jerris)
Date: Fri, 30 Aug 2013 13:48:40 -0400
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
In-Reply-To: <A70AC0C1B7E7FD478A6BF19705461AA0C8DC@IMBER.smb.curriegrad2004.ca>
References: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>	<010301cea596$84c036f0$8e40a4d0$@207me.com>	<CALXCt0r5vhS8kW-s=ZeR9DAHJ4YsnCvMrRKWuUubJLoez1yX_w@mail.gmail.com>	<CAHDL-sx+iU+qJrUezLeqcZ4rx3a38tXutdojKAeE7LEYXX_OCw@mail.gmail.com>	<012301cea59a$c5d30290$517907b0$@207me.com>	<CAMvhjJt7FGg6xGxJYqaOUKJF8ag2h5_Uxe8dOEasV9F1jPj12g@mail.gmail.com>
	<CAMvhjJuPtQzvQk7o24QUEnTNV8bbv+1x3nbbqXVU51pOG8PT8g@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C8DC@IMBER.smb.curriegrad2004.ca>
Message-ID: <A3067657-9568-4012-BA01-000A7D73D97A@jerris.com>

No worries, he's removed from the list now.

Mike

On Aug 30, 2013, at 1:30 PM, Jeff Leung <jleung at v10networks.ca> wrote:

> Please do not offer a PDF of the book on this list. I?m not quite sure if the publisher is going to like this at all.
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ashish Mishra
> Sent: Friday, August 30, 2013 9:31 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Question about the latest FreeSWITCH book
>  
> Kindly let me know if you are interested.
> 
> Regards
> Ashish Mishra
> 
> On Aug 30, 2013 10:00 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:
> Jayanth i will mail you the book...i have the pdf format....if you have not orderd it yet...!!!
> Regards
> Ashish Mishra
> 
> On Aug 30, 2013 9:37 PM, "Stephen Dame" <sdame at 207me.com> wrote:
> When you order from http://www.packtpub.com/  you can download PDFs that work on android and pcs, it embeds your name in pdf to discourage sharing.
>  
> Also I subscribe to the while library for 21 month?  and you can view them all online all the time, and 1 book a month is free to download included in this price.
>  
> Regards,
> Stephen
>  
> HostBBB ? Online Learning Solutions  http://www.hostbbb.com
> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jayanth Acharya
> Sent: Friday, August 30, 2013 11:56 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] Question about the latest FreeSWITCH book
>  
> Thanks folks... and thanks Giovanni.
> 
> Anyhow, I am still going to order the book... before the great deal on the eBook goes away.
> 
> If anyone has purchased the FreeSWITCH 1.2 book in ebook format, can they confirm if it is PDF that I can read on my PC and Android tablet ? Or does it require the special App/client to read it ? Will order it for sure if it doesn't require special reader and not locked to a single device.
>  
> 
> On Fri, Aug 30, 2013 at 9:09 PM, Giovanni Maruzzelli <gmaruzz at gmail.com> wrote:
> The only authoritative source for gsmopen is the wiki page at the moment.
>  
> http://wiki.freeswitch.org/wiki/GSMopen
>  
>  
> -giovanni
>  
>  
>  
> 
> On Fri, Aug 30, 2013 at 5:35 PM, Stephen Dame <sdame at 207me.com> wrote:
> I would suggest picking up all three books as pdf.  Mine are left open on my desktop all the time for reference
> 
>  
> 
> Original 1.2 book
> 
> Alternative endpoints 264
> 
> Skype and GSM endpoints 264
> 
> Skype with mod_skypopen 265
> 
> GSM with mod_gsmopen 266
> 
> TDM with OpenZAP and FreeTDM 267
>  
> Regards,
> Stephen
>  
> HostBBB ? Online Learning Solutions  http://www.hostbbb.com
> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame
>  
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jayanth Acharya
> Sent: Friday, August 30, 2013 11:26 AM
> To: FreeSWITCH Users Help
> Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
>  
> Can anyone having the book let me know if it has material on mod_gsmopen ? It doesn't seem to be a module that many people use, and not sure the book would cover all modules, but it is what I am very interested in.
> 
> 
> I am looking at the general architecture, design principles, configuration etc. Basically something, that is not there in the Wiki.
>  
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> 
> 
>  
> -- 
> Sincerely,
> 
> Giovanni Maruzzelli
> Cell : +39-347-2665618
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
>  
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.com
> 
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
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From nneul at mst.edu  Fri Aug 30 22:14:03 2013
From: nneul at mst.edu (Nathan Neulinger)
Date: Fri, 30 Aug 2013 13:14:03 -0500
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
In-Reply-To: <A3067657-9568-4012-BA01-000A7D73D97A@jerris.com>
References: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>	<010301cea596$84c036f0$8e40a4d0$@207me.com>	<CALXCt0r5vhS8kW-s=ZeR9DAHJ4YsnCvMrRKWuUubJLoez1yX_w@mail.gmail.com>	<CAHDL-sx+iU+qJrUezLeqcZ4rx3a38tXutdojKAeE7LEYXX_OCw@mail.gmail.com>	<012301cea59a$c5d30290$517907b0$@207me.com>	<CAMvhjJt7FGg6xGxJYqaOUKJF8ag2h5_Uxe8dOEasV9F1jPj12g@mail.gmail.com>
	<CAMvhjJuPtQzvQk7o24QUEnTNV8bbv+1x3nbbqXVU51pOG8PT8g@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C8DC@IMBER.smb.curriegrad2004.ca>
	<A3067657-9568-4012-BA01-000A7D73D97A@jerris.com>
Message-ID: <5220E0EB.2010603@mst.edu>

I was going to give him the benefit of the doubt assuming he meant "mail the physical book" since he had the eBook... 
Can't even imagine the gall of the other interpretation...

-- Nathan

On 08/30/2013 12:48 PM, Michael Jerris wrote:
> No worries, he's removed from the list now.
>
> Mike
>
> On Aug 30, 2013, at 1:30 PM, Jeff Leung <jleung at v10networks.ca <mailto:jleung at v10networks.ca>> wrote:
>
>> Please do not offer a PDF of the book on this list. I?m not quite sure if the publisher is going to like this at all.
>> *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:users-bounces at lists.freeswitch.org>]*On Behalf Of*Ashish
>> Mishra
>> *Sent:*Friday, August 30, 2013 9:31 AM
>> *To:*FreeSWITCH Users Help
>> *Subject:*Re: [Freeswitch-users] Question about the latest FreeSWITCH book
>>
>> Kindly let me know if you are interested.
>>
>> Regards
>> Ashish Mishra
>>
>> On Aug 30, 2013 10:00 PM, "Ashish Mishra" <itsme.kunnu at gmail.com <mailto:itsme.kunnu at gmail.com>> wrote:
>>
>> Jayanth i will mail you the book...i have the pdf format....if you have not orderd it yet...!!!
>> Regards
>> Ashish Mishra
>>
>> On Aug 30, 2013 9:37 PM, "Stephen Dame" <sdame at 207me.com <mailto:sdame at 207me.com>> wrote:
>> When you order fromhttp://www.packtpub.com/  you can download PDFs that work on android and pcs, it embeds your name
>> in pdf to discourage sharing.
>> Also I subscribe to the while library for 21 month?  and you can view them all online all the time, and 1 book a month
>> is free to download included in this price.
>> Regards,
>> Stephen
>> HostBBB ? Online Learning Solutions http://www.hostbbb.com <http://www.hostbbb.com/>
>> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame
>> *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*Jayanth Acharya
>> *Sent:*Friday, August 30, 2013 11:56 AM
>> *To:*FreeSWITCH Users Help
>> *Subject:*Re: [Freeswitch-users] Question about the latest FreeSWITCH book
>>
>> Thanks folks... and thanks Giovanni.
>>
>> Anyhow, I am still going to order the book... before the great deal on the eBook goes away.
>>
>> If anyone has purchased the FreeSWITCH 1.2 book in ebook format, can they confirm if it is PDF that I can read on my
>> PC and Android tablet ? Or does it require the special App/client to read it ? Will order it for sure if it doesn't
>> require special reader and not locked to a single device.
>>
>> On Fri, Aug 30, 2013 at 9:09 PM, Giovanni Maruzzelli <gmaruzz at gmail.com <mailto:gmaruzz at gmail.com>> wrote:
>>
>>     The only authoritative source for gsmopen is the wiki page at the moment.
>>     http://wiki.freeswitch.org/wiki/GSMopen
>>     -giovanni
>>
>>     On Fri, Aug 30, 2013 at 5:35 PM, Stephen Dame <sdame at 207me.com <mailto:sdame at 207me.com>> wrote:
>>
>>         *I would suggest picking up all three books as pdf.  Mine are left open on my desktop all the time for reference*
>>
>>         **
>>
>>         *Original 1.2 book*
>>
>>         *Alternative endpoints 264*
>>
>>         Skype and GSM endpoints 264
>>
>>         Skype with mod_skypopen 265
>>
>>         GSM with mod_gsmopen 266
>>
>>         TDM with OpenZAP and FreeTDM 267
>>         Regards,
>>         Stephen
>>         HostBBB ? Online Learning Solutions http://www.hostbbb.com <http://www.hostbbb.com/>
>>         207 Technology Group Inc. 1-888-229-9756 <tel:1-888-229-9756>  skype: Stephen_Dame
>>         *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*Jayanth Acharya
>>         *Sent:*Friday, August 30, 2013 11:26 AM
>>         *To:*FreeSWITCH Users Help
>>         *Subject:*[Freeswitch-users] Question about the latest FreeSWITCH book
>>         Can anyone having the book let me know if it has material on mod_gsmopen ? It doesn't seem to be a module that
>>         many people use, and not sure the book would cover all modules, but it is what I am very interested in.
>>
>>
>>         I am looking at the general architecture, design principles, configuration etc. Basically something, that is
>>         not there in the Wiki.
>>
>>         _________________________________________________________________________
>>         Professional FreeSWITCH Consulting Services:
>>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>         http://www.freeswitchsolutions.com
>>
>>         
>>         
>>
>>         Official FreeSWITCH Sites
>>         http://www.freeswitch.org
>>         http://wiki.freeswitch.org
>>         http://www.cluecon.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
>>
>>
>>
>>     --
>>     Sincerely,
>>
>>     Giovanni Maruzzelli
>>     Cell : +39-347-2665618
>>
>>
>>     _________________________________________________________________________
>>     Professional FreeSWITCH Consulting Services:
>>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>>     http://www.freeswitchsolutions.com
>>
>>     
>>     
>>
>>     Official FreeSWITCH Sites
>>     http://www.freeswitch.org
>>     http://wiki.freeswitch.org
>>     http://www.cluecon.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
>>
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
>> http://www.freeswitch.org
>> http://wiki.freeswitch.org
>> http://www.cluecon.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
>

-- 
------------------------------------------------------------
Nathan Neulinger                       nneul at mst.edu
Missouri S&T Information Technology    (573) 612-1412
System Administrator - Architect


From grcamauer at gmail.com  Fri Aug 30 22:17:28 2013
From: grcamauer at gmail.com (Guillermo Ruiz Camauer)
Date: Fri, 30 Aug 2013 15:17:28 -0300
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
In-Reply-To: <A3067657-9568-4012-BA01-000A7D73D97A@jerris.com>
References: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>
	<010301cea596$84c036f0$8e40a4d0$@207me.com>
	<CALXCt0r5vhS8kW-s=ZeR9DAHJ4YsnCvMrRKWuUubJLoez1yX_w@mail.gmail.com>
	<CAHDL-sx+iU+qJrUezLeqcZ4rx3a38tXutdojKAeE7LEYXX_OCw@mail.gmail.com>
	<012301cea59a$c5d30290$517907b0$@207me.com>
	<CAMvhjJt7FGg6xGxJYqaOUKJF8ag2h5_Uxe8dOEasV9F1jPj12g@mail.gmail.com>
	<CAMvhjJuPtQzvQk7o24QUEnTNV8bbv+1x3nbbqXVU51pOG8PT8g@mail.gmail.com>
	<A70AC0C1B7E7FD478A6BF19705461AA0C8DC@IMBER.smb.curriegrad2004.ca>
	<A3067657-9568-4012-BA01-000A7D73D97A@jerris.com>
Message-ID: <CAPNTnnqoJr5XRt2iQ2B4=C84EaWDDJpEpGfizsb=ORR6HyZerg@mail.gmail.com>

Guys, the book is very good, it's not expensive, and is a great way to show
some support for this project and its authors!



On Fri, Aug 30, 2013 at 2:48 PM, Michael Jerris <mike at jerris.com> wrote:

> No worries, he's removed from the list now.
>
> Mike
>
> On Aug 30, 2013, at 1:30 PM, Jeff Leung <jleung at v10networks.ca> wrote:
>
> Please do not offer a PDF of the book on this list. I?m not quite sure if
> the publisher is going to like this at all.****
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-
> users-bounces at lists.freeswitch.org] *On Behalf Of *Ashish Mishra
> *Sent:* Friday, August 30, 2013 9:31 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Question about the latest FreeSWITCH
> book****
> ** **
>
> Kindly let me know if you are interested.****
>
> Regards
> Ashish Mishra****
> On Aug 30, 2013 10:00 PM, "Ashish Mishra" <itsme.kunnu at gmail.com> wrote:**
> **
>
> Jayanth i will mail you the book...i have the pdf format....if you have
> not orderd it yet...!!!
> Regards
> Ashish Mishra****
> On Aug 30, 2013 9:37 PM, "Stephen Dame" <sdame at 207me.com> wrote:****
> When you order from http://www.packtpub.com/  you can download PDFs that
> work on android and pcs, it embeds your name in pdf to discourage sharing.
> ****
>  ****
> Also I subscribe to the while library for 21 month?  and you can view them
> all online all the time, and 1 book a month is free to download included in
> this price.****
>  ****
> Regards,****
> Stephen****
>  ****
> HostBBB ? Online Learning Solutions  http://www.hostbbb.com****
> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>  ****
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
> Acharya
> *Sent:* Friday, August 30, 2013 11:56 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Question about the latest FreeSWITCH
> book****
>  ****
>
> Thanks folks... and thanks Giovanni.****
>
> Anyhow, I am still going to order the book... before the great deal on the
> eBook goes away.****
> If anyone has purchased the FreeSWITCH 1.2 book in ebook format, can they
> confirm if it is PDF that I can read on my PC and Android tablet ? Or does
> it require the special App/client to read it ? Will order it for sure if it
> doesn't require special reader and not locked to a single device.****
>
>  ****
> On Fri, Aug 30, 2013 at 9:09 PM, Giovanni Maruzzelli <gmaruzz at gmail.com>
> wrote:****
>
> The only authoritative source for gsmopen is the wiki page at the moment.*
> ***
>  ****
> http://wiki.freeswitch.org/wiki/GSMopen****
>  ****
>  ****
> -giovanni****
>  ****
>  ****
>
>  ****
> On Fri, Aug 30, 2013 at 5:35 PM, Stephen Dame <sdame at 207me.com> wrote:****
>
> *I would suggest picking up all three books as pdf.  Mine are left open
> on my desktop all the time for reference*****
>
> * *****
>
> *Original 1.2 book*****
>
> *Alternative endpoints 264*****
>
> Skype and GSM endpoints 264****
>
> Skype with mod_skypopen 265****
>
> GSM with mod_gsmopen 266****
> TDM with OpenZAP and FreeTDM 267****
>  ****
> Regards,****
> Stephen****
>  ****
> HostBBB ? Online Learning Solutions  http://www.hostbbb.com****
> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame****
>  ****
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jayanth
> Acharya
> *Sent:* Friday, August 30, 2013 11:26 AM
> *To:* FreeSWITCH Users Help
> *Subject:* [Freeswitch-users] Question about the latest FreeSWITCH book***
> *
>  ****
> Can anyone having the book let me know if it has material on mod_gsmopen ?
> It doesn't seem to be a module that many people use, and not sure the book
> would cover all modules, but it is what I am very interested in.****
>
>
> I am looking at the general architecture, design principles, configuration
> etc. Basically something, that is not there in the Wiki.****
>  ****
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
>
>
> ****
>  ****
> --
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
>
>  ****
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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****
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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
>
>


-- 
Guillermo Ruiz Camauer
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From aseelye-lists at eltopia.com  Fri Aug 30 23:36:36 2013
From: aseelye-lists at eltopia.com (Aaron Seelye)
Date: Fri, 30 Aug 2013 12:36:36 -0700
Subject: [Freeswitch-users] Question about the latest FreeSWITCH book
In-Reply-To: <012301cea59a$c5d30290$517907b0$@207me.com>
References: <CAHDL-sy5oi-N3ScrRXU70XB3YHjfE1zTJO=hJCQ-eUc7qyRpcA@mail.gmail.com>	<010301cea596$84c036f0$8e40a4d0$@207me.com>	<CALXCt0r5vhS8kW-s=ZeR9DAHJ4YsnCvMrRKWuUubJLoez1yX_w@mail.gmail.com>
	<CAHDL-sx+iU+qJrUezLeqcZ4rx3a38tXutdojKAeE7LEYXX_OCw@mail.gmail.com>
	<012301cea59a$c5d30290$517907b0$@207me.com>
Message-ID: <5220F444.6030807@eltopia.com>

While I bought the first two directly, I read the 1.2 book via 
O'Reilly's Safari service.

-Aaron

On 8/30/2013 9:05 AM, Stephen Dame wrote:
> When you order from http://www.packtpub.com/  you can download PDFs that
> work on android and pcs, it embeds your name in pdf to discourage sharing.
>
> Also I subscribe to the while library for 21 month?  and you can view
> them all online all the time, and 1 book a month is free to download
> included in this price.
>
> Regards,
>
> Stephen
>
> HostBBB ? Online Learning Solutions http://www.hostbbb.com
> <http://www.hostbbb.com/>
>
> 207 Technology Group Inc.   1-888-229-9756  skype: Stephen_Dame
>
> *From:*freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of
> *Jayanth Acharya
> *Sent:* Friday, August 30, 2013 11:56 AM
> *To:* FreeSWITCH Users Help
> *Subject:* Re: [Freeswitch-users] Question about the latest FreeSWITCH book
>
> Thanks folks... and thanks Giovanni.
>
> Anyhow, I am still going to order the book... before the great deal on
> the eBook goes away.
>
> If anyone has purchased the FreeSWITCH 1.2 book in ebook format, can
> they confirm if it is PDF that I can read on my PC and Android tablet ?
> Or does it require the special App/client to read it ? Will order it for
> sure if it doesn't require special reader and not locked to a single device.
>
> On Fri, Aug 30, 2013 at 9:09 PM, Giovanni Maruzzelli <gmaruzz at gmail.com
> <mailto:gmaruzz at gmail.com>> wrote:
>
>     The only authoritative source for gsmopen is the wiki page at the
>     moment.
>
>     http://wiki.freeswitch.org/wiki/GSMopen
>
>     -giovanni
>
>     On Fri, Aug 30, 2013 at 5:35 PM, Stephen Dame <sdame at 207me.com
>     <mailto:sdame at 207me.com>> wrote:
>
>         *I would suggest picking up all three books as pdf.  Mine are
>         left open on my desktop all the time for reference*
>
>         **
>
>         *Original 1.2 book*
>
>         *Alternative endpoints 264*
>
>         Skype and GSM endpoints 264
>
>         Skype with mod_skypopen 265
>
>         GSM with mod_gsmopen 266
>
>         TDM with OpenZAP and FreeTDM 267
>
>         Regards,
>
>         Stephen
>
>         HostBBB ? Online Learning Solutions http://www.hostbbb.com
>         <http://www.hostbbb.com/>
>
>         207 Technology Group Inc. 1-888-229-9756 <tel:1-888-229-9756>
>         skype: Stephen_Dame
>
>         *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 *Jayanth Acharya
>         *Sent:* Friday, August 30, 2013 11:26 AM
>         *To:* FreeSWITCH Users Help
>         *Subject:* [Freeswitch-users] Question about the latest
>         FreeSWITCH book
>
>         Can anyone having the book let me know if it has material on
>         mod_gsmopen ? It doesn't seem to be a module that many people
>         use, and not sure the book would cover all modules, but it is
>         what I am very interested in.
>
>
>
>         I am looking at the general architecture, design principles,
>         configuration etc. Basically something, that is not there in the
>         Wiki.
>
>         _________________________________________________________________________
>         Professional FreeSWITCH Consulting Services:
>         consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>         http://www.freeswitchsolutions.com
>
>         
>         
>
>         Official FreeSWITCH Sites
>         http://www.freeswitch.org
>         http://wiki.freeswitch.org
>         http://www.cluecon.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
>
>
>
>     --
>     Sincerely,
>
>     Giovanni Maruzzelli
>     Cell : +39-347-2665618
>
>
>     _________________________________________________________________________
>     Professional FreeSWITCH Consulting Services:
>     consulting at freeswitch.org <mailto:consulting at freeswitch.org>
>     http://www.freeswitchsolutions.com
>
>     
>     
>
>     Official FreeSWITCH Sites
>     http://www.freeswitch.org
>     http://wiki.freeswitch.org
>     http://www.cluecon.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
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 steveayre at gmail.com  Sat Aug 31 01:03:57 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 30 Aug 2013 22:03:57 +0100
Subject: [Freeswitch-users] Suggestion for additional syntax for
	"continue_on_fail"
In-Reply-To: <5220B82D.1020502@mst.edu>
References: <5220B82D.1020502@mst.edu>
Message-ID: <CAFiqYumHb0OMdpnpX+XR6_oNYd6rgrHzhBxhg5xj0H1prsd8zg@mail.gmail.com>

switch_channel.c line 4517 if you feel confident contributing a patch ;)


On 30 August 2013 16:20, Nathan Neulinger <nneul at mst.edu> wrote:

> It seems like there are a bunch of codes that would typically indicate a
> problem with a gateway, as opposed to a valid
> return - like no answer or busy.
>
> My current list that I'm working with that I'm treating as "upstream
> failed to handle it for some reason" is:
>
>
> CALL_REJECTED,GATEWAY_DOWN,RECOVERY_ON_TIMER_EXPIRE,NORMAL_TEMPORARY_FAILURE,DESTINATION_OUT_OF_ORDER,
> NORMAL_UNSPECIFIED,UNALLOCATED_NUMBER,NO_ROUTE_DESTINATION
>
> It would be nice to be able to both use negation type syntax, as well as
> 'collection's.
>
> I was thinking something like this:
>
>      ALL,!USER_BUSY,!NO_ANSWER
>
> Example from OpenSSL:
>
>         SSLCipherSuite
> ALL:!ADH:!EXPORT56:RC4+RSA:+HIGH:+MEDIUM:+LOW:+SSLv2:+EXP:NULL-MD5:NULL-SHA
>
>
> For now, I'm just using a variable for it defined centrally.
>
> -- Nathan
>
> ------------------------------------------------------------
> Nathan Neulinger                       nneul at mst.edu
> Missouri S&T Information Technology    (573) 612-1412
> System Administrator - Architect
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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 nickolayr at gmail.com  Sat Aug 31 01:33:46 2013
From: nickolayr at gmail.com (Nikolay Rogoshchenkov)
Date: Fri, 30 Aug 2013 17:33:46 -0400
Subject: [Freeswitch-users] inbound-bypass-media for 1.2 (stable version)
Message-ID: <CAAQCs31mkfaAuuCr0v3_5FdZM5=9uVgKYSoKKuLor1Ux02fLOQ@mail.gmail.com>

Is it possible to use this option <param name="inbound-bypass-media"
value="true"/> for 1.2 version?


--
Rogoshchenkov Nikolay
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From steveayre at gmail.com  Sat Aug 31 01:42:35 2013
From: steveayre at gmail.com (Steven Ayre)
Date: Fri, 30 Aug 2013 22:42:35 +0100
Subject: [Freeswitch-users] inbound-bypass-media for 1.2 (stable version)
In-Reply-To: <CAAQCs31mkfaAuuCr0v3_5FdZM5=9uVgKYSoKKuLor1Ux02fLOQ@mail.gmail.com>
References: <CAAQCs31mkfaAuuCr0v3_5FdZM5=9uVgKYSoKKuLor1Ux02fLOQ@mail.gmail.com>
Message-ID: <CAFiqYum+a5=Ge5BE_OE0Lx7Rk0SbXwdpxz8SXfgMyufCg70zgQ@mail.gmail.com>

Yes. It should be set on the SIP profile.


On 30 August 2013 22:33, Nikolay Rogoshchenkov <nickolayr at gmail.com> wrote:

> Is it possible to use this option <param name="inbound-bypass-media"
> value="true"/> for 1.2 version?
>
>
> --
> Rogoshchenkov Nikolay
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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  Sat Aug 31 01:44:39 2013
From: krice at freeswitch.org (Ken Rice)
Date: Fri, 30 Aug 2013 16:44:39 -0500
Subject: [Freeswitch-users] inbound-bypass-media for 1.2 (stable version)
In-Reply-To: <CAAQCs31mkfaAuuCr0v3_5FdZM5=9uVgKYSoKKuLor1Ux02fLOQ@mail.gmail.com>
Message-ID: <CE467C77.79DD0%krice@freeswitch.org>

It is... That?s been around for quite a while... What you need to know tho
is not everything works the same when you use that mode and you may
encounter NAT issues


On 8/30/13 4:33 PM, "Nikolay Rogoshchenkov" <nickolayr at gmail.com> wrote:

> Is it possible to use this option <param name="inbound-bypass-media"
> value="true"/> for 1.2 version?
> 
> 
> --
> Rogoshchenkov Nikolay
> 
> 
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
> 
> 
> 
> 
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
> http://www.cluecon.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

-- 
Ken
http://www.FreeSWITCH.org
http://www.ClueCon.com
http://www.OSTAG.org
irc.freenode.net #freeswitch

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From philq at qsystemsengineering.com  Sat Aug 31 19:15:54 2013
From: philq at qsystemsengineering.com (Phil Quesinberry)
Date: Sat, 31 Aug 2013 11:15:54 -0400
Subject: [Freeswitch-users] Lua script problem
Message-ID: <001201cea65c$ff76f070$fe64d150$@com>

Hi everyone,

Would anyone have any ideas as to why the following statement would not work
and end up assigning a null value to scripts_dir?
scripts_dir =
string.sub(debug.getinfo(1).source,2,string.len(debug.getinfo(1).source)-(st
ring.len(argv[0])+1));

I'm pretty sure that this is related to a much larger problem that we're
experiencing where MWI has stopped working completely on all extensions.  We
started experiencing intermittent problems with this after making recent
updates to FS and FusionPBX before it stopped working entirely.  Marc Crane,
the developer of FusionPBX, is under the impression that this is either an
FS bug or that something has changed with FS' handling of MWI.  I don't know
lua so I can't weigh in on this myself.

I filed a Jira on this, see the following for more details, including the
full script the above excerpt is taken from:
http://jira.freeswitch.org/browse/FS-5739

Any input would be greatly appreciated.

Phil Quesinberry
Q Systems Engineering, Inc.
(410) 969-8002
http://www.qsystemsengineering.com <http://www.qsystemsengineering.com/> 


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